03-03-2022 10:05 AM - edited 03-03-2022 01:14 PM
hey guys,
I have a problem with calls from a Sip Trunk that go through a Local gateway and then send it to webex calling, when the user puts the call on hold the call ends.
I have attached a log with some debugs
03-03-2022 02:07 PM
Hi there,
try enabling MTP on Trunk and see if that makes a difference
Hope this Helps
Cheers
Rath!
***Please rate helpful posts and if applicable mark "Accept as a Solution"***
03-03-2022 02:49 PM
03-03-2022 02:34 PM
I do not see an attachment. When you add it, please help us interpret it faster by noting what each IP is (eg CUBE, WxC, etc) along with calling/called numbers. Assuming alll-SIP, debug ccsip messages is usually very informative.
03-03-2022 02:48 PM - edited 03-20-2024 04:44 PM
The flow is:
Call from 18097692968 comes from ITSP > Local Gateway 192.168.102.116 (Called number 8096870101) > WebEx (Calling) > Auto Attendant > User presses option 0 > Connected to extension 4450 > 4450 attempts to transfer or put caller on hold > call ends as transfer /hold is attempted
I have attached the log
03-03-2022 02:55 PM - edited 03-20-2024 04:44 PM
The call flow is:
Call from 18097692968 comes from ITSP > Local Gateway 192.168.102.116(Called number 8096870101) > WebEx (Calling) > Auto Attendant > User presses option 0 > Connected to extension 4450 > 4450 attempts to transfer or put caller on hold > call ends as transfer /hold is attempted
I attach the log.
03-03-2022 04:31 PM - edited 06-28-2024 07:37 AM
The call flow is:
Call from 18097692968 comes from ITSP > Local Gateway 192.168.102.116(Called number 8096870101) > WebEx (Calling) > Auto Attendant > User presses option 0 > Connected to extension 4450 > 4450 attempts to transfer or put caller on hold > call ends as transfer /hold is attempted
I have attached the log
03-03-2022 04:59 PM - edited 06-28-2024 07:38 AM
The call flow is:
Call from 18097692968 comes from ITSP > Local Gateway 192.168.102.116 (Called number 8096870101) > WebEx (Calling) > Auto Attendant > User presses option 0 > Connected to extension 4450 > 4450 attempts to transfer or put caller on hold > call ends as transfer /hold is attempted
03-20-2024 02:38 AM
Hi, do you manage find the solution? I'm having the same problem with calls from a Sip Trunk that go through a Local gateway and then send it to Webex calling, when the puts the call on hold the call ends. After run the debug, indicate that below error. I have attached a log with some debugs
*Mar 19 05:33:11.430: //21725/DBFAD8CDB7E0/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.22.213.68:5060;branch=z9hG4bK1FBE1276
Call-ID: asbcgdr6de5drjtetj66fprpis3sfjjredss@ATS.rcatshw01.ims.tm.com.my.155
From: sip:0388939947;phone-context=ims.mnc153.mcc502.3gppnetwork.org@ims.mnc153.mcc502.3gppnetwork.org;user=phone;tag=194FD11E-1C9F
To: sip:Anonymous@ims.tm.com.my;tag=5pj665jg-CC-155
CSeq: 102 INVITE
Warning: 399 198.18.9256.ATS.rcatshw01.ims.tm.com.my.155.451.0.0.0.0.0 "The user doesn't have hold ability"
Content-Length: 0
03-20-2024 10:44 AM
pravin.muthu@vads.com If you're going to resurrect a two year old thread, please at least point at the new post you created so people have a chance of finding the answer. Anyone who comes across this, please see Unable to hold call in Webex Calling.
@stobar1101 Apologies for missing your subsequent replies with an attachment. I presume you found a solution but if not, reply and I'll look at the Zip file.
03-20-2024 04:33 PM
In my case the TSP made a change and I no longer have the problem. They did not tell me what change they made
03-20-2024 11:48 PM
Noted, Im checking with TSP and waiting for the feeback
03-20-2024 11:48 PM
Noted Jonathan, thanks
06-11-2024 02:59 PM
We had the exact same error, but the call was from PSTN to UCM and Unity Connection. The Unity Call Handler was playing the greeting then transfering the call to operator extension. However, the call was not transfering, nor dropping, but hung on dead air. Internal and international calls were transfering ok to the operator. Only local calls were the problem. The provider told us they do not support hold on their SIP trunks. We enabled MTP on the SIP trunk from UCM to the local voice gateway, but that was actually dropping the call, so I took it out.
The problem ended up being a codec problem, where one call leg was using g711ulaw, and the other one was getting g711alaw. Once we forced all call legs with g711alaw the problem was solved. We saw the codec discreppancy on the VGW with the "show call active voice brief" command.
06-11-2024 03:00 PM
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide