[asterisk-users] Theoretical and Received SIP addresses causing no
audio
Gavin Henry
gavin.henry at gmail.com
Tue May 29 11:19:29 MST 2007
Hi,
This contacted call has no audio, any ideas?
The conference suite from another provider on internal IP is waiting
for an ACK on port 5605, but * is sending it back to port 2289
Internal between Asterisk and another Conference suite:
* SIP Call
Direction: Outgoing
Call-ID: 0a5d773327e8d067370fa4da03097e58 at 192.168.45.196
Our Codec Capability: 14
Non-Codec Capability: 1
Their Codec Capability: 4
Joint Codec Capability: 4
Format ulaw
Theoretical Address: 192.168.45.183:5605
Received Address: 192.168.45.183:2289
NAT Support: Always
Audio IP: 192.168.45.196 (local)
Our Tag: as31c610d6
Their Tag: t1122b
SIP User agent:
Username: slee
Peername: slee
Original uri: sip:jNetX at 192.168.45.183:5605
Need Destroy: 0
Last Message: Tx: ACK
Promiscuous Redir: No
Route: sip:jNetX at 192.168.45.183:5605
DTMF Mode: rfc2833
SIP Options: (none)
Inbound from SIP Provider:
* SIP Call
Direction: Incoming
Call-ID: 7f73070762b13c4f2445c16f32b80c4b at xx.xx.xx.xx
<------ REMOVED
Our Codec Capability: 14
Non-Codec Capability: 1
Their Codec Capability: 14
Joint Codec Capability: 14
Format gsm
Theoretical Address: 193.111.201.32:5060
Received Address: 193.111.201.32:5060
NAT Support: Always
Audio IP: xx.xx.xx.xx (local)
<------ REMOVED
Our Tag: as65c31c43
Their Tag: as26378dd7
SIP User agent: Asterisk PBX
Original uri: sip:01XXXXXXXXX at xx.xx.xx.xx <------ REMOVED
Caller-ID: 01XXXXXXXXX
<------ REMOVED
Need Destroy: 0
Last Message: Rx: ACK
Promiscuous Redir: No
Route: sip:193.111.201.32;lr=on;ftag=as26378dd7
DTMF Mode: rfc2833
SIP Options: (none)
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