LLamadas pegadas en troncal SIP

**navaismo** · 06-05-2013, 10:19 AM

Como te comentaba en tu tema anterior usa GROUP y GROUP_COUNT calllimit es un setting marcado como 'deprecated' u obsoleto. Sobre el tema de que se queda la linea como busy, haz un sip debug de tu llamada y al colgar se deben enviar los BYE, después de eso revisa el estado de tu linea con sip show peers y core show hints

**Gendo Ikari** · 06-05-2013, 11:54 AM

Me aparece esto:

SIP/2.0 183 Session Description
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bK287a 92c3
Record-Route: <sip:207.249.182.154;lr>
From: "SIPLineUser SIPLineUser" <sip:1000XXXXXXXip@mex1.axtel.net>;tag=as0be0bae 6
To: <sip:0445530016462@mex1.axtel.net:5060>;tag=3204 1
Call-ID: 3e85e1937b9dbdf33ee3b2a02f3f5fa2@187.163.74.213
CSeq: 103 INVITE
Contact: <sip:0445530016462@207.249.182.154:5060;CtRt32d81a 01904c0031=udp:148.245.89.38:5060>
User-Agent: Nortel SESM 12.0.6.13
Supported: com.nortelnetworks.firewall, p-3rdpartycontrol, nosec, join, x-nortel-sipvc, com.nortelnetworks.im.encryption
Content-Type: application/sdp
Content-Length: 302

Al "colgar" no me aparecio BYE, aparece 103 invite. A que se debe esto?

**navaismo** · 06-05-2013, 12:52 PM

Tendríamos que ver el Debug completo de la llamada para poder ayudarte. Quizás el invite es de otra llamada solo se envían INVITES cuando la llamada se intenta establecer o cuando se envía el challenge resuelto.

Pega todo el sip debug de la llamada para saber más.

**Gendo Ikari** · 06-05-2013, 01:20 PM

Me aparece esto:

Scheduling destruction of SIP dialog '4745f1ff4029acf50f5d66c50b7a3c1a@XX.1XX.XX.2XX' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 207.249.182.154:5060:
CANCEL sip:0445530016462@mex14.axtel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 187.163.74.213:5060;branch=z9hG4bK2fd385d1;rport
Max-Forwards: 70
From: "soporte_tecnico" <sip:1000XXXXXXXip@1X.1XX.7X.2ZZ>;tag=as5f59fae5
To: <sip:0445530016462@mex14.axtel.net:5060>
Call-ID: 4745f1ff4029acf50f5d66c50b7a3c1a@XX.1XX.XX.2XX
CSeq: 103 CANCEL
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0

---
Scheduling destruction of SIP dialog '4745f1ff4029acf50f5d66c50b7a3c1a@187.163.74.213' in 6400 ms (Method: INVITE)
== Spawn extension (entrantes, 0445530016462, 3) exited non-zero on 'SIP/110-00000011'

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.16X.7X.XXX:5060;branch=z9hG4bK2fd385d1;rport=5 060
From: "soporte_tecnico" <sip:1000XXXXXXX@1XX.1XX.XX.2XX>;tag=as5f59fae5
To: <sip:0445530016462@mex14.axtel.net:5060>;tag=SK6py vBBy0m3N
Call-ID: 4745f1ff4029acf50f5d66c50b7a3c1a@1XX.1XX.7X.2XX
CSeq: 103 CANCEL
Content-Length: 0

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

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 1XX.1XX.7X.2XX:5060;branch=z9hG4bK2fd385d1;rport=5 060
From: "soporte_tecnico" <sip:1000XXXXXXXXip@1XX.1XX.7X.2XX>;tag=as5f59fa e5
To: <sip:0445530016462@mex14.axtel.net:5060>;tag=tvZF0 QveU9apH
Call-ID: 4745f1ff4029acf50f5d66c50b7a3c1a@1XX.1XX.7X.2XX
CSeq: 103 INVITE
Server: Flexisip/0.5.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 207.249.182.154:5060:
ACK sip:0445530016462@207.249.182.154:5060;CtRt32d81a0 1904c0031=udp:148.245.89.38:5060 SIP/2.0
Via: SIP/2.0/UDP 1XX.1X3.74.2XX:5060;branch=z9hG4bK2fd385d1;rport
Max-Forwards: 70
From: "soporte_tecnico" <sip:XXXXXXXXXip@1XX.1XX.7X.2XX>;tag=as5f59fae5
To: <sip:0445530016462@mex14.axtel.net:5060>;tag=tvZF0 QveU9apH
Contact: <sip:1000XXXXXXXip@1XX.1XX.7X.2X3:5060>
Call-ID: 4745f1ff4029acf50f5d66c50b7a3c1a@1XX.1XX.7X.2XX
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0

Hay algo raro?

