No sound when calling a phone number with sip provider

parisien9999 · October 23, 2012, 11:32am

Hello.

I have an asterisk configuration.

As i am at the office, the local hardphone is on the same network than the asterisk server.
(but when i am at home, it is not on the same network).

when i call the asterisk server from my sip phone, i here the recorded messages (thanks to david55)
But when i use a sip provider to call outside, i don’t here the calling person. there is no sound IN BOTH WAY.

but calling a sip phone to another on the same network works ok.

Here is my sip.conf :

[code][general]
canreinvite=no
Externip=xx.xxx.xxx.xxx
bindport=5060
bindaddr=0.0.0.0
nat=yes
;max_expiry=3600
;min_expiry=3600
defaultexpiry = 3600
localnet=192.168.0.0/255.255.255.0
srvlookup=yes
register = …

[freephonie]
type = peer
insecure = port,invite
host = freephonie.net
username = 0000000000
context = depuisfreephonie
fromuser = 0000000000
secret = password
nat = yes
fromdomain = freephonie.net
disallow = all
allow = alaw,ulaw
dtmfmode=auto

[freephonie_appelsortant]
type=peer
allow=all
host=freephonie.net
fromuser=0000000000
username=000000000
secret=password
dtmfmode=inband
qualify=yes
nat=yes
fromdomain=freephonie.net

[emeline]
type=friend
username=emeline
secret=password
callerid=“Emeline” <31>
host=dynamic
context=interne
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=31@default[/code]

my extension.conf :

[interne] exten => _0[1-578]XXXXXXXX,1,Dial(SIP/freephonie_appelsortant/${EXTEN})

the verbose :

== Using SIP RTP CoS mark 5 -- Executing [0155555555@interne:1] Dial("SIP/emeline-00000027", "SIP/freephonie_appelsortant/0155555555") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/freephonie_appelsortant/0155555555 -- SIP/freephonie_appelsortant-00000028 is ringing -- SIP/freephonie_appelsortant-00000028 is making progress passing it to SIP/emeline-00000027 -- SIP/freephonie_appelsortant-00000028 answered SIP/emeline-00000027 == Spawn extension (interne, 0155555555, 1) exited non-zero on 'SIP/emeline-00000027'

domaine.fr is the domain of the asterisk network
0155555555 is the number i am calling using a sip provider
192.168.0.28 is the ip of the local phone
192.168.0.41 is the ip of the asterisk server
zz.zzz.zzz.zzz is the public ip of the asterisk server network

and the debug :

[code]

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
INVITE sip:0155555555@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK7b38ff23136cd207316bce62c8d80c93;rport
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone
Call-ID: 2350295388@192_168_0_28
CSeq: 2 INVITE
Contact: sip:emeline@192.168.0.28:5060
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 384

v=0
o=emeline 5004 32 IN IP4 192.168.0.28
s=Mapping
c=IN IP4 192.168.0.28
t=0 0
m=audio 5004 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (14 headers 17 lines) —
Sending to 192.168.0.28:5060 (NAT)
Using INVITE request as basis request - 2350295388@192_168_0_28
Found peer ‘emeline’ for ‘emeline’ from 192.168.0.28:5060

<— Reliably Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK7b38ff23136cd207316bce62c8d80c93;received=192.168.0.28;rport=5060
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as679b1a5c
Call-ID: 2350295388@192_168_0_28
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6d2e9b0b"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘2350295388@192_168_0_28’ in 32000 ms (Method: INVITE)

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
ACK sip:0155555555@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK7b38ff23136cd207316bce62c8d80c93;rport
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as679b1a5c
Call-ID: 2350295388@192_168_0_28
CSeq: 2 ACK
Contact: sip:emeline@192.168.0.28:5060
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Content-Length: 0

