Calls with no audio in both sides

Asterisk Asterisk SIP
1

I’m trying to implement a new service on my Asterisk server network.
We have a 3G mobile phone with a softphone app - Linphone, whose number is 9012 and is registered on the Asterisk server with IP # 10.192.124.101 (Sagres). Then there is a Trunksip between this server and another one with IP # 10.192.230.231 (Viriato). The Mobile is associated with a Mobile Operator and has a Public IP address. Public IP access to the Asterisk Sagres server is by NAT filtering through a Firewall. That is, it arrives by Public IP to the FW and then this maps (NAT) to the IP of Sagres (10.192.124.101). The SIP account is properly registered and can even make calls and receive. But in both situations the calls are mute in both directions. I am sending the SIP SET Debug log attached. I already did a TCPDump and in wireshark I noticed that there are RTP packets that are sent by the Viriato server, but there is no RTP packet that is sent by the Sagres server. In conversation I had with my colleague of networks, he says that in my SIP.conf should put Nat = no. Below the settings:

In Sagres:

Sip.conf:

[general]
realm=Jar209Reftel.refertelecom.pt
context=default
allowoverlap=yes
srvlookup=no
defaultexpiry=90
;qualify=8000
videosupport=yes
notifyringing = yes
notifyhold = yes
callcounter = yes
counteronpeer = yes
alwaysauthreject = yes
allowguest=no
disallow=all
allow = alaw
allow = ulaw
;allow = g722
allow = g729
allow = gsm
rtptimeout=300
nat=no
bindaddr=0.0.0.0
bindport=5060
tos=0xb8

[TRUNKSIP-VIRIATO]
type=peer
host=10.192.230.231
context=incoming-iax
disallow=all
allow = ulaw
dtmfmode=inband
canreinvite=yes
directmedia=yes
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.230.231/255.255.255.255

[TRUNKSIP-VIRIATO2]
type=peer
host=10.192.231.231
context=incoming-iax
disallow=all
allow = ulaw
dtmfmode=inband
canreinvite=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.231.231/255.255.255.255

[9012]
type=friend
callerid=“Teste_IPT_2” <9012>
context=and_int
secret=&#9012#$h
host=dynamic
dtmfmode=rfc2833
defaultuser=9012
progressinband=no
promiscredir=yes
canreinvite=no
qualify=no
;deny=0.0.0.0/0.0.0.0
;permit=10.168.242.100/255.255.255.255

Extensions.conf:

[and_int]

exten => _[1-8]XXX,1,Ringing()
exten => _[1-8]XXX,2,Dial(${TRUNKVIRIATO}/${EXTEN},180,tT)
exten => _[1-8]XXX,3,Hangup()

exten => 24067,1,Ringing()
exten => 24067,2,Dial(${TRUNKVIRIATO}/${EXTEN},180,tT)
exten => 24067,3,Hangup()

And in Viriato:

Sip.conf:

[general]
realm=Jar209Reftel.refertelecom.pt
context=default
allowoverlap=yes
srvlookup=no
defaultexpiry=90
;qualify=8000
videosupport=yes
notifyringing = yes
notifyhold = yes
callcounter = yes
counteronpeer = yes
alwaysauthreject = yes
;allowguest=no
disallow=all
allow = alaw
allow = ulaw
;allow = g722
allow = g729
allow = gsm
rtptimeout=300
nat=no
bindaddr=0.0.0.0
bindport=5060
tos=0xb8

[TRUNKSIP-SAGRES]
type=peer
host=10.192.124.101
context=incoming-tlm
disallow=all
allow = ulaw
dtmfmode=inband
;canreinvite=no
canreinvite=yes
directmedia=yes
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
permit=10.192.124.101/255.255.255.255

Extensions.conf:

exten => _99XXX,1,Ringing()
exten => _99XXX,2,Dial(${TRUNKSAGRES}/${EXTEN:1},180,tT)
exten => _99XXX,3,Hangup()

I’ve done a call from 9012 to 24067 and the executed trace log (SIP Set debug) send attached

2

I Can’t attach the log…here it is:

sagres*CLI> sip set debug on
SIP Debugging enabled
Retransmitting #2 (no NAT) to 10.192.207.56:5060:
OPTIONS sip:10.192.207.56 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK168d71c5
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as6eb0c945
To: sip:10.192.207.56
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 11e05f2f15df67362b020a830b150386@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #2 (no NAT) to 10.192.207.55:5060:
OPTIONS sip:10.192.207.55 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK4d8b1345
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as00d5b9b2
To: sip:10.192.207.55
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 6f0df6d0230d7d4c7c4f4a4b5a7f86cb@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #2 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK05659905
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0d7926ae
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 521ad5436e86cdb83088936c735fb5b7@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:62.210.143.116:5069 —>
REGISTER sip:81.90.54.197:5060 SIP/2.0
Via: SIP/2.0/UDP 62.210.143.116:5069;branch=z9hG4bK30309f9c4c0429295a3f744a;rport
From: “2004” sip:2004@81.90.54.197:5060;tag=30309f9c752d
To: “2004” sip:2004@81.90.54.197:5060
Call-ID: 9f9c4c04-7852929-5a3f744a@81.90.54.197
CSeq: 1 REGISTER
Contact: “2004” sip:2004@62.210.143.116:5069
User-Agent: VaxSIPUserAgent/3.1
Expires: 1800
Max-Forwards: 70
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 62.210.143.116:5069 (no NAT)
Sending to 62.210.143.116:5069 (no NAT)