**navaismo** · 06-05-2013, 01:24 PM

Tu debug no esta completo, no te puedo decir que pasa o no con un debug incompleto.

**Gendo Ikari** · 06-05-2013, 02:13 PM

Really destroying SIP dialog '1b39580a7c26b8862b26ca541a780369@x.x.x.x:5060' Method: OPTIONS
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:mex15.axtel.net SIP/2.0
Via: SIP/2.0/UDP 187.163.74.213:5060;branch=z9hG4bK62dc4962;rport
Max-Forwards: 70
From: "asterisk" <sip:1000xxxxxxxip@x.x.x.x>;tag=as70e5de
To: <sip:mex15.axtel.net>
Contact: <sip:1000xxxxxxxip@x.x.x.x:5060>
Call-ID: 6436b06d55adf9283ce6283c57a8d4c9@x.x.x.x:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Wed, 05 Jun 2013 19:09:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK62dc4962;rport=5060
From: "asterisk" <sip:1000xxxxxxxip@x.x.x.x>;tag=as70e5de1a
To: <sip:mex15.axtel.net>;tag=UtK3SSa80XFSD
Call-ID: 6436b06d55adf9283ce6283c57a8d4c9@x.x.x.x:5060
CSeq: 102 OPTIONS
Contact: <sip:1000xxxxxxxip@x.x.x.x:5060>
Server: Flexisip/0.5.0 (sofia-sip-nta/2.0)
Content-Length: 0

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 183 Session Description
Via: SIP/2.0/UDP x.x.x.x:5060;rport=5060;branch=z9hG4bK346a124f
Record-Route: <sip:207.249.182.154;lr>
From: "SIPLineUser SIPLineUser" <sip:1000xxxxxxxip@mex1.axtel.net>;tag=as55f2181 2
To: <sip:0445530016462@mex1.axtel.net:5060>;tag=5819 7
Call-ID: 2168fc9b55aad83e435ba56a0fadc6b8@x.x.x.x
CSeq: 103 INVITE
Contact: <sip:0445530016462@207.249.182.154:5060;CtRt32d81a 01904c0031=udp:148.245.89.38:5060>
User-Agent: Nortel SESM 12.0.6.13
Supported: com.nortelnetworks.firewall, p-3rdpartycontrol, nosec, join, x-nortel-sipvc, com.nortelnetworks.im.encryption
Content-Type: application/sdp
Content-Length: 303

v=0
o=- 720529648 1043797350 IN IP4 148.245.89.102
s=-
e=unknown@invalid.net
c=IN IP4 207.249.182.154
t=0 0
m=audio 32402 RTP/AVP 110 3 0 8 101
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp

ff

Es lo que me sale desde que marco.

**Gendo Ikari** · 06-12-2013, 09:35 AM

Amigos, sigo sin poder hacer que las llamadas se cuelguen, eso que pegue es lo que me sale en el CLI, no hay mas, por favor, ayúdenme, ya probé muchas opciones y sigue igual =(.

**navaismo** · 06-12-2013, 09:51 AM

Lo lamento, pero eso no es un debut completo de una llamada y sin eso es realmente difícil ayudar o diagnosticar el problema.

**Gendo Ikari** · 06-12-2013, 12:20 PM

--- (11 headers 0 lines) ---
Transmitting (NAT) to 207.249.182.154:5060:
ACK sip:0445530016462@mex14.axtel.net:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK5de43c69;rport
Max-Forwards: 70
From: "soporte_tecnico" <sip:xxxxx@x.x.x.x>;tag=as5ee609f5
To: <sip:0445530016462@mex14.axtel.net:5060>;tag=14661 88244
Contact: <sip:xxxxxxxip@x.x.x.x:5060>
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0