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

localhost*CLI>
— (10 headers 0 lines) —

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
INVITE sip:0155555555@domaine.fr;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK87e4a47e8bf36c2dbc698f27e0988ebe;rport
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone
Call-ID: 2350295388@192_168_0_28
CSeq: 3 INVITE
Contact: sip:emeline@192.168.0.28:5060
Authorization: Digest username=“emeline”, realm=“asterisk”, algorithm=MD5, uri="sip:0155555555@domaine.fr;user=phone", nonce=“6d2e9b0b”, response="4025bc0b70ea6cd6c53fdcaa00110ef1"
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 384

v=0
o=emeline 5004 32 IN IP4 192.168.0.28
s=Mapping
c=IN IP4 192.168.0.28
t=0 0
m=audio 5004 RTP/AVP 9 0 8 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (15 headers 17 lines) —
Sending to 192.168.0.28:5060 (NAT)
Using INVITE request as basis request - 2350295388@192_168_0_28
Found peer ‘emeline’ for ‘emeline’ from 192.168.0.28:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x191c (ulaw|alaw|g726|g729|g726aal2|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.28:5004
Looking for 0155555555 in interne (domain domaine.fr)
list_route: hop: sip:emeline@192.168.0.28:5060

<— Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK87e4a47e8bf36c2dbc698f27e0988ebe;received=192.168.0.28;rport=5060
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone
Call-ID: 2350295388@192_168_0_28
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0155555555@192.168.0.41:5060
Content-Length: 0

<------------>
– Executing [0155555555@interne:1] Dial(“SIP/emeline-0000002a”, “SIP/freephonie_appelsortant/0155555555”) in new stack
== Using SIP RTP CoS mark 5

localhost*CLI>
We think we can do text
Audio is at 14912
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x200000000 (speex16) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Reliably Transmitting (NAT) to 212.27.52.5:5060:
INVITE sip:0155555555@freephonie.net SIP/2.0
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;branch=z9hG4bK2fd038c7;rport
Max-Forwards: 70
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net
Contact: sip:0000000000@zz.zzz.zzz.zzz:5060
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 23 Oct 2012 11:19:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 527

v=0
o=root 1905872860 1905872860 IN IP4 zz.zzz.zzz.zzz
s=Asterisk PBX 1.8.17.0
c=IN IP4 zz.zzz.zzz.zzz
t=0 0
m=audio 14912 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=ptime:20
a=sendrecv

-- Called SIP/freephonie_appelsortant/0155555555

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 100 Trying
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 102 INVITE
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK2fd038c7
Content-Length: 0

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 407 authentication required
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
Contact: sip:0155555555@172.17.20.241:5062;user=phone
CSeq: 102 INVITE
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
Proxy-Authenticate: Digest realm=“freephonie.net”,nonce=“0ed98c325778e9925cc6d2232a9e32cd”,opaque=“0ed91b327293b5c”,stale=false,algorithm=MD5
Record-Route: sip:212.27.52.5:5060;transport=udp;lr
To: sip:0155555555@freephonie.net;tag=00-08190-0ed98c59-783d42b46
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK2fd038c7
Allow: UPDATE,REFER,INFO
Server: Cirpack/v4.42q (gw_sip)
Content-Length: 0

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

localhost*CLI>
— (12 headers 0 lines) —

localhost*CLI>
Transmitting (NAT) to 212.27.52.5:5060:
ACK sip:0155555555@freephonie.net SIP/2.0
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;branch=z9hG4bK2fd038c7;rport
Max-Forwards: 70
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net;tag=00-08190-0ed98c59-783d42b46
Contact: sip:0000000000@zz.zzz.zzz.zzz:5060
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

localhost*CLI>
We think we can do text

localhost*CLI>
Audio is at 14912
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP

localhost*CLI>
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP

localhost*CLI>
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP

localhost*CLI>
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP

localhost*CLI>
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding codec 0x200000000 (speex16) to SDP

localhost*CLI>
Adding codec 0x800000000000 (testlaw) to SDP
Reliably Transmitting (NAT) to 212.27.52.5:5060:
INVITE sip:0155555555@freephonie.net SIP/2.0
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;branch=z9hG4bK5720d6e6;rport
Max-Forwards: 70
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net
Contact: sip:0000000000@zz.zzz.zzz.zzz:5060
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username=“0000000000”, realm=“freephonie.net”, algorithm=MD5, uri="sip:0155555555@freephonie.net", nonce=“0ed98c325778e9925cc6d2232a9e32cd”, response=“71b0ccd1471e89c25bf23ec3a8884663”, opaque="0ed91b327293b5c"
Date: Tue, 23 Oct 2012 11:19:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 527

v=0
o=root 1905872860 1905872861 IN IP4 zz.zzz.zzz.zzz
s=Asterisk PBX 1.8.17.0
c=IN IP4 zz.zzz.zzz.zzz
t=0 0
m=audio 14912 RTP/AVP 0 3 8 112 5 10 7 110 97 111 9 118 117
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:118 L16/16000
a=rtpmap:117 speex/16000
a=ptime:20
a=sendrecv