<— Transmitting (no NAT) to 62.210.143.116:5069 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 62.210.143.116:5069;branch=z9hG4bK30309f9c4c0429295a3f744a;received=62.210.143.116;rport=5069
From: “2004” sip:2004@81.90.54.197:5060;tag=30309f9c752d
To: “2004” sip:2004@81.90.54.197:5060;tag=as7a13cae8
Call-ID: 9f9c4c04-7852929-5a3f744a@81.90.54.197
CSeq: 1 REGISTER
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“Jar209Reftel.refertelecom.pt”, nonce="63a628f7"
Content-Length: 0

Retransmitting #3 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK05659905
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0d7926ae
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 521ad5436e86cdb83088936c735fb5b7@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #4 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK05659905
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0d7926ae
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 521ad5436e86cdb83088936c735fb5b7@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

v=0
o=9012 2436 2946 IN IP4 10.117.119.172
s=Talk
c=IN IP4 10.117.119.172
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
— (14 headers 20 lines) —
Sending to 148.69.11.122:38022 (no NAT)
Sending to 148.69.11.122:38022 (no NAT)
Using INVITE request as basis request - BmY052N6TS
Found peer ‘9012’ for ‘9012’ from 148.69.11.122:38022

<— Reliably Transmitting (no NAT) to 148.69.11.122:38022 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.xprm91RdB;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as0ac191be
Call-ID: BmY052N6TS
CSeq: 20 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“Jar209Reftel.refertelecom.pt”, nonce="5834b13f"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘BmY052N6TS’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:148.69.11.122:38022 —>
INVITE sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.xprm91RdB;rport
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197
CSeq: 20 INVITE
Call-ID: BmY052N6TS
Max-Forwards: 70
Route: sip:voip-lx.cloudsolutions.pt;transport=udp;lr
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 547
Contact: sip:9012@148.69.11.122:38022;transport=udp;+sip.instance="urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38"
User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)

v=0
o=9012 2436 2946 IN IP4 10.117.119.172
s=Talk
c=IN IP4 10.117.119.172
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
— (14 headers 20 lines) —
Ignoring this INVITE request

<— SIP read from UDP:148.69.11.122:38022 —>
ACK sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.xprm91RdB;rport
Call-ID: BmY052N6TS
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as0ac191be
Contact: sip:9012@148.69.11.122:38022;transport=udp;+sip.instance="urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38"
Route: sip:voip-lx.cloudsolutions.pt;transport=udp;lr
Max-Forwards: 70
CSeq: 20 ACK

<------------->
— (9 headers 0 lines) —

<— SIP read from UDP:148.69.11.122:38022 —>
INVITE sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;rport
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197
CSeq: 21 INVITE
Call-ID: BmY052N6TS
Max-Forwards: 70
Route: sip:voip-lx.cloudsolutions.pt;transport=udp;lr
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 547
Contact: sip:9012@148.69.11.122:38022;transport=udp;+sip.instance=“urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38"
User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)
Authorization: Digest realm=“Jar209Reftel.refertelecom.pt”, nonce=“5834b13f”, algorithm=MD5, username=“9012”, uri="sip:24067@81.90.54.197”, response=“bff8f4c02d400f4f44b348a169493658”

v=0
o=9012 2436 2946 IN IP4 10.117.119.172
s=Talk
c=IN IP4 10.117.119.172
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
— (15 headers 20 lines) —
Sending to 148.69.11.122:38022 (no NAT)
Using INVITE request as basis request - BmY052N6TS
Found peer ‘9012’ for ‘9012’ from 148.69.11.122:38022
== Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 99
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 100
Found RTP audio format 102
Found audio description format opus for ID 96
Found audio description format speex for ID 97
Found audio description format speex for ID 98
Found audio description format iLBC for ID 99
Found unknown media description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 100
Found audio description format telephone-event for ID 102
Capabilities: us - (alaw|ulaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|g722|g729|opus|speex16|speex|ilbc)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.117.119.172:7076
Looking for 24067 in and_int (domain 81.90.54.197)
sip_route_dump: route/path hop: sip:9012@148.69.11.122:38022;transport=udp

<— Transmitting (no NAT) to 148.69.11.122:38022 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Length: 0

<------------>
– Executing [24067@and_int:1] Ringing(“SIP/9012-00000000”, “”) in new stack

<— Transmitting (no NAT) to 148.69.11.122:38022 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Length: 0

<------------>
– Executing [24067@and_int:2] Dial(“SIP/9012-00000000”, “SIP/TRUNKSIP-VIRIATO/24067,180,tT”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 11410
Adding codec ulaw to SDP
Reliably Transmitting (no NAT) to 10.192.230.231:5060:
INVITE sip:24067@10.192.230.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK57f7b631
Max-Forwards: 70
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231
Contact: sip:9012@10.192.124.101:5060
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 200

v=0
o=root 1219785261 1219785261 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 11410 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv

-- Called SIP/TRUNKSIP-VIRIATO/24067

<— SIP read from UDP:10.192.230.231:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK57f7b631;received=10.192.124.101
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:24067@10.192.230.231:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —

<— SIP read from UDP:10.192.230.231:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK57f7b631;received=10.192.124.101
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231;tag=as329e86b2
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:24067@10.192.230.231:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
sip_route_dump: route/path hop: sip:24067@10.192.230.231:5060
– SIP/TRUNKSIP-VIRIATO-00000001 is ringing

<— Transmitting (no NAT) to 148.69.11.122:38022 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Length: 0

<------------>

<— SIP read from UDP:10.192.230.231:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK57f7b631;received=10.192.124.101
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231;tag=as329e86b2
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:24067@10.192.230.231:5060
Content-Length: 0

