Call from IP phone trought SIP trunk uses 2 VCM chanels 1

[Orintas]

Problem with SIP. With SIP NR. 1, a call from IP phone (avaya 5610 SW), trought SIP trunk, uses 2 VCM chanels, for all call. SIP trafic (debuginfo) trace log:

http://www.savefile.com/files/1923714

With anoter SIP provider, all is OK. A call from IP phone trought SIP uses 1 VCM chanel only for call initiation. SIP trafic trace:

http://www.savefile.com/files/1923723

IP Office 500 firmware - 4.2(11).
In both cases used the same system, only diferent SIP providers.

 

[IPOfficeIreland]

the 1st provider is using a Cirpack switch v4.41f.

This does not support RFC2833 for Inband DTMF, look at your sip message

m=audio 31176 RTP/AVP 8 0 18
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
You will notice the 101 missing from the RX Message even though AVaya sent it with the TX.

Unless your provider will upgrade Cirpack (which is a Class 5 switch and will have many reprocussions) then you are banging your head against a wall here - it will never work. Put an asterisk server inbetween for 0 VCMs both ways or a Session Border Controller at least so that you can use 2 VCMs for outgoing and 0 for incoming.

 

[Orintas]

Thank You