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 100 Trying
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 103 INVITE
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK5720d6e6
Content-Length: 0

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhostCLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 180 Ringing
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
Contact: sip:172.17.20.241:5062
Content-Type: application/sdp
CSeq: 103 INVITE
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
Record-Route: sip:212.27.52.5:5060;transport=udp;lr
To: sip:0155555555@freephonie.net;tag=00-08190-0ed98c5b-137ad7f74
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK5720d6e6
Allow: UPDATE,REFER,INFO
S
localhostCLI>
erver: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 135099119446 135099119447 IN IP4 172.25.50.22
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 36408 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv
<------------->

localhost*CLI>
— (12 headers 10 lines) —

localhost*CLI>
list_route: hop: sip:212.27.52.5:5060;transport=udp;lr

localhost*CLI>
Found RTP audio format 8

localhost*CLI>
Found audio description format PCMA for ID 8

localhost*CLI>
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)

localhost*CLI>
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)

localhost*CLI>
Peer audio RTP is at port 212.27.52.130:36408

localhost*CLI>
– SIP/freephonie_appelsortant-0000002b is ringing

localhost*CLI>
<— Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK87e4a47e8bf36c2dbc698f27e0988ebe;received=192.168.0.28;rport=5060
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
Call-ID: 2350295388@192_168_0_28
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0155555555@192.168.0.41:5060
Content-Length: 0

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

localhost*CLI>
– SIP/freephonie_appelsortant-0000002b is making progress passing it to SIP/emeline-0000002a

localhost*CLI>
Audio is at 17322

localhost*CLI>
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK87e4a47e8bf36c2dbc698f27e0988ebe;received=192.168.0.28;rport=5060
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
Call-ID: 2350295388@192_168_0_28
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0155555555@192.168.0.41:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1033994131 1033994131 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 17322 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Reliably Transmitting (NAT) to 192.168.0.28:5060:
OPTIONS sip:test@192.168.0.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK1214b247;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.0.41;tag=as23fc449d
To: sip:test@192.168.0.28:5060
Contact: sip:asterisk@192.168.0.41:5060
Call-ID: 55c75ff77d8e09f420722b2d7e678b03@192.168.0.41:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 23 Oct 2012 11:20:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK1214b247;rport=5060
From: “asterisk” sip:asterisk@192.168.0.41;tag=as23fc449d
To: sip:test@192.168.0.28:5060;tag=ar32gb558e
Call-ID: 55c75ff77d8e09f420722b2d7e678b03@192.168.0.41:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: A510 IP/42.072.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘55c75ff77d8e09f420722b2d7e678b03@192.168.0.41:5060’ Method: OPTIONS

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 200 OK
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
Contact: sip:172.17.20.241:5062
Content-Type: application/sdp
CSeq: 103 INVITE
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
Record-Route: sip:212.27.52.5:5060;transport=udp;lr
To: sip:0155555555@freephonie.net;tag=00-08190-0ed98c5b-137ad7f74
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK5720d6e6
Allow: UPDATE,REFER,INFO
Server: Cirpack/v4.42q (gw_sip)
Content-Length: 184

v=0
o=cp10 135099119446 135099119447 IN IP4 172.25.50.22
s=SIP Call
c=IN IP4 212.27.52.130
t=0 0
m=audio 36408 RTP/AVP 8
b=AS:75
a=rtpmap:8 PCMA/8000/1
a=ptime:30
a=sendrecv
<------------->

localhost*CLI>
— (12 headers 10 lines) —

localhost*CLI>
list_route: hop: sip:212.27.52.5:5060;transport=udp;lr

localhost*CLI>
set_destination: Parsing sip:212.27.52.5:5060;transport=udp;lr for address/port to send to

localhost*CLI>
set_destination: set destination to 212.27.52.5:5060

localhost*CLI>
Transmitting (NAT) to 212.27.52.5:5060:
ACK sip:172.17.20.241:5062 SIP/2.0
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;branch=z9hG4bK0db4b567;rport
Route: sip:212.27.52.5:5060;transport=udp;lr
Max-Forwards: 70
From: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
To: sip:0155555555@freephonie.net;tag=00-08190-0ed98c5b-137ad7f74
Contact: sip:0000000000@zz.zzz.zzz.zzz:5060
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.17.0
Content-Length: 0