Reliably Transmitting (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK62f46f0f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0e14b656
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #1 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK62f46f0f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0e14b656
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #2 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK62f46f0f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0e14b656
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #3 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK62f46f0f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0e14b656
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘6b1a43de715897e845a452271a15e272@10.192.124.101:5060’ Method: OPTIONS
Really destroying SIP dialog ‘TgTELLrQcD’ Method: REGISTER
Retransmitting #4 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK62f46f0f
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as0e14b656
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘5ed2495f3c993e40360b76642f3e1ab6@10.192.124.101:5060’ Method: OPTIONS
Really destroying SIP dialog ‘2d02bb920b5838ab548f621744745cf3@10.192.230.231:5060’ Method: OPTIONS

<— SIP read from UDP:10.192.230.231:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK57f7b631;received=10.192.124.101
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231;tag=as329e86b2
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:24067@10.192.230.231:5060
Content-Type: application/sdp
Require: timer
Content-Length: 200

v=0
o=root 1001407096 1001407096 IN IP4 10.192.230.231
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.230.231
t=0 0
m=audio 18462 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
— (14 headers 10 lines) —
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.192.230.231:18462
sip_route_dump: route/path hop: sip:24067@10.192.230.231:5060
set_destination: Parsing sip:24067@10.192.230.231:5060 for address/port to send to
set_destination: set destination to 10.192.230.231:5060
Transmitting (no NAT) to 10.192.230.231:5060:
ACK sip:24067@10.192.230.231:5060 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK00e1ce45
Max-Forwards: 70
From: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
To: sip:24067@10.192.230.231;tag=as329e86b2
Contact: sip:9012@10.192.124.101:5060
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0

-- SIP/TRUNKSIP-VIRIATO-00000001 answered SIP/9012-00000000

Audio is at 19982
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 148.69.11.122:38022 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1396079092 1396079092 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 19982 RTP/AVP 8 0 18 3 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
– Channel SIP/TRUNKSIP-VIRIATO-00000001 joined ‘simple_bridge’ basic-bridge <83a14422-daa9-4de3-a45f-a3f84d15e72b>
– Channel SIP/9012-00000000 joined ‘simple_bridge’ basic-bridge <83a14422-daa9-4de3-a45f-a3f84d15e72b>
> 0x184a230 – Probation passed - setting RTP source address to 10.192.230.231:18462
Retransmitting #1 (no NAT) to 148.69.11.122:38022:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1396079092 1396079092 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 19982 RTP/AVP 8 0 18 3 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Retransmitting #8 (no NAT) to 62.210.149.114:5139:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 62.210.149.114:5139;branch=z9hG4bK-e185467fac146e0d30cd307d54d7d161;received=62.210.149.114;rport=5139
From: 8009sip:8009@81.90.54.197;tag=1334cac9
To: 00441865679904sip:00441865679904@81.90.54.197;tag=as09051958
Call-ID: e185467fac146e0d30cd307d54d7d161
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“Jar209Reftel.refertelecom.pt”, nonce="599d1eab"
Content-Length: 0

Retransmitting #2 (no NAT) to 148.69.11.122:38022:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1396079092 1396079092 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 19982 RTP/AVP 8 0 18 3 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<— SIP read from UDP:10.192.230.231:5060 —>
BYE sip:9012@10.192.124.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.192.230.231:5060;branch=z9hG4bK6e047190
Max-Forwards: 70
From: sip:24067@10.192.230.231;tag=as329e86b2
To: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.10.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.192.230.231:5060 (no NAT)
Scheduling destruction of SIP dialog ‘49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 10.192.230.231:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.230.231:5060;branch=z9hG4bK6e047190;received=10.192.230.231
From: sip:24067@10.192.230.231;tag=as329e86b2
To: “Teste_IPT_2” sip:9012@10.192.124.101;tag=as5f74c4f3
Call-ID: 49937c836fc9b0f2244d2df545c6607e@10.192.124.101:5060
CSeq: 102 BYE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/TRUNKSIP-VIRIATO-00000001 left ‘simple_bridge’ basic-bridge <83a14422-daa9-4de3-a45f-a3f84d15e72b>
– Channel SIP/9012-00000000 left ‘simple_bridge’ basic-bridge <83a14422-daa9-4de3-a45f-a3f84d15e72b>
== Spawn extension (and_int, 24067, 2) exited non-zero on 'SIP/9012-00000000’
Scheduling destruction of SIP dialog ‘BmY052N6TS’ in 32000 ms (Method: INVITE)
[Jan 19 09:33:37] ERROR[17095]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (1045) Access denied for user ‘root’@‘localhost’ (using password: YES)

<— SIP read from UDP:62.210.143.116:5095 —>
REGISTER sip:81.90.54.197:5060 SIP/2.0
Via: SIP/2.0/UDP 62.210.143.116:5095;branch=z9hG4bK30569f9c967b29675a3f745d;rport
From: “10” sip:10@81.90.54.197:5060;tag=30569f9cbfe2
To: “10” sip:10@81.90.54.197:5060
Call-ID: 9f9c967b-7972967-5a3f745d@81.90.54.197
CSeq: 1 REGISTER
Contact: “10” sip:10@62.210.143.116:5095
User-Agent: VaxSIPUserAgent/3.1
Expires: 1800
Max-Forwards: 70
Content-Length: 0

Retransmitting #4 (no NAT) to 148.69.11.122:38022:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.117.119.172:42502;branch=z9hG4bK.FaXIvZIUN;received=148.69.11.122;rport=38022
From: sip:9012@81.90.54.197;tag=3zA~CSYSU
To: sip:24067@81.90.54.197;tag=as47eb21b7
Call-ID: BmY052N6TS
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:24067@10.192.124.101:5060
Content-Type: application/sdp
Content-Length: 350

v=0
o=root 1396079092 1396079092 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 19982 RTP/AVP 8 0 18 3 102
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

Reliably Transmitting (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK55fec2b5
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.192.124.101;tag=as38102481
To: sip:10.192.231.231
Contact: sip:asterisk@10.192.124.101:5060
Call-ID: 590a5be0328f0794192c226d65aaf12b@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 09:33:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

3

Although your configuration is complex, and I haven’t taken in all the detail, it seems to me that you have two disjoint private networks, and need to set directmedia to no (and remove canreinvite, as it is just a different name for the same thing).

