No audio, no nat, no firewall denies logged

Asterisk Asterisk SIP
1

I’m building a connection from 2 asterisk servers.

Server 1 at 10.123.240.144
[Siptel01]
type=peer
context=outbound
directmedia=no
qualify=yes
host=172.24.220.18
disallow=all
allow=g729
allow=ulaw
#nat=force_rport,comedia
nat=no

server 2 @ 172.24.220.18
[optel35]
type=peer
context=gateways-inbound
directmedia=no
qualify=yes
host=10.123.240.144
;insecure=port,invite
;allowexternaldomains=yes
nat=no
disallow=all
;allow=g729
allow=ulaw

I have no audio either direction. No NAT is involved.
on 10.123.244.144
With rtp debug set ip 10.123.240.144 I get tones of output
RTP Debugging Enabled for address: 172.24.220.18:0
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019029, ts 000000, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019030, ts 000160, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019031, ts 000320, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019032, ts 000480, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019033, ts 000640, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019034, ts 000800, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019035, ts 000960, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019036, ts 001120, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019037, ts 001280, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019038, ts 001440, len 000160)
Sent RTP packet to 172.24.220.18:18494 (type 00, seq 019039, ts 001600, len 000160)

on the otherside I get nothing. I have it set to rtp set debug ip 10.123.240.144

I’m passing a call from 10.123.240.144 to 172.24.220.18

2

using standard ports 10-20k for rtp

3

You would need to provide the output of “sip set debug on” and a call attempt to show what is being negotiated. As it is, it appears that Asterisk is at least sending audio to the target that was provided by the remote side.

4

on teh 172.24.220.18 side

<— SIP read from UDP:10.123.240.144:5060 —>
INVITE sip:+15719188345@Siptel01.multiservice.com SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK4378c554
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com
Contact: sip:+18166121399@10.123.240.144:5060
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:34:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 992238638 992238638 IN IP4 10.123.240.144
s=Asterisk PBX 16.1.0
c=IN IP4 54.209.146.133
t=0 0
m=audio 15346 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------->
— (14 headers 13 lines) —
Sending to 10.123.240.144:5060 (no NAT)
Sending to 10.123.240.144:5060 (no NAT)
Using INVITE request as basis request - 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
Found peer ‘optel35’ for ‘+18166121399’ from 10.123.240.144:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(gsm|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 54.209.146.133:15346
Looking for +15719188345 in gateways-inbound (domain Siptel01.multiservice.com)
list_route: hop: sip:+18166121399@10.123.240.144:5060

<— Transmitting (no NAT) to 10.123.240.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK4378c554;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Length: 0

<------------>
– Executing [+15719188345@gateways-inbound:1] NoOp(“SIP/optel35-000b6f63”, “Level# AWS testNumber”) in new stack
– Executing [+15719188345@gateways-inbound:2] Dial(“SIP/optel35-000b6f63”, “Local/32897@stations”) in new stack
– Called Local/32897@stations
– Executing [32897@stations:1] Set(“Local/32897@stations-00048f55;2”, “HASH(station,context)=stations”) in new stack
– Executing [32897@stations:2] Set(“Local/32897@stations-00048f55;2”, “HASH(station,localexten)=32897”) in new stack
– Executing [32897@stations:3] Set(“Local/32897@stations-00048f55;2”, “HASH(station,extstate)=NOT_INUSE”) in new stack
– Executing [32897@stations:4] Set(“Local/32897@stations-00048f55;2”, “HASH(station,hasmailbox)=1”) in new stack
– Executing [32897@stations:5] Set(“Local/32897@stations-00048f55;2”, “extinfo=SIP/Daniel-Siemens&SIP/dwsiemens~20~NONE”) in new stack
– Executing [32897@stations:6] Set(“Local/32897@stations-00048f55;2”, “HASH(station,devicestring)=SIP/Daniel-Siemens&SIP/dwsiemens”) in new stack
– Executing [32897@stations:7] Set(“Local/32897@stations-00048f55;2”, “HASH(station,timeout)=20”) in new stack
– Executing [32897@stations:8] Set(“Local/32897@stations-00048f55;2”, “HASH(station,mobile)=NONE”) in new stack
– Executing [32897@stations:9] GotoIf(“Local/32897@stations-00048f55;2”, “1?entry:setcid”) in new stack
– Goto (stations,32897,11)
– Executing [32897@stations:11] GotoIf(“Local/32897@stations-00048f55;2”, “0?mailboxonly:valid”) in new stack
– Goto (stations,32897,16)
– Executing [32897@stations:16] GotoIf(“Local/32897@stations-00048f55;2”, “1?exten-route:exten-oper”) in new stack
– Goto (stations,32897,17)
– Executing [32897@stations:17] GotoIf(“Local/32897@stations-00048f55;2”, “1?exten-vmail:exten-mobile”) in new stack
– Goto (stations,32897,18)
– Executing [32897@stations:18] Dial(“Local/32897@stations-00048f55;2”, “SIP/Daniel-Siemens&SIP/dwsiemens,20,kKtT”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/Daniel-Siemens
– Called SIP/dwsiemens
– SIP/dwsiemens-000b6f65 connected line has changed. Saving it until answer for Local/32897@stations-00048f55;2
– SIP/Daniel-Siemens-000b6f64 connected line has changed. Saving it until answer for Local/32897@stations-00048f55;2
– SIP/Daniel-Siemens-000b6f64 is ringing
– Local/32897@stations-00048f55;1 is ringing

