Hey all -- thanks in advanced for even reading the post.
We just ordered a few SIP trunks from Bandwidth and have our vendor configuring the trunks with our 3300 PBX.
Up to this point, we are still having issues with one way audio -- they can hear us, we cannot hear them. Test calls are usually placed from my teleworker phone and captures have been done using port mirroring and Wireshark.
Our PBX is NAT'ed behind a Cisco ASA security appliance, which is SIP aware. From what I can see, it looks like our firewall is properly handling the translations/etc to/from Bandwidth's SIP proxy.
Bandwidth is reporting a ‘Not Acceptable’ from the PBX, from what they have said.
VoIP Flow:
|Time | 10.1.2.2 | 68.xxx.xxx.xxx |
| | | 216.82.xxx.xxx |
|0.000000000| INVITE | | |SIP From: "C.User" <sip:+281xxxxxxx@10.1.2.2 To:<sip:+1281xxxxxxx@216.82.xxx.xxx
| |(5060) ------------------> (5060) | |
|0.000449000| | INVITE | |SIP Request
| | |(5060) <------------------ (5060) |
|0.097827000| | 100 Giving a try |SIP Status
| | |(5060) ------------------> (5060) |
|0.097941000| 100 Giving a try | |SIP Status
| |(5060) <------------------ (5060) | |
|4.205087000| | 183 Session Progress |SIP Status
| | |(5060) ------------------> (5060) |
|4.205356000| 183 Session Progress | |SIP Status
| |(5060) <------------------ (5060) | |
|19.623845000| | 200 OK SDP (g711U telephone-eventRTPType-101) |SIP Status
| | |(5060) ------------------> (5060) |
|19.624424000| 200 OK SDP (g711U telephone-eventRTPType-101) | |SIP Status
| |(5060) <------------------ (5060) | |
|19.672179000| ACK SDP (g711U telephone-eventRTPType-101) | |SIP Request
| |(5060) ------------------> (5060) | |
|19.672840000| | ACK SDP (g711U telephone-eventRTPType-101) |SIP Request
| | |(5060) <------------------ (5060) |
|26.235692000| BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|26.236031000| | BYE | |SIP Request
| | |(5060) <------------------ (5060) |
|26.346427000| | 200 OK | |SIP Status
| | |(5060) ------------------> (5060) |
|26.346695000| 200 OK | | |SIP Status
| |(5060) <------------------ (5060) | |
Here is the trace from Bandwidth support:
U 2012/11/06 20:32:48.834529 68.xxx.xxx.xxx:5060 -> 216.82.xxx.xxx:5060
SIP/2.0 406 Not Acceptable.
Via: SIP/2.0/UDP 216.82.xxx.xxx;branch=z9hG4bKb2b.d003737.0,SIP/2.0/UDP 67.231.xxx.xxx;branch=z9hG4bKb2b.5dbb3216.3,SIP/2.0/UDP 192.168.37.68:5060;branch=z9hG4bK0bB86c148cbb1cb7710.
Record-Route: <sip:216.82.xxx.xxx;lr;ftag=gK0b07b3aa>,<sip:67.231.xxx.xxx;lr=on;ftag=gK0b07b3aa>.
Warning: 399 10.1.2.2 "19H Reject".
From: <sip:+1919xxxxxxx@192.168.37.68;isup-oli=0>;tag=gK0b07b3aa.
To: <sip:+1281xxxxxxx@67.231.xxx.xxx>;tag=0_2240970064-98773126.
Call-ID: 1863043143_16435119@192.168.37.68.
CSeq: 17638 INVITE.
Contact: <sip:68.xxx.xxx.xxx>.
Server: Mitel-3300-ICP 11.0.1.26.
Content-Length: 0.
Happy Thanksgiving!