Problemas DTMF en Trunk SIP Asterisk vs OXE

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Visin

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Apr 23, 2015, 3:18:33 PM4/23/15
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Hola,

Tengo enlazado por trunk SIP un Asterisk contra una OXE. Las llamadas entran por el trunk SIP desde la OXE al Asterisk y en el Asterisk tengo un IVR y no detecta los tonos DTMF.


<--- SIP read from UDP:X.X.X.X:5060 --->
INVITE sip:10...@X.X.X.X;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Supported: histinfo,replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 
Session-Expires: 1800;refresher=uac
Min-SE: 180
P-Asserted-Identity: "NGN RED" <sip:6XXX...@X.X.X.X;user=phone>
Content-Type: application/sdp
To: <sip:10...@X.X.X.X;user=phone>
From: "NGN RED" <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
Contact: <sip:6XXX...@X.X.X.X;transport=UDP>
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332116 INVITE
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1d3ef5eec0e3a395e274be269bfb2429
Max-Forwards: 70
Content-Length:284

v=0
o=OXE 1429801636 1429801636 IN IP4 X.X.X.X
s=abs
c=IN IP4 X.X.X.X
t=0 0
m=audio 11418 RTP/AVP 18 8 97
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:30
a=maxptime:40
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:97 telephone-event/8000
<------------->
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: --- (16 headers 15 lines) ---
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Sending to X.X.X.X:5060 (no NAT)
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Using INVITE request as basis request - 77a265044c717268...@X.X.X.X
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found peer 'TRUNK' for '6XXXXXXXX' from X.X.X.X:5060
[2015-04-23 17:07:16] VERBOSE[13331] netsock2.c:   == Using SIP RTP TOS bits 184
[2015-04-23 17:07:16] VERBOSE[13331] netsock2.c:   == Using SIP RTP CoS mark 5
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found RTP audio format 18
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found RTP audio format 8
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found RTP audio format 97
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found audio description format G729 for ID 18
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found audio description format PCMA for ID 8
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Found audio description format telephone-event for ID 97
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Peer audio RTP is at port X.X.X.X:11418
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: Looking for 10550 in from-trunk (domain X.X.X.X)
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: list_route: hop: <sip:6XXX...@X.X.X.X;transport=UDP>
[2015-04-23 17:07:16] VERBOSE[13331] chan_sip.c: 
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1d3ef5eec0e3a395e274be269bfb2429;received=X.X.X.X
From: "NGN RED" <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
To: <sip:10...@X.X.X.X;user=phone>
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332116 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.X:5060>
Content-Length: 0

[2015-04-23 17:07:17] VERBOSE[24653] pbx.c:     -- Executing [s@citas:6] Answer("SIP/TRUNK-00000031", "") in new stack
[2015-04-23 17:07:17] VERBOSE[24653] chan_sip.c: Audio is at 14682
[2015-04-23 17:07:17] VERBOSE[24653] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2015-04-23 17:07:17] VERBOSE[24653] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-04-23 17:07:17] VERBOSE[24653] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1d3ef5eec0e3a395e274be269bfb2429;received=X.X.X.X
From: "NGN RED" <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332116 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.X:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 236

v=0
o=root 1863362771 1863362771 IN IP4 X.X.X.X
s=Asterisk PBX 1.8.32.3
c=IN IP4 X.X.X.X
t=0 0
m=audio 14682 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

<------------>
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.X:5060 --->
ACK sip:10...@X.X.X.X:5060 SIP/2.0
Contact: sip:6XXX...@X.X.X.X
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
From: "NGN RED" <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332116 ACK
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK893ec2ad2147f82f84b0f9ef5a04c1a7
Max-Forwards: 70
Content-Length: 0

<------------->
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: --- (10 headers 0 lines) ---
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.X:5060 --->
INVITE sip:10...@X.X.X.X:5060 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:6XXX...@X.X.X.X
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
Session-Expires: 1800;refresher=uac
Min-SE: 180
Content-Type: application/sdp
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
From: <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332117 INVITE
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1e4fe0e17199c7298c90b3ab774971d9
Max-Forwards: 70
Content-Length: 214

v=0
o=OXE 1429801636 1429801637 IN IP4 X.X.X.X
s=abs
c=IN IP4 X.X.X.X
t=0 0
m=audio 64374 RTP/AVP 8 97
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:97 telephone-event/8000
<------------->
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: --- (15 headers 11 lines) ---
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Sending to X.X.X.X:5060 (no NAT)
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Found RTP audio format 8
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Found RTP audio format 97
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Found audio description format PCMA for ID 8
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Found audio description format telephone-event for ID 97
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Peer audio RTP is at port X.X.X.X:64374
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: 
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1e4fe0e17199c7298c90b3ab774971d9;received=X.X.X.X
From: <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332117 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.X:5060>
Content-Length: 0


<------------>
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Audio is at 14682
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK1e4fe0e17199c7298c90b3ab774971d9;received=X.X.X.X
From: <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332117 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.X:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 236

v=0
o=root 1863362771 1863362772 IN IP4 X.X.X.X
s=Asterisk PBX 1.8.32.3
c=IN IP4 X.X.X.X
t=0 0
m=audio 14682 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=ptime:20
a=sendrecv