localhost*CLI>
– SIP/freephonie_appelsortant-0000002b answered SIP/emeline-0000002a

localhost*CLI>
Audio is at 17322

localhost*CLI>
Adding codec 0x4 (ulaw) to SDP

localhost*CLI>
Adding codec 0x8 (alaw) to SDP

localhost*CLI>
Adding non-codec 0x1 (telephone-event) to SDP

localhost*CLI>
<— Reliably Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK87e4a47e8bf36c2dbc698f27e0988ebe;received=192.168.0.28;rport=5060
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
Call-ID: 2350295388@192_168_0_28
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0155555555@192.168.0.41:5060
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 1033994131 1033994132 IN IP4 192.168.0.41
s=Asterisk PBX 1.8.17.0
c=IN IP4 192.168.0.41
t=0 0
m=audio 17322 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

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

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
ACK sip:0155555555@192.168.0.41:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bKa771c1b8cc5d8e1cb9f69e6b377a074e;rport
From: “emeline” sip:emeline@domaine.fr;tag=3352651620
To: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
Call-ID: 2350295388@192_168_0_28
CSeq: 3 ACK
Contact: sip:emeline@192.168.0.28:5060
Authorization: Digest username=“emeline”, realm=“asterisk”, algorithm=MD5, uri="sip:0155555555@domaine.fr;user=phone", nonce=“6d2e9b0b”, response="4025bc0b70ea6cd6c53fdcaa00110ef1"
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Content-Length: 0

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

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Really destroying SIP dialog ‘3821391563@192_168_1_18’ Method: REGISTER

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Really destroying SIP dialog ‘4146727082@192_168_1_18’ Method: REGISTER

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
REGISTER sip:bureau.domaine.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK73ddcf4980ebe2cff1222eb346c7f9a;rport
From: “test” sip:test@bureau.domaine.fr;tag=2826242201
To: “test” sip:test@bureau.domaine.fr
Call-ID: 3949043359@192_168_1_18
CSeq: 97 REGISTER
Contact: sip:test@192.168.0.28:5060
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

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

localhost*CLI>
— (12 headers 0 lines) —

localhost*CLI>
Sending to 192.168.0.28:5060 (NAT)

localhost*CLI>
<— Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK73ddcf4980ebe2cff1222eb346c7f9a;received=192.168.0.28;rport=5060
From: “test” sip:test@bureau.domaine.fr;tag=2826242201
To: “test” sip:test@bureau.domaine.fr;tag=as20105a64
Call-ID: 3949043359@192_168_1_18
CSeq: 97 REGISTER
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4ce09804"
Content-Length: 0

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

localhost*CLI>
Scheduling destruction of SIP dialog ‘3949043359@192_168_1_18’ in 32000 ms (Method: REGISTER)

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
REGISTER sip:bureau.domaine.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK911607513d5065194f4748a967d2ced2;rport
From: “test” sip:test@bureau.domaine.fr;tag=2826242201
To: “test” sip:test@bureau.domaine.fr
Call-ID: 3949043359@192_168_1_18
CSeq: 98 REGISTER
Contact: sip:test@192.168.0.28:5060
Authorization: Digest username=“test”, realm=“asterisk”, algorithm=MD5, uri=“sip:bureau.domaine.fr”, nonce=“4ce09804”, response="669b6d84c71ada29c371e3c3baaedceb"
Max-Forwards: 70
User-Agent: A510 IP/42.072.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

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

localhost*CLI>
— (13 headers 0 lines) —

localhostCLI>
Sending to 192.168.0.28:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.0.28:5060:
OPTIONS sip:test@192.168.0.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK27f05238;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.0.41;tag=as5142d094
To: sip:test@192.168.0.28:5060
Contact: sip:asterisk@192.168.0.41:5060
Call-ID: 183e6892111e971d367bd1a10d43fe1c@192.168.0.41:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 23 Oct 2012 11:20:18 GMT
A
localhostCLI>
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

localhost*CLI>
<— Transmitting (NAT) to 192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.28:5060;branch=z9hG4bK911607513d5065194f4748a967d2ced2;received=192.168.0.28;rport=5060
From: “test” sip:test@bureau.domaine.fr;tag=2826242201
To: “test” sip:test@bureau.domaine.fr;tag=as20105a64
Call-ID: 3949043359@192_168_1_18
CSeq: 98 REGISTER
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 180
Contact: sip:test@192.168.0.28:5060;expires=180
Date: Tue, 23 Oct 2012 11:20:18 GMT
Content-Length: 0

