Hola a todos..

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...

captura de tshark... el de Openser:

Internet Protocol, Src: 192.168.1.6 (192.168.1.6), Dst: 192.168.1.116 ( 192.168.1.116) User Datagram Protocol, Src Port: 8342 (8342), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header To: sip:8889991@my.domain.com;user=phone;tag=5636e002 From: 8888888sip:8888888@my.domain.com;user=phone;tag=6Scf2-18Yzu0 Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK486c.f39169b5.0 ;received=192.168.1.116 Via: SIP/2.0/UDP 192.168.0.55:5060;rport=33654;received= 192.168.1.240;branch=z9hG4bKwE0f2-W2w7sRiu Call-ID: 9RqF70-3dT0sf2@my.domain.com CSeq: 103 INVITE Record-Route: sip:192.168.1.116;lr=on;ftag=6Scf2-18Yzu0 Contact: sip:8889991@192.168.1.180:8342;transport=udp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Content-Length: 285 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 15858836 15859206 IN IP4 192.168.1.180 Session Name (s): eyeBeam Connection Information (c): IN IP4 192.168.1.180 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 8360 RTP/AVP 18 101 Media Type: audio Media Port: 8360 Media Proto: RTP/AVP Media Format: ITU-T G.729 Media Format: 101 Media Attribute (a): alt:1 1 : 3E824724 5CE3B7E1 192.168.1.1808360 Media Attribute Fieldname: alt Media Attribute Value: 1 1 : 3E824724 5CE3B7E1 192.168.1.1808360 Media Attribute (a): alt:2 3 : 65F4C37E 6639080D 192.168.1.68360 Media Attribute Fieldname: alt Media Attribute Value: 2 3 : 65F4C37E 6639080D 192.168.1.68360 Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Format: 101 Media format specific parameters: 0-15 Media Attribute (a): sendrecv

el de asterisk:

Internet Protocol, Src: 192.168.1.116 (192.168.1.116), Dst: 192.168.0.67 ( 192.168.0.67) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.0.67:5060;rport=33550;received= 192.168.1.109;branch=z9hG4bK6F0f2-gSHit0Vlv Record-Route: sip:192.168.1.116;lr=on;ftag=gVuf2-SJ*GZ0 From: 8888888sip:8888888@my.domain.com;user=phone;tag=gVuf2-SJ*GZ0 To: sip:8889991@my.domain.com;user=phone;tag=as7e0858ab Call-ID: 5dyHK-dgl020f2@my.domain.com CSeq: 102 INVITE User-Agent: CityMoon SIP/1.8.0.004 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:8889991@192.168.1.111:5099 Content-Type: application/sdp Content-Length: 263 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 7913 7914 IN IP4 192.168.1.111 Session Name (s): session Connection Information (c): IN IP4 192.168.1.111 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13124 RTP/AVP 18 101 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute Fieldname: rtpmap Media Format: 18 MIME Type: G729 Media Attribute (a): fmtp:18 annexb=no Media Attribute Fieldname: fmtp Media Format: 18 [G729] Media format specific parameters: annexb=no Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv

Alguna idea de porqué es TAN distinto???

Saludos