Hi All
I have a strange issue that i would like some help with please
Call flow :
Mediant 800 Transcoder -- SIP -- Freeswitch --- SIP --- CUBE --- SIP --- CUCM
Incoming call to cisco phone works fine and i can answer and have 2 way audio ,when i press transfer called party is placed on hold fine ,then phone 2nd cisco phone and when answered there and call is transferred there is only audio in 1 direction ,outgoing from Cisco phone.
Strange issue i am seeing in traces are that initial call leg gets established between Freeswitch and CUBE but when CUBE sends invite from newly established call ,it is sending it directly to Transcoder. In the Orginal INVITE the only reference i can see to the Transcoder IP is in FROM field and
P-Asserted-Identity field as per below :
From: "+27827016165" <sip:27827016165@197.221.111.1>;tag=1c1543675873 --- 197.221.111.1 is transcoder ip
To: <sip:27874401900@197.221.113.17> --- 197.221.113.17 is CUBE Ip
Call-ID: 1543625752235201073934@197.221.113.50
CSeq: 1 INVITE
Contact: <sip:mod_sofia@197.221.113.50:5060>
Supported: timer,replaces,path
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
User-Agent: Mediant 800B/v.6.80A.285
Privacy: none
P-Asserted-Identity: "+27827016165" <sip:27827016165@197.221.111.1>
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Length: 383
X-FS-Support: update_display,send_info
v=0
o=FreeSWITCH 1542808898 1542808807 IN IP4 197.221.113.50
s=FreeSWITCH
c=IN IP4 197.221.113.50
t=0 0
m=audio 6070 RTP/AVP 18 4 8 118 100
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes
a=gpmd:110 vbd=yes
a=pmft: T38
Can anyone shed some light on this ?
Attached is the debug from CUBE
