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: 8888888<sip: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(a)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: 8888888<sip:8888888@my.domain.com;user=phone>;tag=gVuf2-SJ*GZ0
To: <sip:8889991@my.domain.com;user=phone>;tag=as7e0858ab
Call-ID: 5dyHK-dgl020f2(a)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