---
We think we can do text
Audio is at 18192
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) 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 0x100 (g729) 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
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 207.249.182.154:5060:
INVITE sip:0445530016462@mex14.axtel.net:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK03c80372;rport
Max-Forwards: 70
From: "soporte_tecnico" <sip:xxxxxip@x.x.x.x>;tag=as5ee609f5
To: <sip:0445530016462@mex14.axtel.net:5060>
Contact: <sip:xxxxxip@x.x.x.x:5060>
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.15.0
Proxy-Authorization: Digest username="10003361548ip", realm="Realm", algorithm=MD5, uri="sip:0445530016462@mex14.axtel.net:5060", nonce="MTM3MTA1NjkzNTA2MzI1ZGNhYWRmYmUxMDJjZjg3OGI 3NGJkOWI2ZWYzMGQy", response="d86c8cc13ebfb71fdb0aef038e2e6691", qop=auth, cnonce="5dd88152", nc=00000001
Date: Wed, 12 Jun 2013 17:11:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 655

v=0
o=root 117921971 117921972 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.15.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 18192 RTP/AVP 3 0 8 112 5 10 7 18 110 97 111 9 118 117 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/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:18 G729/8000
a=fmtp:18 annexb=no
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=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv

---
Retransmitting #1 (NAT) to 207.249.182.154:5060:
INVITE sip:0445530016462@mex14.axtel.net:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK03c80372;rport
Max-Forwards: 70
From: "soporte_tecnico" <sip:xxxxxxip@x.x.x.x>;tag=as5ee609f5
To: <sip:0445530016462@mex14.axtel.net:5060>
Contact: <sip:xxxxxxxip@x.x.x.x:5060>
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.15.0
Proxy-Authorization: Digest username="10003361548ip", realm="Realm", algorithm=MD5, uri="sip:0445530016462@mex14.axtel.net:5060", nonce="MTM3MTA1NjkzNTA2MzI1ZGNhYWRmYmUxMDJjZjg3OGI 3NGJkOWI2ZWYzMGQy", response="d86c8cc13ebfb71fdb0aef038e2e6691", qop=auth, cnonce="5dd88152", nc=00000001
Date: Wed, 12 Jun 2013 17:11:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 655

v=0
o=root 117921971 117921972 IN IP4 x.x.x.x
s=Asterisk PBX 1.8.15.0
c=IN IP4 x.x.x.x
t=0 0
m=audio 18192 RTP/AVP 3 0 8 112 5 10 7 18 110 97 111 9 118 117 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/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:18 G729/8000
a=fmtp:18 annexb=no
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=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp

ff - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK5de43c69;rport=5060
From: "soporte_tecnico" <sip:xxxxxxip@x.x.x.x>;tag=as5ee609f5
To: <sip:0445530016462@mex14.axtel.net:5060>
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK03c80372;rport=5060
From: "soporte_tecnico" <sip:xxxxxxxip@x.x.x.x>;tag=as5ee609f5
To: <sip:0445530016462@mex14.axtel.net:5060>
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 183 Session Description
Via: SIP/2.0/UDP 1 x.x.x.x:5060;rport=5060;branch=z9hG4bK03c80372
Record-Route: <sip:207.249.182.154;lr>
From: "SIPLineUser SIPLineUser" <sip:xxxxxxxxip@mex1.axtel.net>;tag=as5ee609f5
To: <sip:0445530016462@mex1.axtel.net:5060>;tag=9000 0
Call-ID: 3915e25c6e66cb5435be8af266e4b66b@x.x.x.x
CSeq: 103 INVITE
Contact: <sip:0445530016462@207.249.182.154:5060;CtRt32d81a 01904c0031=udp:148.245.89.38:5060>
User-Agent: Nortel SESM 12.0.6.13
Supported: com.nortelnetworks.firewall, p-3rdpartycontrol, nosec, join, x-nortel-sipvc, com.nortelnetworks.im.encryption
Content-Type: application/sdp
Content-Length: 377

v=0
o=PVG 960514676 960514676 IN IP4 148.245.89.100
s=-
p=+1 6135555555
c=IN IP4 207.249.182.154
t=0 0
a=sqn: 0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 58064 RTP/AVP 110 3 0 8 101
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (12 headers 16 lines) ---
list_route: hop: <sip:207.249.182.154;lr>
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80030c7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|spe ex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261 |h263|h263p|h264|mpeg4|red|t140|siren7|siren14|tes tlaw|g719), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20e (gsm|ulaw|alaw|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 207.249.182.154:58064
-- SIP/63877770-00000019 is making progress passing it to SIP/110-00000018
[Jun 12 12:11:19] WARNING[3956]: dsp.c:1354 ast_dsp_process: Inband DTMF is not supported on codec speex. Use RFC2833