Also do not allow anything other than mulaw or alaw if you are forced to use dtmfmode=inband, as, in particular, DTMF will not work over a GSM codec (cellular networks send it out of band), and reconstruct it in the base station.

4

Thank you very much for your attention. The problem of missing audio persists.
The current settings:

Viriato:

[TRUNKSIP-SAGRES]
type=peer
host=10.192.124.101
context=incoming-tlm
;disallow=all
allow = all
dtmfmode=inband
directmedia=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.124.101/255.255.255.255

Sagres:

[TRUNKSIP-VIRIATO]
type=peer
host=10.192.230.231
context=incoming-iax
;disallow=all
allow = all
dtmfmode=inband
directmedia=no
qualify=yes
nat=no
deny=0.0.0.0/0.0.0.0
permit=10.192.230.231/255.255.255.255

[9012]
type=friend
callerid=“Teste_IPT_2” <9012>
context=and_int
secret=#$%1290#$gh
host=dynamic
dtmfmode=inband
directmedia=no
defaultuser=9012
progressinband=no
promiscredir=yes
qualify=no
;deny=0.0.0.0/0.0.0.0
;permit=10.168.242.100/255.255.255.255

5

Please provide just the INVITE dialogues with the current configuration. Please use the forum editing tools to mark the logs as pre-formatted text (</> button at top of edit window).

6 
sagres*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:148.69.14.203:57370 --->
INVITE sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.w1OF~VfVG;rport
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197
CSeq: 20 INVITE
Call-ID: VMQkvsTivf
Max-Forwards: 70
Route: <sip:voip-lx.cloudsolutions.pt;transport=udp;lr>
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 547
Contact: <sip:9012@148.69.14.203:57370;transport=udp>;+sip.instance="<urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38>"
User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)

v=0
o=9012 1094 3417 IN IP4 10.196.199.180
s=Talk
c=IN IP4 10.196.199.180
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (14 headers 20 lines) ---
Sending to 148.69.14.203:57370 (no NAT)
Sending to 148.69.14.203:57370 (no NAT)
Using INVITE request as basis request - VMQkvsTivf
Found peer '9012' for '9012' from 148.69.14.203:57370

<--- Reliably Transmitting (no NAT) to 148.69.14.203:57370 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.w1OF~VfVG;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as0d347f9c
Call-ID: VMQkvsTivf
CSeq: 20 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="Jar209Reftel.refertelecom.pt", nonce="54eb1bd8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'VMQkvsTivf' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:148.69.14.203:57370 --->
INVITE sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.w1OF~VfVG;rport
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197
CSeq: 20 INVITE
Call-ID: VMQkvsTivf
Max-Forwards: 70
Route: <sip:voip-lx.cloudsolutions.pt;transport=udp;lr>
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 547
Contact: <sip:9012@148.69.14.203:57370;transport=udp>;+sip.instance="<urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38>"
User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)

v=0
o=9012 1094 3417 IN IP4 10.196.199.180
s=Talk
c=IN IP4 10.196.199.180
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (14 headers 20 lines) ---
Ignoring this INVITE request

<--- SIP read from UDP:148.69.14.203:57370 --->
ACK sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.w1OF~VfVG;rport
Call-ID: VMQkvsTivf
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: <sip:24067@81.90.54.197>;tag=as0d347f9c
Contact: <sip:9012@148.69.14.203:57370;transport=udp>;+sip.instance="<urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38>"
Route: <sip:voip-lx.cloudsolutions.pt;transport=udp;lr>
Max-Forwards: 70
CSeq: 20 ACK

<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (no NAT) to 10.192.207.56:5060:
OPTIONS sip:10.192.207.56 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK54446a30
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.192.124.101>;tag=as525d44f9
To: <sip:10.192.207.56>
Contact: <sip:asterisk@10.192.124.101:5060>
Call-ID: 103034fa7723b6ac00fb7cdc3e6f9fbc@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 12:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:148.69.14.203:57370 --->
INVITE sip:24067@81.90.54.197 SIP/2.0
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;rport
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197
CSeq: 21 INVITE
Call-ID: VMQkvsTivf
Max-Forwards: 70
Route: <sip:voip-lx.cloudsolutions.pt;transport=udp;lr>
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 547
Contact: <sip:9012@148.69.14.203:57370;transport=udp>;+sip.instance="<urn:uuid:994e9f3e-30d8-43a3-adf8-92831073dc38>"
User-Agent: LinphoneAndroid/3.3.2 (belle-sip/1.6.3)
Authorization: Digest realm="Jar209Reftel.refertelecom.pt", nonce="54eb1bd8", algorithm=MD5, username="9012", uri="sip:24067@81.90.54.197", response="b03f212093243300839a9de11e496c40"

v=0
o=9012 1094 3417 IN IP4 10.196.199.180
s=Talk
c=IN IP4 10.196.199.180
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 96 97 98 0 8 3 9 99 18 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 speex/16000
a=fmtp:97 vbr=on
a=rtpmap:98 speex/8000
a=fmtp:98 vbr=on
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
a=rtcp-fb:* ccm tmmbr
<------------->
--- (15 headers 20 lines) ---
Sending to 148.69.14.203:57370 (no NAT)
Using INVITE request as basis request - VMQkvsTivf
Found peer '9012' for '9012' from 148.69.14.203:57370
  == Using SIP RTP CoS mark 5
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 99
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 100
Found RTP audio format 102
Found audio description format opus for ID 96
Found audio description format speex for ID 97
Found audio description format speex for ID 98
Found audio description format iLBC for ID 99
Found unknown media description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 100
Found audio description format telephone-event for ID 102
Capabilities: us - (alaw|ulaw|g729|gsm), peer - audio=(ulaw|gsm|alaw|g722|g729|opus|speex16|speex|ilbc)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729|gsm)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 10.196.199.180:7076
Looking for 24067 in and_int (domain 81.90.54.197)
sip_route_dump: route/path hop: <sip:9012@148.69.14.203:57370;transport=udp>