<— Transmitting (no NAT) to 10.123.240.144:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK4378c554;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com;tag=as6a126078
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Length: 0

<------------>
– SIP/dwsiemens-000b6f65 is ringing
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-
– SIP/Daniel-Siemens-000b6f64 connected line has changed. Saving it until answer for Local/32897@stations-00048f55;2
– SIP/Daniel-Siemens-000b6f64 answered Local/32897@stations-00048f55;2
– Local/32897@stations-00048f55;1 answered SIP/optel35-000b6f63
Audio is at 15144
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 10.123.240.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK4378c554;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com;tag=as6a126078
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Type: application/sdp
Require: timer
Content-Length: 238

v=0
o=root 1041162809 1041162809 IN IP4 172.24.220.18
s=Asterisk PBX 11.25.1
c=IN IP4 172.24.220.18
t=0 0
m=audio 15144 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

<— SIP read from UDP:10.123.240.144:5060 —>
ACK sip:+15719188345@172.24.220.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK03d41430
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com;tag=as6a126078
Contact: sip:+18166121399@10.123.240.144:5060
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.1.0
Content-Length: 0

<------------->
— (10 headers 0 lines) —
– Locally bridging SIP/optel35-000b6f63 and SIP/Daniel-Siemens-000b6f64
== Spawn extension (stations, 32897, 18) exited non-zero on ‘Local/32897@stations-00048f55;2’
– SIP/lwmgw23-000b6f69 is making progress passing it to SIP/klstates-000b6f68
[Jan 7 15:34:40] NOTICE[18796][C-0005bd3c]: res_rtp_asterisk.c:4519 ast_rtp_read: Unknown RTP codec 95 received from ‘10.119.100.17:8000’
== Spawn extension (stations, 998328029860, 5) exited non-zero on ‘SIP/Donna-Hammond-000b6f59’
– SIP/opmgw23-000b6f62 answered SIP/Nicole-Chavez-000b6f61
– Remote UNIX connection
– Remote UNIX connection disconnected

<— SIP read from UDP:10.123.240.144:5060 —>
BYE sip:+15719188345@172.24.220.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK503a4a57
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com;tag=as6a126078
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 10.123.240.144:5060 (no NAT)
Scheduling destruction of SIP dialog ‘744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060’ in 6400 ms (Method: BYE)

<— Transmitting (no NAT) to 10.123.240.144:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK503a4a57;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as5df2d3c2
To: sip:+15719188345@Siptel01.multiservice.com;tag=as6a126078
Call-ID: 744775e448270fd65f8bf6b527d4aa57@10.123.240.144:5060
CSeq: 103 BYE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

5

Content-Length: 0 I think the buffer may be a bit short but it looks like its their.

— (13 headers 0 lines) —
Sending to 172.24.220.18:5060 (no NAT)
Looking for s in public (domain 10.123.240.144)