SIP/2.0 183 Session Description
Via: SIP/2.0/UDP x.x.x.x:5060;rport=5060;branch=z9hG4bK349689e3
Record-Route: <sip:207.249.182.154;lr>
From: "SIPLineUser SIPLineUser" <sip:1000XXXXip@mex1.axtel.net>;tag=as7ad856f5
To: <sip:0445530016462@mex1.axtel.net:5060>;tag=9577 4
Call-ID: 5a503571626e0219046672462fe2e43e@x.x.x.x
CSeq: 103 INVITE
Contact: <sip:0445530016462@207.249.182.154:5060;CtRt32d81a 01904c0031=udp:148.245.89.38:5060>
User-Agent: Nortel SESM 12.0.6.13
Supported: com.nortelnetworks.firewall, p-3rdpartycontrol, nosec, join, x-nortel-sipvc, com.nortelnetworks.im.encryption
Content-Type: application/sdp
Content-Length: 406

v=0
o=PVG 962171386 962171386 IN IP4 148.245.89.102
s=-
p=+1 6135555555
c=IN IP4 0.0.0.0
t=0 0
a=sqn: 0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 9748 RTP/AVP 110 3 0 8 101
c=IN IP4 207.249.182.154
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendonly
<------------->
--- (12 headers 18 lines) ---

He puesto lo mas que me permite el CLI, ojala me puedan ayudar.

**navaismo** · 06-12-2013, 12:30 PM

Sigue igual, haz lo siguiente para recolectar un debug completo:

1.- Entra via ssh a tu equipo
2.- Ejecuta el comando desde el shell de linux: asterisk -rnvvvvd | tee debugsip.txt
3.- Cuando entres al cli de asterisk ejecuta sip set debug on.
4.- Haz la llamada mientras estas viendo el cli, cuelga y espera 5 segundos antes de salir del cli.
5.- Sal del cli con CTRL+C
6.- En el directorio actual de tu sesión ssh existirá un archivo llamado debugsip.txt, sube ese archivo a pastebin.com o adjuntalo aquí.

Opcional: antes de subir o adjuntar el archivo ejecuta el comando sed -i "s:\x1B\[[0-9;]*[mK]::g" debugsip.txt
para eliminar los caracteres de color del archivo de texto.

Saludos.

**Gendo Ikari** · 06-12-2013, 01:43 PM

Aquí tienes amigo, espero que te sirva:

debugsip

http://www.mediafire.com/download/zrupas63hj6y8y6/debugsip.txt

MediaFire is a simple to use free service that lets you put all your photos, documents, music, and video in a single place so you can access them anywhere and share them everywhere.

**navaismo** · 06-12-2013, 02:00 PM

Hola, cuando hagas un debug no enmascares las IPs, si quieres tener seguridad en tu sipdebug que vas a pegar lo que tienes que enmascarar o remover es algunos dígitos del número que marcaste y en los INVITE enmascara/remueve el "nonce", "realm" y "sipuri". Si enmascaras las IPs se vuelve complicado saber qué es lo que pasa algunas veces.

Con la información que adjuntaste lo que te puedo decir es que en realidad si tiene contacto con tu troncal por estas líneas:

Código:

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6c0d781c;rport=5060
From: "soporte_tecnico" <sip:1000XXXXXXXip@x.x.x.x>;tag=as65416877
To: <sip:0445520906991@mex14.axtel.net:5060>
Call-ID: 10c43c6b7c86777b4dfebf0c1687afca@x.x.x.x
CSeq: 103 INVITE
Content-Length: 0

Y

Código:

    -- SIP/63877770-00000021 is making progress passing it to SIP/110-00000020

Lo cual quiere decir que ya envio la peticion a tu peer '63877770'(imagino que es tu "trunk").

Sin embargo te envía un warning:

Código:

[Jun 12 13:20:35] WARNING[3997]: dsp.c:1354 ast_dsp_process: Inband DTMF is not supported on codec speex. Use RFC2833

Cambia ese modo dtmf por el rfc2833 e intentalo de nuevo ya que recibes muchos:

Código:

SIP/2.0 488 Not Acceptable Here

Y después de eso se ve el BYE que creo lo envias tu desde el teléfono al colgar.

**Gendo Ikari** · 06-12-2013, 04:31 PM

Hola Navaismo, gracias por la ayuda, el detalle es que si pongo dtfm=rfc2833 no me detecta los tonos en las llamadas cuando me lo piden las grabaciones. No se a que se deba, no conozco los valores del dtfm, alguna idea?

**navaismo** · 06-12-2013, 04:46 PM

Que codecs estas utilizando en tu línea SIP y en tus peers?

Según el intercambio SDP de tu sip debug:

Código:

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=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x20e (gsm|ulaw|alaw|speex)

Solo puedes soportar de modo combinado: gsm|ulaw|alaw|speex Quita el speex e intenta con gsm o ulaw.

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