SIP Server call flow

[cavagnaro]

Hi guys,
I'm facing an issue here and would like to know if it is normal behavior or can be modified...

Scenario:

Have an AudioCodes
AudioCodes send call to SIP Server
SIP Server to a Routing Point
Routing point to an agent

So at the end the call on SIP messaging exchange ends up like being between the Routing Point and the AudioCodes, but I'd like it to be between the agent extension and the AudioCodes...

Have you faced a similar issue?
I'm using version G8.0 of SIP Server

AudioCodes Trunk is like:

[Tserver]
contact = sip:192.168.10.168:5060
prefix = 9

Thanks

[borkokrz]

Can you post some logs ? I have AudioCodes with 7.5 and just before EventEstablished on place i see:

sipcs: 10:25:57.387 Received [428,UDP] 940 bytes from 126.61.1.22:5060 <<<<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 126.61.253.41:5060;branch=z9hG4bK36B24A9D-688C-4D7A-ACE5-AFCEC5695B1C-4689447
From: <sip:519124462@126.61.253.45>;tag=1c1752452173
To: <sip:323236000@126.61.253.41>;user=phone;tag=FCEFFFA7-E68BBB1C
CSeq: 1 INVITE
Call-ID: B048C95F-6DD0-4423-98DD-98E981364F69-1049437@126.61.253.41
Contact: <sip:7210014@126.61.1.22:5060>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_550-UA/2.1.0.2708
Call-Info: <http://genesyslab.com>; 175245131323122011102530%40126.61.253.45;gen-rt=1c1752452173;gen-lt=9DF8AC05-3F24-4739-9AD4-2FEAFEC83BF1-208095
Content-Type: application/sdp
Content-Length: 223


v=0
o=- 1324632312 1324632312 IN IP4 126.61.1.22
s=Polycom IP Phone
c=IN IP4 126.61.1.22
t=0 0
m=audio 2248 RTP/AVP 18 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

I've refer-enabled=true on AudioCodes trunk.

[kolonil]

Hi,

I may be wrong, but i think it is normal behavior in case when agent extensions registered directly on SIP server, that is: both RP and extensions are in DNs folder of SIP switch and agents managed by SIP Server ("peered deployment" in documentation terms). In such case SIP server acts as some "bridge" in SIP messaging between Audiocodes gateway and agents endpoint. This applies only to messaging, real voice goes directly from gateway to endpoint (if no recording by streammanager implemented). This can easily be checked by wireshark capturing traffic on sip server host.

Another question - what is the goal of exchanging SIP messaging directly between gateway and extension in the end of call routing...
I think there may be many solutions depending on the problem

Regards,
Mike

[Fra]

Cav,

SIP Server is a B2BUA, you can't take it off the signalling path with SIP DNs (what for, btw?  ).

Fra