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

localhost*CLI>
Scheduling destruction of SIP dialog ‘651f4cd20a9771d03f9211df135c8278@192.168.0.41:5060’ in 6400 ms (Method: NOTIFY)

localhost*CLI>
Reliably Transmitting (NAT) to 192.168.0.28:5060:
NOTIFY sip:test@192.168.0.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK6796220a;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.0.41;tag=as502960e3
To: sip:test@192.168.0.28:5060
Contact: sip:asterisk@192.168.0.41:5060
Call-ID: 651f4cd20a9771d03f9211df135c8278@192.168.0.41:5060
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 1.8.17.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 92

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.0.41
Voice-Message: 0/0 (0/0)

localhost*CLI>
Scheduling destruction of SIP dialog ‘3949043359@192_168_1_18’ in 32000 ms (Method: REGISTER)

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK27f05238;rport=5060
From: “asterisk” sip:asterisk@192.168.0.41;tag=as5142d094
To: sip:test@192.168.0.28:5060;tag=ar4053e185
Call-ID: 183e6892111e971d367bd1a10d43fe1c@192.168.0.41:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: A510 IP/42.072.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Accept: application/sdp,application/dtmf-relay,application/simple-message-summary,message/sipfrag
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

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

localhost*CLI>
— (13 headers 0 lines) —

localhost*CLI>
Really destroying SIP dialog ‘183e6892111e971d367bd1a10d43fe1c@192.168.0.41:5060’ Method: OPTIONS

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK6796220a;rport=5060
From: “asterisk” sip:asterisk@192.168.0.41;tag=as502960e3
To: sip:test@192.168.0.28:5060;tag=ar413871d2
Call-ID: 651f4cd20a9771d03f9211df135c8278@192.168.0.41:5060
CSeq: 102 NOTIFY
User-Agent: A510 IP/42.072.00.000.000
Content-Length: 0

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

localhost*CLI>
— (8 headers 0 lines) —

localhost*CLI>
Really destroying SIP dialog ‘651f4cd20a9771d03f9211df135c8278@192.168.0.41:5060’ Method: NOTIFY

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
BYE sip:0000000000@zz.zzz.zzz.zzz:5060 SIP/2.0
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 243897261 BYE
From: sip:0155555555@freephonie.net;tag=00-08190-0ed98c5b-137ad7f74
Max-Forwards: 30
Record-Route: sip:C=on;t=DDKDD@212.27.52.5:5060;lr
To: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-DDKD-0495c203-376402ff
Reason: q.850;cause=16
User-Agent: Cirpack/v4.42q (gw_sip)
Content-Length: 0

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

localhost*CLI>
— (11 headers 0 lines) —

localhost*CLI>
Sending to 212.27.52.5:5060 (NAT)

localhost*CLI>
Scheduling destruction of SIP dialog ‘6b761acc4835e9070979bc8453dc708d@freephonie.net’ in 6400 ms (Method: BYE)

localhost*CLI>
<— Transmitting (NAT) to 212.27.52.5:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-DDKD-0495c203-376402ff;received=212.27.52.5;rport=5060
Record-Route: sip:C=on;t=DDKDD@212.27.52.5:5060;lr
From: sip:0155555555@freephonie.net;tag=00-08190-0ed98c5b-137ad7f74
To: “Emeline” sip:0000000000@freephonie.net;tag=as6febc0a9
Call-ID: 6b761acc4835e9070979bc8453dc708d@freephonie.net
CSeq: 243897261 BYE
Server: Asterisk PBX 1.8.17.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

localhost*CLI>
== Spawn extension (interne, 0155555555, 1) exited non-zero on ‘SIP/emeline-0000002a’

localhost*CLI>
Scheduling destruction of SIP dialog ‘2350295388@192_168_0_28’ in 32000 ms (Method: ACK)

localhost*CLI>
set_destination: Parsing sip:emeline@192.168.0.28:5060 for address/port to send to

