06-27-2013 09:30 PM - edited 03-17-2019 03:21 PM
I've got Webex on site meeting server V1.0 deployed with IRP. The issue that I experience is that I can dial into the audio conference, but as soon as I enter the conference number followed by the # key, it drops the call. This happens when I dial internally as well as through the PSTN. Any ideas out there that what can be wrong?
Thanks
Henry
Solved! Go to Solution.
10-06-2014 01:44 PM
Hi There
I understand this is an old post but we got the same problem here. We could solve the issue enabling the re-routing field in the Load Balance CUCM Sip Trunk. Just make sure the re-routing CSS has the Partitions you need to reach the destination.
Another trick for us was that in the SIP Pattern we were using the FQDN instead of the IP address.
Now its working fine.
Mártin
06-27-2013 09:32 PM
Hi Henry,
Seems your CUCM Configuration is missing specailly around SIP trunk/s. Please check below link to confirm the same:
http://www.cisco.com/en/US/docs/collaboration/CWMS/b_planningGuide_chapter_0101.html
Thanks, Arun
06-27-2013 10:23 PM
Thanks Arun
The trunks seems to be working fine. Webex can dial me back to my internal extension. When I dial into Webex, the system answers. When I type the wrong meeting number, it tells me that the meeting does not exist or did not start yet. When I type the correct meeting number, the audio call terminate immediatly.
06-29-2013 03:13 PM
Hi Henry,
This is an issue with trunks only. Let me know what is the system size, are you using TLS or non-TLS and I'll let you know how to configure them correctly. Here issue seems with your REFER trunk (which plays role when doing dial-in, not dial-out).
Thanks, Arun
06-30-2013 01:39 AM
Arun
I've got a 50 user system with IRP. I use non-TLS.
Regards
Henry
06-30-2013 11:14 AM
07-01-2013 10:15 AM
Arun
Thank you. I will check the configurations and let you know.
Regards
Henry
Arun
I've tried your configuration still the same issue. See attach my configuration.
Regards
Henry
09-18-2013 07:23 AM
Thanks for the document!
Had some issues with dialing out, this document solved it.
Jan
09-18-2013 08:01 AM
Pleasure
11-14-2013 05:15 PM
Hello,
I am having the exact same problem but cannot find anything wrong with my sip trunks. Henry, Jan what exactly was wrong with yours?
Thanks a lot!
Alex.
11-15-2013 10:20 AM
Never mind guys I figure it out. It was my SIP routing pattern partition. After the SIP refer from CWMS the CUCM didn’t found the new sip address because this pattern was in a wrong partition and it was replying with a Notify SIP/2.0 404 Not Found. The SDI logs pointed this out. Simply putting the pattern in the none partition fixed it.
Thanks a lot!
12-17-2013 03:59 PM
I also setup lab without IRP, no partition, same issue like this. still not resolve.
any other possible reason?
12-18-2013 04:57 AM
Hi,
Please, review this guide and ensure everything is configured as per documentation:
http://www.cisco.com/en/US/docs/collaboration/CWMS/1_5/Planning_Guide_chapter_0110.html
If everything is setup as documented, and the appropriate Partitions and CSS are used, this should work without a problem.
-Dejan
10-06-2014 01:44 PM
Hi There
I understand this is an old post but we got the same problem here. We could solve the issue enabling the re-routing field in the Load Balance CUCM Sip Trunk. Just make sure the re-routing CSS has the Partitions you need to reach the destination.
Another trick for us was that in the SIP Pattern we were using the FQDN instead of the IP address.
Now its working fine.
Mártin
10-07-2014 05:58 AM
I have the same problem... last week was working fine.
I have the pattern in a none partition, so dont have re-routing css. i tried specifiying a partition and re-routings css and still the same.
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