<— Transmitting (no NAT) to 172.24.220.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.24.220.18:5060;branch=z9hG4bK42a87dcc;received=172.24.220.18
From: “asterisk” sip:asterisk@172.24.220.18;tag=as368af213
To: sip:10.123.240.144;tag=as1d07bdf9
Call-ID: 32f0baee4446d191085cee497889fe60@172.24.220.18:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 16.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:10.123.240.144:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘32f0baee4446d191085cee497889fe60@172.24.220.18:5060’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 172.25.220.18:5060:
OPTIONS sip:172.25.220.18 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK27d7a413
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.123.240.144;tag=as2dce5b3b
To: sip:172.25.220.18
Contact: sip:asterisk@10.123.240.144:5060
Call-ID: 747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #1 (no NAT) to 172.25.220.18:5060:
OPTIONS sip:172.25.220.18 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK27d7a413
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.123.240.144;tag=as2dce5b3b
To: sip:172.25.220.18
Contact: sip:asterisk@10.123.240.144:5060
Call-ID: 747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:8.48.100.197:5060 —>
INVITE sip:+15719188345@54.209.146.133:5060 SIP/2.0
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5769c93fc8e7cb3e
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688951 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,OPTIONS
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060
Recv-Info: x-broadworks-client-session-info
Content-Length: 308
Content-Disposition: session; handling=required
Content-Type: application/sdp

v=0
o=Sonus_UAC 491917 593378 IN IP4 8.48.100.197
s=SIP Media Capabilities
c=IN IP4 8.48.100.196
t=0 0
m=audio 6582 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
— (14 headers 14 lines) —
Sending to 8.48.100.197:5060 (no NAT)
Sending to 8.48.100.197:5060 (no NAT)
Using INVITE request as basis request - 1059112438_132107038@8.48.100.197
Found peer ‘level3a’ for ‘+18166121399’ from 8.48.100.197:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g729|ulaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (g729|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 8.48.100.196:6582
Looking for +15719188345 in incoming (domain 54.209.146.133)
sip_route_dump: route/path hop: sip:+18166121399@8.48.100.197:5060

<— Transmitting (NAT) to 8.48.100.197:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5769c93fc8e7cb3e;received=8.48.100.197;rport=5060
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688951 INVITE
Server: Asterisk PBX 16.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+15719188345@54.209.146.133:5060
Content-Length: 0

<------------>
Audio is at 11568
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.24.220.18:5060:
INVITE sip:+15719188345@Siptel01.multiservice.com SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK297d0e1e
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com
Contact: sip:+18166121399@10.123.240.144:5060
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1022987235 1022987235 IN IP4 10.123.240.144
s=Asterisk PBX 16.1.0
c=IN IP4 54.209.146.133
t=0 0
m=audio 11568 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<— SIP read from UDP:172.24.220.18:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK297d0e1e;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Length: 0

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

<— SIP read from UDP:172.24.220.18:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK297d0e1e;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com;tag=as745a19c8
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
sip_route_dump: route/path hop: sip:+15719188345@172.24.220.18:5060

<— Transmitting (NAT) to 8.48.100.197:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5769c93fc8e7cb3e;received=8.48.100.197;rport=5060
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060;tag=as421a880c
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688951 INVITE
Server: Asterisk PBX 16.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+15719188345@54.209.146.133:5060
Content-Length: 0

<------------>
Retransmitting #2 (no NAT) to 172.25.220.18:5060:
OPTIONS sip:172.25.220.18 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK27d7a413
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.123.240.144;tag=as2dce5b3b
To: sip:172.25.220.18
Contact: sip:asterisk@10.123.240.144:5060
Call-ID: 747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #3 (no NAT) to 172.25.220.18:5060:
OPTIONS sip:172.25.220.18 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK27d7a413
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.123.240.144;tag=as2dce5b3b
To: sip:172.25.220.18
Contact: sip:asterisk@10.123.240.144:5060
Call-ID: 747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:172.24.220.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK297d0e1e;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com;tag=as745a19c8
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 102 INVITE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:+15719188345@172.24.220.18:5060
Content-Type: application/sdp
Require: timer
Content-Length: 236

v=0
o=root 178532178 178532178 IN IP4 172.24.220.18
s=Asterisk PBX 11.25.1
c=IN IP4 172.24.220.18
t=0 0
m=audio 10946 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
— (14 headers 11 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.24.220.18:10946
sip_route_dump: route/path hop: sip:+15719188345@172.24.220.18:5060
set_destination: Parsing sip:+15719188345@172.24.220.18:5060 for address/port to send to
set_destination: set destination to 172.24.220.18:5060
Transmitting (no NAT) to 172.24.220.18:5060:
ACK sip:+15719188345@172.24.220.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK4c389932
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com;tag=as745a19c8
Contact: sip:+18166121399@10.123.240.144:5060
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.1.0
Content-Length: 0

