<div>Hola a todos..</div>
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<div>Un pregunta un poco cochambrosa... tengo un ATA que cuando la llamada pasa por asterisk, se que congelado, y si va por openser funciona perfecto... se muere por lo "200 OK" que manda asterisk... el que recibe de Openser, es COMPLETAMENTE distinto...</div>
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<div>captura de tshark...</div>
<div>el de Openser:</div>
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<div>Internet Protocol, Src: <a href="http://192.168.1.6">192.168.1.6</a> (<a href="http://192.168.1.6">192.168.1.6</a>), Dst: <a href="http://192.168.1.116">192.168.1.116</a> (<a href="http://192.168.1.116">192.168.1.116</a>)<br>
User Datagram Protocol, Src Port: 8342 (8342), Dst Port: 5060 (5060)<br>Session Initiation Protocol<br> Status-Line: SIP/2.0 200 OK<br> Message Header<br> To: <<a href="mailto:sip:8889991@my.domain.com">sip:8889991@my.domain.com</a>;user=phone>;tag=5636e002<br>
From: 8888888<<a href="mailto:sip:8888888@my.domain.com">sip:8888888@my.domain.com</a>;user=phone>;tag=6Scf2-18Yzu0<br> Via: SIP/2.0/UDP <a href="http://192.168.1.116">192.168.1.116</a>;branch=z9hG4bK486c.f39169b5.0;received=<a href="http://192.168.1.116">192.168.1.116</a><br>
Via: SIP/2.0/UDP <a href="http://192.168.0.55:5060">192.168.0.55:5060</a>;rport=33654;received=<a href="http://192.168.1.240">192.168.1.240</a>;branch=z9hG4bKwE0f2-W2w7sRiu<br> Call-ID: <a href="mailto:9RqF70-3dT0sf2@my.domain.com">9RqF70-3dT0sf2@my.domain.com</a><br>
CSeq: 103 INVITE<br> Record-Route: <sip:<a href="http://192.168.1.116">192.168.1.116</a>;lr=on;ftag=6Scf2-18Yzu0><br> Contact: <sip:8889991@192.168.1.180:8342;transport=udp><br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br> Content-Length: 285<br> Message body<br> Session Description Protocol<br> Session Description Protocol Version (v): 0<br> Owner/Creator, Session Id (o): - 15858836 15859206 IN IP4 <a href="http://192.168.1.180">192.168.1.180</a><br>
Session Name (s): eyeBeam<br> Connection Information (c): IN IP4 <a href="http://192.168.1.180">192.168.1.180</a><br> Time Description, active time (t): 0 0<br> Media Description, name and address (m): audio 8360 RTP/AVP 18 101<br>
Media Type: audio<br> Media Port: 8360<br> Media Proto: RTP/AVP<br> Media Format: ITU-T G.729<br> Media Format: 101<br> Media Attribute (a): alt:1 1 : 3E824724 5CE3B7E1 <a href="http://192.168.1.180">192.168.1.180</a> 8360<br>
Media Attribute Fieldname: alt<br> Media Attribute Value: 1 1 : 3E824724 5CE3B7E1 <a href="http://192.168.1.180">192.168.1.180</a> 8360<br> Media Attribute (a): alt:2 3 : 65F4C37E 6639080D <a href="http://192.168.1.6">192.168.1.6</a> 8360<br>
Media Attribute Fieldname: alt<br> Media Attribute Value: 2 3 : 65F4C37E 6639080D <a href="http://192.168.1.6">192.168.1.6</a> 8360<br> Media Attribute (a): fmtp:101 0-15<br> Media Attribute Fieldname: fmtp<br>
Media Format: 101<br> Media format specific parameters: 0-15<br> Media Attribute (a): sendrecv</div>
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<div>el de asterisk:</div>
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<div>Internet Protocol, Src: <a href="http://192.168.1.116">192.168.1.116</a> (<a href="http://192.168.1.116">192.168.1.116</a>), Dst: <a href="http://192.168.0.67">192.168.0.67</a> (<a href="http://192.168.0.67">192.168.0.67</a>)<br>
User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060)<br>Session Initiation Protocol<br> Status-Line: SIP/2.0 200 OK<br> Message Header<br> Via: SIP/2.0/UDP <a href="http://192.168.0.67:5060">192.168.0.67:5060</a>;rport=33550;received=<a href="http://192.168.1.109">192.168.1.109</a>;branch=z9hG4bK6F0f2-gSHit0Vlv<br>
Record-Route: <sip:<a href="http://192.168.1.116">192.168.1.116</a>;lr=on;ftag=gVuf2-SJ*GZ0><br> From: 8888888<<a href="mailto:sip:8888888@my.domain.com">sip:8888888@my.domain.com</a>;user=phone>;tag=gVuf2-SJ*GZ0<br>
To: <<a href="mailto:sip:8889991@my.domain.com">sip:8889991@my.domain.com</a>;user=phone>;tag=as7e0858ab<br> Call-ID: <a href="mailto:5dyHK-dgl020f2@my.domain.com">5dyHK-dgl020f2@my.domain.com</a><br>
CSeq: 102 INVITE<br> User-Agent: CityMoon SIP/1.8.0.004<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br> Supported: replaces<br> Contact: <sip:8889991@192.168.1.111:5099><br>
Content-Type: application/sdp<br> Content-Length: 263<br> Message body<br> Session Description Protocol<br> Session Description Protocol Version (v): 0<br> Owner/Creator, Session Id (o): root 7913 7914 IN IP4 <a href="http://192.168.1.111">192.168.1.111</a><br>
Session Name (s): session<br> Connection Information (c): IN IP4 <a href="http://192.168.1.111">192.168.1.111</a><br> Time Description, active time (t): 0 0<br> Media Description, name and address (m): audio 13124 RTP/AVP 18 101<br>
Media Attribute (a): rtpmap:18 G729/8000<br> Media Attribute Fieldname: rtpmap<br> Media Format: 18<br> MIME Type: G729<br> Media Attribute (a): fmtp:18 annexb=no<br>
Media Attribute Fieldname: fmtp<br> Media Format: 18 [G729]<br> Media format specific parameters: annexb=no<br> Media Attribute (a): rtpmap:101 telephone-event/8000<br>
Media Attribute Fieldname: rtpmap<br> Media Format: 101<br> MIME Type: telephone-event<br> Media Attribute (a): fmtp:101 0-16<br> Media Attribute Fieldname: fmtp<br>
Media Format: 101 [telephone-event]<br> Media format specific parameters: 0-16<br> Media Attribute (a): silenceSupp:off - - - -<br> Media Attribute Fieldname: silenceSupp<br>
Media Attribute Value: off - - - -<br> Media Attribute (a): ptime:20<br> Media Attribute Fieldname: ptime<br> Media Attribute Value: 20<br> Media Attribute (a): sendrecv<br>
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<div>Alguna idea de porqué es TAN distinto???</div>
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<div>Saludos</div>