<------------>
[2015-04-23 17:07:17] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.X:5060 --->
ACK sip:10...@X.X.X.X:5060 SIP/2.0
Contact: sip:6XXX...@X.X.X.X
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
To: <sip:10...@X.X.X.X;user=phone>;tag=as5f583b93
From: <sip:6XXX...@X.X.X.X;user=phone>;tag=b6224009174e610ef0bb71c91723fdfd
Call-ID: 77a265044c717268...@X.X.X.X
CSeq: 837332117 ACK
Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0296ad75b111d4861025c5007e6e394b
Max-Forwards: 70
Content-Length: 0

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



He borrado las IPs, para ver "quien es quien" dejo el User-agent.


Negocia el codec y el dtmf ... y acuerdan poner el dtmf en payload 97, pero no funciona. Si pongo música la escucho sin problema, por lo que entiendo que el RTP circula bien, por descartar problemas de firewall.


Ya sabeis como va esto, la empresa de la OXE dice que ellos lo tienen todo bien y que el problema debe ser de mi Asterisk. Ahora me toca demostrar que no tiene razón (o si), dando con la solución.


¿A alguien se le ocurre que más puedo probar?.



Muchas gracias, un saludo.

Fernando Villares

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Apr 23, 2015, 11:17:57 PM4/23/15
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En asterisk el payload no es 97 es 101 por eso no te funciona...tenes q o poner inband o cambiar el payload de todo en la pbx y fonos a 97

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Raúl Alexis Betancor Santana

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Apr 24, 2015, 1:08:57 AM4/24/15
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El playload se negocia ... por lo menos en eso lo hace bien Asterisk ... ahora que estoy seguro casi al 2000%, de que la OXE le importa un carajo y lo ignora, con lo que el consejo de Fernando es muy válido.


Visin

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Apr 24, 2015, 4:38:38 AM4/24/15
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<--- SIP read from UDP:X.X.X.77:5060 --->
INVITE sip:10...@X.X.X.235;user=phone SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE, INFO
Supported: histinfo,replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
Session-Expires: 1800;refresher=uac
Min-SE: 180
P-Asserted-Identity: "NGN RED" <sip:6XXX...@X.X.X.77;user=phone>
Content-Type: application/sdp
To: <sip:10...@X.X.X.235;user=phone>
From: "NGN RED" <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
Contact: <sip:6XXX...@X.X.X.77;transport=UDP>
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253469 INVITE
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bK8c8a880ed1eedf4c0e2f0326005bbf0b
Max-Forwards: 70
Content-Length:292

v=0
o=OXE 1429863381 1429863381 IN IP4 X.X.X.14
s=abs
c=IN IP4 X.X.X.14
t=0 0
m=audio 10626 RTP/AVP 18 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:30
a=maxptime:40
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:101 telephone-event/8000

<------------->
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: --- (16 headers 15 lines) ---
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Sending to X.X.X.77:5060 (no NAT)
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Using INVITE request as basis request - e93bbc6725df1fe0...@X.X.X.77
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found peer 'TRUNK' for '6XXXXXXXX' from X.X.X.77:5060
[2015-04-24 10:16:21] VERBOSE[13331] netsock2.c:   == Using SIP RTP TOS bits 184
[2015-04-24 10:16:21] VERBOSE[13331] netsock2.c:   == Using SIP RTP CoS mark 5
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found RTP audio format 18
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found RTP audio format 8
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found RTP audio format 101
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found audio description format G729 for ID 18
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found audio description format PCMA for ID 8
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Found audio description format telephone-event for ID 101
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x108 (alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Peer audio RTP is at port X.X.X.14:10626
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: Looking for 10550 in from-trunk (domain X.X.X.235)
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: list_route: hop: <sip:6XXX...@X.X.X.77;transport=UDP>
[2015-04-24 10:16:21] VERBOSE[13331] chan_sip.c: 
<--- Transmitting (no NAT) to X.X.X.77:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bK8c8a880ed1eedf4c0e2f0326005bbf0b;received=X.X.X.77
From: "NGN RED" <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
To: <sip:10...@X.X.X.235;user=phone>
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253469 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.235:5060>
Content-Length: 0


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

[2015-04-24 10:16:26] VERBOSE[19816] pbx.c:     -- Executing [s@citas:7] Answer("SIP/TRUNK-00000048", "") in new stack
[2015-04-24 10:16:26] VERBOSE[19816] chan_sip.c: Audio is at 15932
[2015-04-24 10:16:26] VERBOSE[19816] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2015-04-24 10:16:26] VERBOSE[19816] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-04-24 10:16:26] VERBOSE[19816] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to X.X.X.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bK8c8a880ed1eedf4c0e2f0326005bbf0b;received=X.X.X.77
From: "NGN RED" <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253469 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.235:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 239

v=0
o=root 1466440538 1466440538 IN IP4 X.X.X.235
s=Asterisk PBX 1.8.32.3
c=IN IP4 X.X.X.235
t=0 0
m=audio 15932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.77:5060 --->
ACK sip:10...@X.X.X.235:5060 SIP/2.0
Contact: sip:6XXX...@X.X.X.77
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
From: "NGN RED" <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253469 ACK
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bK3eea79fe931abefb79e572532f43e7bc
Max-Forwards: 70
Content-Length: 0