localhost*CLI>
set_destination: set destination to 192.168.0.28:5060

localhost*CLI>
Reliably Transmitting (NAT) to 192.168.0.28:5060:
BYE sip:emeline@192.168.0.28:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK30d1bd27;rport
Max-Forwards: 70
From: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
To: “emeline” sip:emeline@domaine.fr;tag=3352651620
Call-ID: 2350295388@192_168_0_28
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.17.0
Proxy-Authorization: Digest username=“emeline”, realm=“asterisk”, algorithm=MD5, uri=“sip:domaine.fr”, nonce="", response="9b0f72c134975cbf7d30eda345144b47"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

localhost*CLI>
<— SIP read from UDP:192.168.0.28:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.41:5060;branch=z9hG4bK30d1bd27;rport=5060
From: sip:0155555555@domaine.fr;user=phone;tag=as629beb0f
To: “emeline” sip:emeline@domaine.fr;tag=3352651620
Call-ID: 2350295388@192_168_0_28
CSeq: 102 BYE
Contact: sip:emeline@192.168.0.28:5060
Supported: replaces
User-Agent: A510 IP/42.072.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

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

localhost*CLI>
— (11 headers 0 lines) —

localhost*CLI>
SIP Response message for INCOMING dialog BYE arrived

localhost*CLI>
Really destroying SIP dialog ‘2350295388@192_168_0_28’ Method: ACK

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Reliably Transmitting (NAT) to 212.27.52.5:5060:
OPTIONS sip:freephonie.net SIP/2.0
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;branch=z9hG4bK39c3437a;rport
Max-Forwards: 70
From: “asterisk” sip:0000000000@zz.zzz.zzz.zzz;tag=as7034ba46
To: sip:freephonie.net
Contact: sip:0000000000@zz.zzz.zzz.zzz:5060
Call-ID: 2e3861887f5b42a909c39d293de83209@zz.zzz.zzz.zzz:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.17.0
Date: Tue, 23 Oct 2012 11:20:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
SIP/2.0 501 Not Implemented Yet
Call-ID: 2e3861887f5b42a909c39d293de83209@zz.zzz.zzz.zzz:5060
CSeq: 102 OPTIONS
From: “asterisk” sip:0000000000@zz.zzz.zzz.zzz;tag=as7034ba46
To: sip:freephonie.net;tag=00-31187-0947105b-30b4ee592
Via: SIP/2.0/UDP zz.zzz.zzz.zzz:5060;received=zz.zzz.zzz.zzz;rport=5060;branch=z9hG4bK39c3437a
Content-Length: 0

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

localhost*CLI>
— (7 headers 0 lines) —

localhost*CLI>
Really destroying SIP dialog ‘2e3861887f5b42a909c39d293de83209@zz.zzz.zzz.zzz:5060’ Method: OPTIONS

localhost*CLI>
<— SIP read from UDP:212.27.52.5:5060 —>
Cirpack KeepAlive Packet
<------------->

localhost*CLI>
Really destroying SIP dialog ‘6b761acc4835e9070979bc8453dc708d@freephonie.net’ Method: BYE

localhost*CLI> [/code]

do you know what can be the problem ?

thank you

parisien9999 · October 23, 2012, 1:09pm

i made tests.

incoming call from provider to local sip phone works perfectly. (there is sound in both way)

it is when we use the provider to call from local sip phone to an outside phone number that there is no sound at all. (no sound in both way)

parisien9999 · October 23, 2012, 11:28pm

Hello.

i write the sip show peers that maybe can help :

* Name : freephonie_appelsortant Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened FromUser : 0900000000 FromDomain : freephonie.net Port 5060 Callgroup : Pickupgroup : MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : inband Timer T1 : 500 Timer B : 32000 ToHost : freephonie.net Addr->IP : 212.27.52.5:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: 0900000000 SIP Options : (none) Codecs : 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719) Codec Order : (none) Auto-Framing : No Status : Unmonitored Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No

(this is the peer of the phone when i am at home so not the same network than the asterisk server, but the problem is exactly the same)