<--- Transmitting (no NAT) to 148.69.14.203:57370 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Length: 0


<------------>
    -- Executing [24067@and_int:1] Ringing("SIP/9012-00000002", "") in new stack

<--- Transmitting (no NAT) to 148.69.14.203:57370 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Length: 0


<------------>
    -- Executing [24067@and_int:2] Dial("SIP/9012-00000002", "SIP/TRUNKSIP-VIRIATO/24067,180,tT") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
We think we can do text
Audio is at 17320
Video is at 10.192.124.101:15066
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec gsm to SDP
Adding codec g723 to SDP
Adding codec g726 to SDP
Adding codec g726aal2 to SDP
Adding codec adpcm to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec slin to SDP
Adding codec lpc10 to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec speex to SDP
Adding codec ilbc to SDP
Adding codec g722 to SDP
Adding codec siren7 to SDP
Adding codec siren14 to SDP
Adding codec testlaw to SDP
Adding codec g719 to SDP
Adding codec opus to SDP
Adding video codec h261 to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding video codec mpeg4 to SDP
Adding video codec vp8 to SDP
Reliably Transmitting (no NAT) to 10.192.230.231:5060:
INVITE sip:24067@10.192.230.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK63355de1
Max-Forwards: 70
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>
Contact: <sip:9012@10.192.124.101:5060>
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 12:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1095

v=0
o=root 1141843004 1141843004 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
b=CT:384
t=0 0
m=audio 17320 RTP/AVP 8 0 18 3 4 111 112 5 10 118 7 110 117 119 97 9 102 115 116 107
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=maxptime:20
a=sendrecv
m=video 15066 RTP/AVP 31 34 98 99 104 100
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

---
    -- Called SIP/TRUNKSIP-VIRIATO/24067

<--- SIP read from UDP:10.192.230.231:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK63355de1;received=10.192.124.101
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:24067@10.192.230.231:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:10.192.230.231:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK63355de1;received=10.192.124.101
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:24067@10.192.230.231:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:24067@10.192.230.231:5060>
    -- SIP/TRUNKSIP-VIRIATO-00000003 is ringing

<--- Transmitting (no NAT) to 148.69.14.203:57370 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Length: 0


<--- SIP read from UDP:10.192.230.231:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK63355de1;received=10.192.124.101
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:24067@10.192.230.231:5060>
Content-Length: 0

<--- SIP read from UDP:10.192.230.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK63355de1;received=10.192.124.101
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:24067@10.192.230.231:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 1271

v=0
o=root 223866265 223866265 IN IP4 10.192.230.231
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.230.231
b=CT:384
t=0 0
m=audio 10282 RTP/AVP 8 0 18 3 4 111 112 5 10 118 7 110 117 119 97 9 102 115 116 107
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:111 G726-32/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:117 speex/16000
a=rtpmap:119 speex/32000
a=rtpmap:97 iLBC/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=maxptime:20
a=sendrecv
m=video 18180 RTP/AVP 31 34 98 99 104 100
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=0;QCIF=0;CIF=0;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 SQCIF=0;QCIF=0;CIF=0;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=rtpmap:104 MP4V-ES/90000
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv
<------------->
--- (14 headers 46 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 5
Found RTP audio format 10
Found RTP audio format 118
Found RTP audio format 7
Found RTP audio format 110
Found RTP audio format 117
Found RTP audio format 119
Found RTP audio format 97
Found RTP audio format 9
Found RTP audio format 102
Found RTP audio format 115
Found RTP audio format 116
Found RTP audio format 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format GSM for ID 3
Found audio description format G723 for ID 4
Found audio description format G726-32 for ID 111
Found audio description format AAL2-G726-32 for ID 112
Found audio description format DVI4 for ID 5
Found audio description format L16 for ID 10
Found audio description format L16 for ID 118
Found audio description format LPC for ID 7
Found audio description format speex for ID 110
Found audio description format speex for ID 117
Found audio description format speex for ID 119
Found audio description format iLBC for ID 97
Found audio description format G722 for ID 9
Found audio description format G7221 for ID 102
Found audio description format G7221 for ID 115
Found audio description format G719 for ID 116
Found audio description format opus for ID 107
Found RTP video format 31
Found RTP video format 34
Found RTP video format 98
Found RTP video format 99
Found RTP video format 104
Found RTP video format 100
Found video description format H261 for ID 31
Found video description format H263 for ID 34
Found video description format h263-1998 for ID 98
Found video description format H264 for ID 99
Found video description format MP4V-ES for ID 104
Found video description format VP8 for ID 100
Capabilities: us - (alaw|ulaw|g729|gsm|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|red|t140), peer - audio=(ulaw|gsm|g723|adpcm|lpc10|alaw|g722|slin|g729|ilbc|siren7|opus|speex|g726|g726aal2|siren14|g719|speex16|slin16|speex32)/video=(h261|h263|h263p|h264|vp8|mpeg4)/text=(nothing), combined - (alaw|ulaw|g729|gsm|g723|g726|g726aal2|adpcm|slin|slin|lpc10|speex|speex|speex|ilbc|g722|siren7|siren14|g719|opus|h261|h263|h263p|h264|mpeg4|vp8)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.192.230.231:10282
Peer video RTP is at port 10.192.230.231:18180
sip_route_dump: route/path hop: <sip:24067@10.192.230.231:5060>
set_destination: Parsing <sip:24067@10.192.230.231:5060> for address/port to send to
set_destination: set destination to 10.192.230.231:5060
Transmitting (no NAT) to 10.192.230.231:5060:
ACK sip:24067@10.192.230.231:5060 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK26659782
Max-Forwards: 70
From: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
To: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
Contact: <sip:9012@10.192.124.101:5060>
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0