<------------->
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: --- (10 headers 0 lines) ---
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.77:5060 --->
INVITE sip:10...@X.X.X.235:5060 SIP/2.0
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER, SUBSCRIBE, OPTIONS, UPDATE
Contact: sip:6XXX...@X.X.X.77
Supported: replaces,timer,path,100rel
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
Session-Expires: 1800;refresher=uac
Min-SE: 180
Content-Type: application/sdp
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
From: <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253470 INVITE
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bKb67ab38b30e0250de11dc5543e276c0c
Max-Forwards: 70
Content-Length: 216

v=0
o=OXE 1429863381 1429863382 IN IP4 X.X.X.77
s=abs
c=IN IP4 X.X.X.77
t=0 0
m=audio 64452 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=maxptime:30
a=rtpmap:101 telephone-event/8000
<------------->
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: --- (15 headers 11 lines) ---
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Sending to X.X.X.77:5060 (no NAT)
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Found RTP audio format 8
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Found RTP audio format 101
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Found audio description format PCMA for ID 8
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Found audio description format telephone-event for ID 101
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Peer audio RTP is at port X.X.X.77:64452
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: 
<--- Transmitting (no NAT) to X.X.X.77:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bKb67ab38b30e0250de11dc5543e276c0c;received=X.X.X.77
From: <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253470 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.235:5060>
Content-Length: 0


<------------>
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Audio is at 15932
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Adding codec 0x8 (alaw) to SDP
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to X.X.X.77:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bKb67ab38b30e0250de11dc5543e276c0c;received=X.X.X.77
From: <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253470 INVITE
Server: FPBX-12.0.54(1.8.32.3)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Contact: <sip:10...@X.X.X.235:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 239

v=0
o=root 1466440538 1466440539 IN IP4 X.X.X.235
s=Asterisk PBX 1.8.32.3
c=IN IP4 X.X.X.235
t=0 0
m=audio 15932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[2015-04-24 10:16:26] VERBOSE[13331] chan_sip.c: 
<--- SIP read from UDP:X.X.X.77:5060 --->
ACK sip:10...@X.X.X.235:5060 SIP/2.0
Contact: sip:6XXX...@X.X.X.77
User-Agent: OmniPCX Enterprise R10.0 j1.410.49.a
To: <sip:10...@X.X.X.235;user=phone>;tag=as7cde3a58
From: <sip:6XXX...@X.X.X.77;user=phone>;tag=0c1062a51132c7ccf3f413202e9f0d74
Call-ID: e93bbc6725df1fe0...@X.X.X.77
CSeq: 1007253470 ACK
Via: SIP/2.0/UDP X.X.X.77;branch=z9hG4bK7e8e52700ef67cc399bb522ad5a80856
Max-Forwards: 70
Content-Length: 0

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



Los del extremo de OXE, me han cambiado el payload a 101. Ahora la OXE me envía en el INVITE el payload 101 y el asterisk da el OK (Antes en el INVITE de OXE ponia el payload de DTMF a 97 y Asterisk daba el OK tb). Pero sigue sin detectar la pulsación de teclas.



Fernando Villares

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Apr 24, 2015, 4:51:12 AM4/24/15
to aster...@googlegroups.com

Haceme caso y pone inband en esa troncal....vas a ver...

Visin

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Apr 24, 2015, 5:55:41 AM4/24/15
to aster...@googlegroups.com
con dftmfmode=inband en la configuración del trunk no me funciona el reconocimiento de teclas tampoco.


:(


Fernando Villares

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Apr 24, 2015, 9:45:55 AM4/24/15
to aster...@googlegroups.com
Si con inband en alaw no funciona entonces el,problema va por otro lado ya...

Enviado desde mi iPad

El 24/04/2015, a las 06:55, Visin <visi...@gmail.com> escribió:

con dftmfmode=inband en la configuración del trunk no me funciona el reconocimiento de teclas tampoco.


:(


Visin

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Apr 24, 2015, 12:58:38 PM4/24/15
to aster...@googlegroups.com
Ya funciona, con rfc2833


¿Qué era?:

Ejecutaba RInging() antes de Answer. No entiendo por qué, he quitado el Ringing y ya detecta los dtmf.


¿Alguien sabe por qué pasaba esto?.



Un saludo.

Raúl Alexis Betancor Santana

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Apr 26, 2015, 6:04:03 PM4/26/15
to aster...@googlegroups.com
El problema no es la diferencia del ID de playload ...

Lo mejor es que hagas una captura de los RTP ... y veas si los está enviado o nó, con casi toda seguridad, la OXE no los está enviando o peor ... los envía como le sale del pito a pesar de que le digas inband o RFC2843

De: "Fernando Villares" <fvil...@gmail.com>
Para: aster...@googlegroups.com
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