Audio is at 18982
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (NAT) to 8.48.100.197:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5769c93fc8e7cb3e;received=8.48.100.197;rport=5060
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060;tag=as421a880c
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688951 INVITE
Server: Asterisk PBX 16.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:+15719188345@54.209.146.133:5060
Content-Type: application/sdp
Content-Length: 284

v=0
o=root 5253363 5253363 IN IP4 54.209.146.133
s=Asterisk PBX 16.1.0
c=IN IP4 54.209.146.133
t=0 0
m=audio 18982 RTP/AVP 18 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

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

<— SIP read from UDP:8.48.100.197:5060 —>
ACK sip:+15719188345@54.209.146.133:5060 SIP/2.0
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B583b2e19c8e7cb3e
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060;tag=as421a880c
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688951 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Retransmitting #4 (no NAT) to 172.25.220.18:5060:
OPTIONS sip:172.25.220.18 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK27d7a413
Max-Forwards: 70
From: “asterisk” sip:asterisk@10.123.240.144;tag=as2dce5b3b
To: sip:172.25.220.18
Contact: sip:asterisk@10.123.240.144:5060
Call-ID: 747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 16.1.0
Date: Mon, 07 Jan 2019 21:40:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Really destroying SIP dialog ‘747ede3f34afb7761e0221fd283fe4eb@10.123.240.144:5060’ Method: OPTIONS

<— SIP read from UDP:8.48.100.197:5060 —>
BYE sip:+15719188345@54.209.146.133:5060 SIP/2.0
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5968fa79c8e7cb3e
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060;tag=as421a880c
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688952 BYE
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Sending to 8.48.100.197:5060 (NAT)
Scheduling destruction of SIP dialog ‘1059112438_132107038@8.48.100.197’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 8.48.100.197:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 8.48.100.197:5060;branch=z9hG4bK00B5968fa79c8e7cb3e;received=8.48.100.197;rport=5060
From: “Siemens Daniel” sip:+18166121399@8.48.100.197:5060;tag=gK00293ad7
To: sip:+15719188345@54.209.146.133:5060;tag=as421a880c
Call-ID: 1059112438_132107038@8.48.100.197
CSeq: 688952 BYE
Server: Asterisk PBX 16.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:+15719188345@172.24.220.18:5060 for address/port to send to
set_destination: set destination to 172.24.220.18:5060
Reliably Transmitting (no NAT) to 172.24.220.18:5060:
BYE sip:+15719188345@172.24.220.18:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK1230a9af
Max-Forwards: 70
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com;tag=as745a19c8
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:172.24.220.18:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.240.144:5060;branch=z9hG4bK1230a9af;received=10.123.240.144
From: “Siemens Daniel” sip:+18166121399@10.123.240.144;tag=as13c7f629
To: sip:+15719188345@Siptel01.multiservice.com;tag=as745a19c8
Call-ID: 270af8bb1dab1d7262d1a21448966a4b@10.123.240.144:5060
CSeq: 103 BYE
Server: Asterisk PBX 11.25.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

6

Have you looked at the IP addresses in the SIP signaling and confirmed that they are as you’d expect in the situation?

7

I did a casual look and I only seen what I was expecting.

8

The INVITE to 172.24.220.18 has an SDP address of 54.209.146.133 - is this correct?

9

That would be a issue. why would it have the public ip and not the private IP, NAT is set to no in teh profile. I have nat setup as i have to get from 10.123.240.144 to the 8.48.100.197 address,

eg
[level3a]
type=peer
context=incoming
directmedia=no
qualify=yes
host=8.48.100.197
disallow=all
allow=g729
allow=ulaw
nat=force_rport,comedia

but the Profile from this server(10.123.240.144) to the next (172.24.220.18) is using nat=no .

10

What is the actual sip.conf configuration in the general section? The “nat” section only applies for incoming traffic from the remote side - that is whether the remote side is behind NAT or not.

11

externaddr = 54.209.146.133
externip = 54.209.146.133
localnet=10.123.240.0/255.255.255.0
localnet=172.24.220.0/255.255.255.0
localnet=172.25.220.0/255.255.255.0
localnet=10.119.220.0/255.255.254.0
media_address=54.209.146.133
rtptimeout=200

12

You have set the “media_address” option which overrides what is placed in the SDP. Remove that line. The extern/localnet options will update the SDP accordingly.

; The IP address used for media (audio, video, and text) in the SDP can also be overridden by using
; the media_address configuration option. This is only applicable to the general section and
; can not be set per-user or per-peer.
13

Thanks, That was it.