[RESUELTO] Llamadas entrantes Axtel no hacen sonar los telefonos

**Christian Cabrera** · 02-04-2014, 03:48 PM

Eso significa que Axtel no está enviando las llamadas a tu servicio de Axtel conmigo, solo a tus líneas análogas.

¿Qué te dice el comando del CLI sip show registry?

**Malagana** · 02-04-2014, 03:54 PM

Hola, Christian

En el "sip show registry" me muestra las lineas de Axtel registradas

Host dnsmgr Username Refresh State Reg.Time
gdl13.axtel.net:5060 N 11004017457i 105 Registered Tue, 04 Feb 2014 16:53:08
gdl11.axtel.net:5060 N 11004026580i 105 Registered Tue, 04 Feb 2014 16:53:08
sip.skype.com:5060 N 991510001674 105 Registered Tue, 04 Feb 2014 16:53:09
gdl11.axtel.net:5060 N 11004026582i 105 Registered Tue, 04 Feb 2014 16:53:09
gdl11.axtel.net:5060 N 11004026581i 105 Registered Tue, 04 Feb 2014 16:53:09

Ya anteriormente y en ocasiones pasa lo que tu mencionas que no se registran las lineas, pero en ese caso no recibo ni salen llamadas.

**Christian Cabrera** · 02-04-2014, 04:51 PM

Me ha pasado también. He encontrado que el servicio de Axtel conmigo es muy inestable, y me imagino que es debido al bajo uso o bien, que está pensado para usuarios residenciales, no para usuarios de Asterisk.

Mis preguntas serían:

¿Has probado configurándolo directamente en un softphone solo para ver si el problema persiste?

En caso de que si funcione en softphone, ¿has probado activando el sip debug al momento en que marcar para recibir una llamada? A lo mejor hay errores a nivel de SIP que no estamos viendo

**Malagana** · 02-04-2014, 05:11 PM

Christian,

Directamente en un softphone funciona sin problemas. Y esto es lo que me sale en el sip debug, pero realmente no estoy familiarizado.

<--- SIP read from UDP:207.249.182.154:5060 --->
INVITE sip:3338178438@172.168.1.10:5060;doroute SIP/2.0
Via: SIP/2.0/UDP 207.249.182.154;branch=z9hG4bK.cNHZ7rXUBjtXg7yKrgv F11mgHa
Via: SIP/2.0/UDP 189.205.62.211:5060;branch=z9hG4bK-35c5b5d-20c4e4ae-4cfc96ce
Record-Route: <sip:207.249.182.154;lr>
Max-Forwards: 18
From: <sip:3339681375@gdl1.axtel.net;user=phone>;tag=592 60
To: "SIPLineUser SIPLineUser" <sip:10004026581ip@gdl1.axtel.net>
Call-ID: 4f609db3_143ff5d616b@GUADSSLI01
CSeq: 66345 INVITE
Contact: <sip:3339681375@207.249.182.154:5060;user=phone;Ct Rtc4bf9b0f64066e2e=udp:189.205.62.211:5060>
User-Agent: Nortel SESM 12.0.12.2
Allow: UPDATE, REFER
Supported: com.nortelnetworks.firewall, p-3rdpartycontrol, nosec, join, x-nortel-sipvc, com.nortelnetworks.im.encryption, replaces, 100rel
Content-Type: application/sdp
Content-Length: 276
remote-party-id: <sip:3339681375@gdl1.axtel.net;user=phone>;screen= yes;party=calling;npi=NPI_E164;ton=TON_NATIONAL
p-asserted-identity: <sip:3339681375@189.205.62.211>
x-nt-corr-id: 10542.003-18-05-46-35@007
x-nt-location: -1
x-nt-service: zoneid=0

v=0
o=PVG 1391557510230 1391557510230 IN IP4 207.249.182.154
s=-
p=+1 6135555555
t=0 0
m=audio 30152 RTP/AVP 100 0 8 101
c=IN IP4 207.249.182.154
a=rtpmap:100 speex/8000
a=fmtp:100 vbr=on
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (20 headers 12 lines) ---
Sending to 207.249.182.154:5060 (NAT)
Using INVITE request as basis request - 4f609db3_143ff5d616b@GUADSSLI01
Found peer 'Axtel-SIP-15882348' for '3339681375' from 207.249.182.154:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 100
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 100
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 - 0x102 (gsm|g729), peer - audio=0x20c (ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