---
    -- SIP/TRUNKSIP-VIRIATO-00000003 answered SIP/9012-00000002
Audio is at 13840
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding codec gsm to SDP

<--- Reliably Transmitting (no NAT) to 148.69.14.203:57370 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 181375234 181375234 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 13840 RTP/AVP 8 0 18 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
    -- Channel SIP/TRUNKSIP-VIRIATO-00000003 joined 'simple_bridge' basic-bridge <7e1ffe5b-b919-468a-b0af-8c81c69a17e4>
    -- Channel SIP/9012-00000002 joined 'simple_bridge' basic-bridge <7e1ffe5b-b919-468a-b0af-8c81c69a17e4>
Retransmitting #2 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK6dd9471c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.192.124.101>;tag=as2fcd6ee8
To: <sip:10.192.231.231>
Contact: <sip:asterisk@10.192.124.101:5060>
Call-ID: 34bfcc1a26f053264137fc4077d7f0ba@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 12:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
       > 0x7f543400b1a0 -- Probation passed - setting RTP source address to 10.192.230.231:10282
Retransmitting #1 (no NAT) to 148.69.14.203:57370:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 181375234 181375234 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 13840 RTP/AVP 8 0 18 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

---
Retransmitting #3 (no NAT) to 62.210.149.114:5115:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 62.210.149.114:5115;branch=z9hG4bK-84aab4107e63ccdc6909f9ccf5ed56df;received=62.210.149.114;rport=5115
From: 9701<sip:9701@81.90.54.197>;tag=dac896a1
To: 00441865679904<sip:00441865679904@81.90.54.197>;tag=as397bbaec
Call-ID: 84aab4107e63ccdc6909f9ccf5ed56df
CSeq: 1 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="Jar209Reftel.refertelecom.pt", nonce="7588c248"
Content-Length: 0

---
Retransmitting #3 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK6dd9471c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.192.124.101>;tag=as2fcd6ee8
To: <sip:10.192.231.231>
Contact: <sip:asterisk@10.192.124.101:5060>
Call-ID: 34bfcc1a26f053264137fc4077d7f0ba@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 12:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog 'a03e2b00-aca307e-5a3f9dbb@81.90.54.197' Method: REGISTER
Retransmitting #2 (no NAT) to 148.69.14.203:57370:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 181375234 181375234 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 13840 RTP/AVP 8 0 18 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv

Really destroying SIP dialog '103034fa7723b6ac00fb7cdc3e6f9fbc@10.192.124.101:5060' Method: OPTIONS

<--- SIP read from UDP:10.192.230.231:5060 --->
BYE sip:9012@10.192.124.101:5060 SIP/2.0
Via: SIP/2.0/UDP 10.192.230.231:5060;branch=z9hG4bK4b3818ef
Max-Forwards: 70
From: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
To: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.10.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.192.230.231:5060 (no NAT)
Scheduling destruction of SIP dialog '7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.192.230.231:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.192.230.231:5060;branch=z9hG4bK4b3818ef;received=10.192.230.231
From: <sip:24067@10.192.230.231>;tag=as5fd2c8ab
To: "Teste_IPT_2" <sip:9012@10.192.124.101>;tag=as513c3443
Call-ID: 7bdaf6dd2d0f8858141b32996dc157d3@10.192.124.101:5060
CSeq: 102 BYE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #4 (no NAT) to 10.192.231.231:5060:
OPTIONS sip:10.192.231.231 SIP/2.0
Via: SIP/2.0/UDP 10.192.124.101:5060;branch=z9hG4bK6dd9471c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.192.124.101>;tag=as2fcd6ee8
To: <sip:10.192.231.231>
Contact: <sip:asterisk@10.192.124.101:5060>
Call-ID: 34bfcc1a26f053264137fc4077d7f0ba@10.192.124.101:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.10.0
Date: Fri, 19 Jan 2018 12:30:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Really destroying SIP dialog '34bfcc1a26f053264137fc4077d7f0ba@10.192.124.101:5060' Method: OPTIONS
Really destroying SIP dialog 'a03e2ea8-ae93081-5a3f9dbc@81.90.54.197' Method: REGISTER
Retransmitting #3 (no NAT) to 148.69.14.203:57370:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.196.199.180:42502;branch=z9hG4bK.bXsc5dxOC;received=148.69.14.203;rport=57370
From: <sip:9012@81.90.54.197>;tag=n5UwM5FrB
To: sip:24067@81.90.54.197;tag=as5fc8fa1a
Call-ID: VMQkvsTivf
CSeq: 21 INVITE
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:24067@10.192.124.101:5060>
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 181375234 181375234 IN IP4 10.192.124.101
s=Asterisk PBX 13.10.0
c=IN IP4 10.192.124.101
t=0 0
m=audio 13840 RTP/AVP 8 0 18 3
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=ptime:20
a=maxptime:150
a=sendrecv
7

The requestor appears to be on a public IP address, but has told Asterisk to send media to a private one. I believe nat=comedia is a work around for this, but it can end in a Mexican standoff, as one side needs to send media without first receiving it. Really, though, the peer needs to be made aware that it is behind NAT, and to provide its public address for media.