* Name : Maison Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : interne Subscr.Cont. : <Not set> Language : fr AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 39@default VM Extension : asterisk LastMsgsSent : 0/0 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : "Maison" <39> MaxCallBR : 384 kbps Expire : 29 Insecure : port Force rport : Yes ACL : No DirectMedACL : No T.38 support : No T.38 EC mode : Unknown T.38 MaxDtgrm: -1 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : xx.xxx.xxx.xxx:5060 (home network public ip where i am calling from) Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: Maison SIP Options : replaces replace Codecs : 0x80000008000e (gsm|ulaw|alaw|h263|testlaw) Codec Order : (none) Auto-Framing : No Status : Unmonitored Useragent : A580 IP/022270000000 Reg. Contact : sip:Maison@xx.xx.xxx.xxx:5060 (home network public ip where i am calling from) Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No

another thing that can help :

localhost*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 212.27.52.5 0155555555 21fe94494142968 0x8 (alaw) No Tx: ACK freephonie xx.xx.xxx.xxx Maison 1401570624@192_ 0x4 (ulaw) No Rx: ACK Maison 2 active SIP dialogs

( xx.xx.xxx.xxx is the public ip of the home network where i am calling from)
192.168.0.28 is the sip local phone that is on the work network and that i don’t use right now.

[code]localhost*CLI> sip show channel 21fe94494142968

SIP Call
Curr. trans. direction: Outgoing
Call-ID: 21fe94494142968d4833df8509946e16@freephonie.net
Owner channel ID: SIP/freephonie-00000041
Our Codec Capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: 0x8 (alaw)
Joint Codec Capability: 0x8 (alaw)
Format: 0x8 (alaw)
T.38 support No
Video support No
MaxCallBR: 384 kbps
Theoretical Address: 212.27.52.5:5060
Received Address: 212.27.52.5:5060
SIP Transfer mode: open
Force rport: Yes
Audio IP: yy.yy.yyy.yyy (local)
Our Tag: as1e23e64e
Their Tag: 00-08188-0ee7945b-6b48f3f46
SIP User agent:
Username: 0155555555
Peername: freephonie
Original uri: sip:172.17.20.241:5062
Need Destroy: No
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:212.27.52.5:5060;transport=udp;lr
DTMF Mode: inband
SIP Options: (none)
Session-Timer: Inactive
[/code]
yy.yy.yyy.yyy is the public ip address of the asterisk server network

[code]localhost*CLI> sip show channel 1401570624@192_

SIP Call
Curr. trans. direction: Incoming
Call-ID: 1401570624@192_168_1_75
Owner channel ID: SIP/Maison-00000040
Our Codec Capability: 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
Non-Codec Capability (DTMF): 1
Their Codec Capability: 0x191c (ulaw|alaw|g726|g729|g726aal2|g722)
Joint Codec Capability: 0xc (ulaw|alaw)
Format: 0x4 (ulaw)
T.38 support No
Video support No
MaxCallBR: 384 kbps
Theoretical Address: xx.xx.xxx.xxx:5060
Received Address: xx.xx.xxx.xxx:5060
SIP Transfer mode: open
Force rport: Yes
Audio IP: yy.yy.yyy.yyy (local)
Our Tag: as4929fe69
Their Tag: 1797182532
SIP User agent: A580 IP/022270000000
Username: Maison
Peername: Maison
Original uri: sip:Maison@xx.xx.xxx.xxx:5060
Caller-ID: 39
Need Destroy: No
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:Maison@xx.xx.xxx.xxx:5060
DTMF Mode: rfc2833
SIP Options: replaces replace
Session-Timer: Inactive
[/code]

yy.yy.yyy.yyy is the public ip address of the asterisk server network
xx.xx.xxx.xxx is the public ip address of the network from where i am calling (where is the sip phone)

(in the both case, incoming and outgoing, the Audio IP is the public ip of the asterisk server network, can it be the problem ?)

really nobody have an idea ?

thank you

pmlco · October 24, 2012, 12:07pm

Can you turn on RTP debugging to see where the media packets are going?

Ian

parisien9999 · October 24, 2012, 12:22pm

Hello.

can you tell me how i can do that ?

Thank you

pmlco · October 25, 2012, 12:34pm

CLI> rtp set debug on

then make a call.

Ian.

parisien9999 · October 25, 2012, 1:07pm

Hello.

Thank you for your answer.

I don’t understand what happened but i didn’t hange almost anything and suddainly it starts working perfectly.

So no need that i send it anymore and thank you for your answer.

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