<--- Reliably Transmitting (NAT) to 207.249.182.154:5060 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 207.249.182.154;branch=z9hG4bK.cNHZ7rXUBjtXg7yKrgv F11mgHa;received=207.249.182.154;rport=5060
Via: SIP/2.0/UDP 189.205.62.211:5060;branch=z9hG4bK-35c5b5d-20c4e4ae-4cfc96ce
From: <sip:3339681375@gdl1.axtel.net;user=phone>;tag=592 60
To: "SIPLineUser SIPLineUser" <sip:10004026581ip@gdl1.axtel.net>;tag=as55d78f7 1
Call-ID: 4f609db3_143ff5d616b@GUADSSLI01
CSeq: 66345 INVITE
Server: FPBX-2.10.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '4f609db3_143ff5d616b@GUADSSLI01' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:207.249.182.154:5060 --->
ACK sip:3338178438@172.168.1.10:5060;doroute SIP/2.0
Via: SIP/2.0/UDP 207.249.182.154;branch=z9hG4bK.cNHZ7rXUBjtXg7yKrgv F11mgHa
Max-Forwards: 18
From: <sip:3339681375@gdl1.axtel.net;user=phone>;tag=592 60
To: "SIPLineUser SIPLineUser" <sip:10004026581ip@gdl1.axtel.net>;tag=as55d78f7 1
Call-ID: 4f609db3_143ff5d616b@GUADSSLI01
CSeq: 66345 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4f609db3_143ff5d616b@GUADSSLI01' Method: ACK
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:gdl11.axtel.net SIP/2.0
Via: SIP/2.0/UDP 187.176.67.29:5060;branch=z9hG4bK610fb11e;rport
Max-Forwards: 70
From: "Unknown" <sip:10004026580ip@187.176.67.29>;tag=as416c6284
To: <sip:gdl11.axtel.net>
Contact: <sip:10004026580ip@187.176.67.29:5060>
Call-ID: 1173ca3d09b096ca0dd63d477c16f50a@187.176.67.29:506 0
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Wed, 05 Feb 2014 00:06:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:gdl11.axtel.net SIP/2.0
Via: SIP/2.0/UDP 187.176.67.29:5060;branch=z9hG4bK13f18bfc;rport
Max-Forwards: 70
From: "Unknown" <sip:10004026582ip@187.176.67.29>;tag=as6f04c4a0
To: <sip:gdl11.axtel.net>
Contact: <sip:10004026582ip@187.176.67.29:5060>
Call-ID: 4fc5a9be51a874ff4b98057e41e2a70b@187.176.67.29:506 0
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Wed, 05 Feb 2014 00:06:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:gdl13.axtel.net SIP/2.0
Via: SIP/2.0/UDP 187.176.67.29:5060;branch=z9hG4bK3aa7d94d;rport
Max-Forwards: 70
From: "Unknown" <sip:10004017457ip@187.176.67.29>;tag=as5af7858a
To: <sip:gdl13.axtel.net>
Contact: <sip:10004017457ip@187.176.67.29:5060>
Call-ID: 32aa2b3f47d591b94704594e4ce96eb4@187.176.67.29:506 0
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Wed, 05 Feb 2014 00:06:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---
Reliably Transmitting (NAT) to 207.249.182.154:5060:
OPTIONS sip:gdl11.axtel.net SIP/2.0
Via: SIP/2.0/UDP 187.176.67.29:5060;branch=z9hG4bK495cb81f;rport
Max-Forwards: 70
From: "Unknown" <sip:10004026581ip@187.176.67.29>;tag=as3fd7c796
To: <sip:gdl11.axtel.net>
Contact: <sip:10004026581ip@187.176.67.29:5060>
Call-ID: 55ea891d03f5321251df4e5936f78a00@187.176.67.29:506 0
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Wed, 05 Feb 2014 00:06:05 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 172.168.1.10:5060;branch=z9hG4bK610fb11e;rport=506 0
From: "Unknown" <sip:10004026580ip@172.168.1.10:5060>;tag=as416c62 84
To: <sip:gdl11.axtel.net>;tag=my8Bm4mcgF17D
Call-ID: 1173ca3d09b096ca0dd63d477c16f50a@187.176.67.29:506 0
CSeq: 102 OPTIONS
Contact: <sip:10004026580ip@172.168.1.10:5060>
Server: Flexisip/0.6.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1173ca3d09b096ca0dd63d477c16f50a@187.176.67.29:50 60' Method: OPTIONS

