SIP Server - problem with 2 step transfer calls - mute audio

[cavagnaro]

Hi guys,
I have a solution:
AudioCodes - SIP Server - Bria - GAD

By some reason when a customer calls and call is received by an agent A and he tries to transfer to agent B by 2 step transfer, Agent A and B can't talk, Agent B does hear agent A but Agent A doesn't hear Agent B. If I complete the transfer then the call with caller does work fine and Agent B can hear the caller.


I already configured everything as G711 (PCMU) so I avoid any codec issues. Sniffer show this as ok.

If during the consult call agent B puts on hold to agent A and then recovers then Audio is OK! I can complete transfer then but caller call is released after a few seconds ()

On wireshark traces what I see "Destination Unreachable (Port unreachable)" so it seems that the RTP port can't be used...

On BRIA I already configured a port limit and no problem with that. If I do the tests with Bria (transfer) then I can hear the other agent (A and B) but after I transfer the call the caller drops after a few seconds and no audio between agent B and the caller...

My DN annex tab is like this:
[quote]

[TServer]
contact=sip:3042@192.168.100.121:30000;rinstance=6582c9fcaa463c26;transport=udp
dual-dialog-enabled=false
refer-enabled=true
sip-cti-control=talk,hold
transfer-complete-by-refer=true
[/quote]


Will post full wireshark logs and sip server logs

[cavagnaro]

All logs are here:
http://www.sendspace.com/file/o1j41u

I'm completely stucked...please any help would be kindly appreciated

AudioCodes: 192.168.100.168
SIP Server: 192.168.90.101
PC A: 192.168.100.121
PC B: 192.168.100.61

[Adam_W]

I can see from the SIP Server log that one of the INVITES is being rejected as the consultation leg of the call is established:

[code]
17:45:43.988:(1) TRNMNGR: rejecting INVITE with non-empty to-tag - we don't have a matching transaction for it
17:45:43.988:(1) Sending  [476,UDP] 396 bytes to 192.168.100.61:30000 >>>>>
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.100.61:30000;branch=z9hG4bK-d8754z-9d7cf1ece6c7e702-1---d8754z-;rport;received=192.168.100.61
To: <sip:3042@192.168.90.101:5060>;tag=BE670F4B-634D-432F-91F3-D9C425BA4870-13
From: <sip:3041@192.168.90.101>;tag=cc6c2985
Call-ID: BD8C0485-DD7B-4B8B-B3D7-312806B01C5A-8@192.168.90.101
CSeq: 2 INVITE
Content-Length: 0
[/code]

I don't know exactly what's causing that, but I guess it's the reason for the one-way speech.  I had to experiment for a while with different extension options to get Bria/GAD/SIP Server to work together correctly.

These are the options I ended up with in case you want to try them.  I'm using the same version of Bria as you but I'm not using an Audiocodes gateway so obviously bear in mind that this could have an impact on compatibility!  The ones in bold I think would particularly affect transfer scenarios:

[quote]
contact=*
[b]dual-dialog-enabled=true[/b]
make-call-rfc3725-flow=1
[b]refer-enabled=false[/b]
ring-tone-on-make-call=true
sip-cti-control=talk, hold
sip-hold-rfc3264=true
[b]transfer-complete-by-refer=false[/b]
use-register-for-service-state=true
[/quote]

[cavagnaro]

Thanks a lot for the input. What we did was to go directly to the customer and try directly there and the issue didn't reproduce there...weird...
anyway will still try to make my lab just like customer one to have it working fine.