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:firstname.lastname@example.org>;tag=1c1543675873 --- 184.108.40.206 is transcoder ip
To: <sip:email@example.com> --- 220.127.116.11 is CUBE Ip
CSeq: 1 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
User-Agent: Mediant 800B/v.6.80A.285
P-Asserted-Identity: "+27827016165" <sip:firstname.lastname@example.org>
Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
o=FreeSWITCH 1542808898 1542808807 IN IP4 18.104.22.168
c=IN IP4 22.214.171.124
m=audio 6070 RTP/AVP 18 4 8 118 100
Can anyone shed some light on this ?
Attached is the debug from CUBE
Solved! Go to Solution.
The ReInvite is sent to "email@example.com" by CUBE because it was received in Contact header of the first Invite.
Do you have "midcall-signaling passthru media-change" configured under "voice service voip" in CUBE? If not can you configure it and test.
Just FYI.. there is no DTMF configuration under dial peer to CUCM. The DTMF payload used by Mediant is 100. Cisco uses by default 101 for rtp-nte.
I have that configured ,please see below :
voice service voip
mode border-element license capacity 20
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
bind control source-interface Loopback0
bind media source-interface Loopback0
This command will consume the ReInvite received from CUCM for transfer so no ReInvite will be sent to Mediant. This stops renogotiating of media again. ReInvite will only be sent to Mediant for a call if it changing from audio to fax/video. Check below document