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.168.1.10:5060;branch=z9hG4bK13f18bfc;rport=506 0
From: "Unknown" <sip:10004026582ip@172.168.1.10:5060>;tag=as6f04c4 a0
To: <sip:gdl11.axtel.net>;tag=N714NZ5FDrQtS
Call-ID: 4fc5a9be51a874ff4b98057e41e2a70b@187.176.67.29:506 0
CSeq: 102 OPTIONS
Contact: <sip:10004026582ip@172.168.1.10:5060>
Server: Flexisip/0.6.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '4fc5a9be51a874ff4b98057e41e2a70b@187.176.67.29:50 60' Method: OPTIONS

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.168.1.10:5060;branch=z9hG4bK3aa7d94d;rport=506 0
From: "Unknown" <sip:10004017457ip@172.168.1.10:5060>;tag=as5af785 8a
To: <sip:gdl13.axtel.net>;tag=pgUXQtpKa1DDN
Call-ID: 32aa2b3f47d591b94704594e4ce96eb4@187.176.67.29:506 0
CSeq: 102 OPTIONS
Contact: <sip:10004017457ip@172.168.1.10:5060>
Server: Flexisip/0.6.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '32aa2b3f47d591b94704594e4ce96eb4@187.176.67.29:50 60' Method: OPTIONS

<--- SIP read from UDP:207.249.182.154:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 172.168.1.10:5060;branch=z9hG4bK495cb81f;rport=506 0
From: "Unknown" <sip:10004026581ip@172.168.1.10:5060>;tag=as3fd7c7 96
To: <sip:gdl11.axtel.net>;tag=QSmpSN7p793Zg
Call-ID: 55ea891d03f5321251df4e5936f78a00@187.176.67.29:506 0
CSeq: 102 OPTIONS
Contact: <sip:10004026581ip@172.168.1.10:5060>
Server: Flexisip/0.6.0 (sofia-sip-nta/2.0)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '55ea891d03f5321251df4e5936f78a00@187.176.67.29:50 60' Method: OPTIONS
Reliably Transmitting (NAT) to 63.209.144.201:5060:
OPTIONS sip:sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 187.176.67.29:5060;branch=z9hG4bK0308590e;rport
Max-Forwards: 70
From: "Unknown" <sip:99051000167470@187.176.67.29>;tag=as6b0f91d a
To: <sip:sip.skype.com>
Contact: <sip:99051000167470@187.176.67.29:5060>
Call-ID: 385a7b233d708f5e05ad34683b696e2b@187.176.67.29:506 0
CSeq: 102 OPTIONS
User-Agent: FPBX-2.10.1(1.8.13.0)
Date: Wed, 05 Feb 2014 00:06:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

---

<--- SIP read from UDP:63.209.144.201:5060 --->
SIP/2.0 200 OK
From: "Unknown" <sip:99051000167470@172.168.1.10:5060>;tag=as6b0f9 1da
To: <sip:sip.skype.com>;tag=c990d13f-9cd3b47-0-ff5d7c01-0
Call-ID: 385a7b233d708f5e05ad34683b696e2b@187.176.67.29:506 0
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.168.1.10:5060;branch=z9hG4bK0308590e;rport=506 0
Content-Length: 0

**Marco Zink** · 02-04-2014, 06:57 PM

Parece que es un problema de codecs:

Código:

Capabilities: us - 0x102 (gsm|g729), peer - audio=0x20c (ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

Por lo que entiendo tu tienes sólo permitido gsm y g729 pero axtel tiene ulaw, alaw y speex.

Intenta habilitar el ulaw y alaw en tu troncal e intenta de nuevo.

Código:

disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm

**Christian Cabrera** · 02-04-2014, 07:07 PM

Marco y yo tenemos el caso en el cual uno de los dos trabaja la respuesta y el otro da la estocada final. Al parecer me ganó la respuesta en esta ocasión.

Como te comenta, el problema es de codecs. Según el debug que pones, esta linea te da la respuesta:

Capabilities: us - 0x102 (gsm|g729), peer - audio=0x20c (ulaw|alaw|speex)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)

El carrier quiere hablar ulaw, alaw o speex, mientras que tu solo quieres hablar gsm y g729. Habilita cualquiera de los que el carrier quiere y deben empezar a pasar las llamadas.

Saludos,

**Malagana** · 02-04-2014, 08:07 PM

Marco y Christian, tienen todo la razón en lo que me indican, de esa manera ya volvieron a comenzar a entrar las llamadas. Basto con poner

Código:

allow=ulaw&alaw&gsm&g729

en cada configuración SIP y listo. Lo curioso del caso, es que solía funcionar sin los otros codecs.

Muchas gracias por su apoyo, hoy aprendí algo nuevo gracias a ustedes.

[RESUELTO] Llamadas entrantes Axtel no hacen sonar los telefonos

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