I don’t understand how Asterisk is accepting the call, as the source address is not a known peer and allowguest=no!

Asterisk seems to have sent its private network address as media address to the requestor. That’s because you haven’t told it about its public address.
.

8

Hi,

So, I’ve changed to the following:

Sagres:

[general]
realm=Jar209Reftel.refertelecom.pt
context=default
allowoverlap=yes
srvlookup=no
defaultexpiry=90
;qualify=8000
videosupport=yes
notifyringing = yes
notifyhold = yes
callcounter = yes
counteronpeer = yes
alwaysauthreject = yes
allowguest=no
disallow=all
allow = alaw
allow = ulaw
;allow = g722
allow = g729
allow = gsm
rtptimeout=300
bindaddr=0.0.0.0
bindport=5060
tos=0xb8
externip=81.90.54.197 

[TRUNKSIP-VIRIATO]
type=peer
host=10.192.230.231
context=incoming-iax
;disallow=all
allow = all
dtmfmode=inband
directmedia=no
qualify=yes
nat=comedia
deny=0.0.0.0/0.0.0.0
permit=10.192.230.231/255.255.255.255

It is supposed to work with these changes? (It didn’t work)

Yes, you’re right, the asterisk must reject the calll…

I’m confused, I would like to get this service up and running …

9

Although it is not your problem, do not allow g729 or gsm, if you have dtmfmode=inband, as DTMF will not work well.

10

Yes, I’ll correct Thank you. But how can I even solve this problem …?

11

There seem to be two problems:

  1. Why is the call not failing to match on sip.conf. That would typically only happen if you spoofed the number of a local device, which is why it is usually best to use type=peer on local devices.

  2. Why is the audio not routing. For that you need to get new logs and confirm that you are sending the right media address.

You should also enable rtp debugging to see if you are receiving RTP, and from where.

Note that comedia will not work if the peer doesn’t send anything to you, or it sends it to the wrong port number, as Asterisk needs to learn the peer media address from the incoming media.

12

Regarding point 1, what do you indicate is that since I am accessing from outside with an internal number (9012), the call is accepted? But in fact I do not want the call to be rejected!

As for point 2 send below the RTP logs:


sagres*CLI> rtp set debug on
RTP Debugging Enabled
  == Using SIP RTP CoS mark 5
    -- Executing [24067@and_int:1] Ringing("SIP/9012-00000000", "") in new stack
    -- Executing [24067@and_int:2] Dial("SIP/9012-00000000", "SIP/TRUNKSIP-VIRIATO/24067,180,tT") in new stack
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
    -- Called SIP/TRUNKSIP-VIRIATO/24067
    -- SIP/TRUNKSIP-VIRIATO-00000001 is ringing
    -- SIP/TRUNKSIP-VIRIATO-00000001 is ringing
[Jan 19 15:06:47] WARNING[1455]: chan_sip.c:4118 retrans_pkt: Timeout on 1566064498-395189467-334858515 on non-critical invite transaction.
    -- SIP/TRUNKSIP-VIRIATO-00000001 answered SIP/9012-00000000
    -- Channel SIP/TRUNKSIP-VIRIATO-00000001 joined 'simple_bridge' basic-bridge <5bc0dca6-32fa-402f-8995-b73e3a30f124>
    -- Channel SIP/9012-00000000 joined 'simple_bridge' basic-bridge <5bc0dca6-32fa-402f-8995-b73e3a30f124>
       > 0x2bf4cb0 -- Probation passed - setting RTP source address to 10.192.230.231:16128
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016462, ts 000000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016463, ts 000160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016464, ts 000320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016465, ts 000480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016466, ts 000640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016467, ts 000800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016468, ts 000960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016469, ts 001120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016470, ts 001280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016471, ts 001440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016472, ts 001600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016473, ts 001760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016474, ts 001920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016475, ts 002080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016476, ts 002240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016477, ts 002400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016478, ts 002560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016479, ts 002720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016480, ts 002880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016481, ts 003040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016482, ts 003200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016483, ts 003360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016484, ts 003520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016485, ts 003680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016486, ts 003840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016487, ts 004000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016488, ts 004160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016489, ts 004320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016490, ts 004480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016491, ts 004640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016492, ts 004800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016493, ts 004960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016494, ts 005120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016495, ts 005280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016496, ts 005440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016497, ts 005600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016498, ts 005760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016499, ts 005920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016500, ts 006080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016501, ts 006240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016502, ts 006400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016503, ts 006560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016504, ts 006720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016505, ts 006880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016506, ts 007040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016507, ts 007200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016508, ts 007360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016509, ts 007520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016510, ts 007680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016511, ts 007840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016512, ts 008000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016513, ts 008160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016514, ts 008320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016515, ts 008480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016516, ts 008640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016517, ts 008800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016518, ts 008960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016519, ts 009120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016520, ts 009280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016521, ts 009440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016522, ts 009600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016523, ts 009760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016524, ts 009920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016525, ts 010080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016526, ts 010240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016527, ts 010400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016528, ts 010560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016529, ts 010720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016530, ts 010880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016531, ts 011040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016532, ts 011200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016533, ts 011360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016534, ts 011520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016535, ts 011680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016536, ts 011840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016537, ts 012000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016538, ts 012160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016539, ts 012320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016540, ts 012480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016541, ts 012640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016542, ts 012800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016543, ts 012960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016544, ts 013120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016545, ts 013280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016546, ts 013440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016547, ts 013600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016548, ts 013760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016549, ts 013920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016550, ts 014080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016551, ts 014240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016552, ts 014400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016553, ts 014560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016554, ts 014720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016555, ts 014880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016556, ts 015040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016557, ts 015200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016558, ts 015360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016559, ts 015520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016560, ts 015680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016561, ts 015840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016562, ts 016000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016563, ts 016160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016564, ts 016320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016565, ts 016480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016566, ts 016640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016567, ts 016800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016568, ts 016960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016569, ts 017120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016570, ts 017280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016571, ts 017440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016572, ts 017600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016573, ts 017760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016574, ts 017920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016575, ts 018080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016576, ts 018240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016577, ts 018400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016578, ts 018560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016579, ts 018720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016580, ts 018880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016581, ts 019040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016582, ts 019200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016583, ts 019360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016584, ts 019520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016585, ts 019680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016586, ts 019840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016587, ts 020000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016588, ts 020160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016589, ts 020320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016590, ts 020480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016591, ts 020640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016592, ts 020800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016593, ts 020960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016594, ts 021120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016595, ts 021280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016596, ts 021440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016597, ts 021600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016598, ts 021760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016599, ts 021920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016600, ts 022080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016601, ts 022240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016602, ts 022400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016603, ts 022560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016604, ts 022720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016605, ts 022880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016606, ts 023040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016607, ts 023200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016608, ts 023360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016609, ts 023520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016610, ts 023680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016611, ts 023840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016612, ts 024000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016613, ts 024160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016614, ts 024320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016615, ts 024480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016616, ts 024640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016617, ts 024800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016618, ts 024960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016619, ts 025120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016620, ts 025280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016621, ts 025440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016622, ts 025600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016623, ts 025760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016624, ts 025920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016625, ts 026080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016626, ts 026240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016627, ts 026400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016628, ts 026560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016629, ts 026720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016630, ts 026880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016631, ts 027040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016632, ts 027200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016633, ts 027360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016634, ts 027520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016635, ts 027680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016636, ts 027840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016637, ts 028000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016638, ts 028160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016639, ts 028320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016640, ts 028480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016641, ts 028640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016642, ts 028800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016643, ts 028960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016644, ts 029120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016645, ts 029280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016646, ts 029440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016647, ts 029600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016648, ts 029760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016649, ts 029920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016650, ts 030080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016651, ts 030240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016652, ts 030400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016653, ts 030560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016654, ts 030720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016655, ts 030880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016656, ts 031040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016657, ts 031200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016658, ts 031360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016659, ts 031520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016660, ts 031680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016661, ts 031840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016662, ts 032000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016663, ts 032160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016664, ts 032320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016665, ts 032480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016666, ts 032640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016667, ts 032800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016668, ts 032960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016669, ts 033120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016670, ts 033280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016671, ts 033440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016672, ts 033600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016673, ts 033760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016674, ts 033920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016675, ts 034080, len 000160)
[Jan 19 15:06:55] NOTICE[1455]: chan_sip.c:28468 handle_request_register: Registration from '"8000" <sip:8000@81.90.54.197:5060>' failed for '62.210.143.116:5085' - Wrong password
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016676, ts 034240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016677, ts 034400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016678, ts 034560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016679, ts 034720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016680, ts 034880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016681, ts 035040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016682, ts 035200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016683, ts 035360, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016684, ts 035520, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016685, ts 035680, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016686, ts 035840, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016687, ts 036000, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016688, ts 036160, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016689, ts 036320, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016690, ts 036480, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016691, ts 036640, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016692, ts 036800, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016693, ts 036960, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016694, ts 037120, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016695, ts 037280, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016696, ts 037440, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016697, ts 037600, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016698, ts 037760, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016699, ts 037920, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016700, ts 038080, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016701, ts 038240, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016702, ts 038400, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016703, ts 038560, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016704, ts 038720, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016705, ts 038880, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016706, ts 039040, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016707, ts 039200, len 000160)
Got  RTP packet from    10.192.230.231:16128 (type 08, seq 016708, ts 039360, len 000160)
    -- Channel SIP/TRUNKSIP-VIRIATO-00000001 left 'simple_bridge' basic-bridge <5bc0dca6-32fa-402f-8995-b73e3a30f124>
    -- Channel SIP/9012-00000000 left 'simple_bridge' basic-bridge <5bc0dca6-32fa-402f-8995-b73e3a30f124>
  == Spawn extension (and_int, 24067, 2) exited non-zero on 'SIP/9012-00000000'
[Jan 19 15:06:55] ERROR[1440]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (1045) Access denied for user 'root'@'localhost' (using password: YES)
sagres*CLI> rtp set debug off
13

Looks like you have never received any media from the side that is not 10.192.230.231, so there is nothing to forward from that side, and the comedia cannot be trained to know how to send to it.

What c= line are you sending to the one that is not 10.192.230.231. Is that address routable from it and not blocked by firewalls?

I see that you have a local device outside the NAT, and it is a genuine match. However, you should not need type=friend, as Asterisk will learn the IP address from the Register, and use that.

14

As if fw were barring RTP to the Sagres server right? Soon the problem is in FW and not in the Sagres server. But my colleague of networks says he lets it all go … I do not understand.

What do you mean by “What c= line are you sending to the one that is not 10.192.230.231. Is that address routable from it and not blocked by firewalls”. What addresss?

But the type is already as a “friend”…

[9012]
type=friend
callerid="Teste_IPT_2" <9012>
context=and_int
15

peer is safer than friend.

You have SDP being sent to two peers. the peer with the IP address I quote seems to be sending to the right address, so you need to look at the other peer. That peer will send to the address in the c= line that Asterisk sends to it. Is Asterisk now sending a public address there? The actual media line contains a port number. Will media sent to that port number on the public address actually reach Asterisk?

16

I’ve just forward this information to my network colleague. Tks.

17

Hi,

It finally works!

The magical configuration:

externip=xx.xx.xx.xx
nat=yes
localnet=10.192.250.0/255.255.255.0