Webex call me problem

[mohamed sebaey]

hello

I have two issues for CWMS v 2.5:-

A)- When i am going to do join conference after entering meeting access code , unfortunately call ended.

B)- Call me did not work with my SIP trunk. I have SIP trunk between my CUCM and VG , if i am going to do webex call me i can see call hits but no call 

All possible debugging has been turned off
RY-N-VG1#debug  ccsip messages
SIP Call messages tracing is enabled
RY-N-VG1#
Feb 26 14:52:36.327: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.40.90:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKpthoudp2h22adtsukkaot7u7pT19075
Call-ID: isbcuhk7uuuueo4kc4bdkkfdbcud2dekk7hp@SoftX3000
From: <sip:172.29.40.90:5060>;tag=sbc0804stfutfet
To: <sip:172.29.40.90>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

Feb 26 14:52:36.327: //97781/EFC37EF59992/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKpthoudp2h22adtsukkaot7u7pT19075
From: <sip:172.29.40.90:5060>;tag=sbc0804stfutfet
To: <sip:172.29.40.90>;tag=6BEC171C-149A
Date: Thu, 26 Feb 2015 14:52:36 GMT
Call-ID: isbcuhk7uuuueo4kc4bdkkfdbcud2dekk7hp@SoftX3000
Server: Cisco-SIPGateway/IOS-15.2.2.T2
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 166

v=0
o=CiscoSystemsSIP-GW-UserAgent 9233 192 IN IP4 172.29.40.90
s=SIP Call
c=IN IP4 172.29.40.90
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.40.90

Feb 26 14:52:56.363: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.40.90:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKsa7bssuokeufsc2b4h7fffhsuT13497
Call-ID: isbcfbuks7a24a7hpe7sfbeusdhufkhf4obe@SoftX3000
From: <sip:172.29.40.90:5060>;tag=sbc0806sds2ps7a
To: <sip:172.29.40.90>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

Feb 26 14:52:56.367: //97782/FBB55B1D9993/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKsa7bssuokeufsc2b4h7fffhsuT13497
From: <sip:172.29.40.90:5060>;tag=sbc0806sds2ps7a
To: <sip:172.29.40.90>;tag=6BEC6564-12
Date: Thu, 26 Feb 2015 14:52:56 GMT
Call-ID: isbcfbuks7a24a7hpe7sfbeusdhufkhf4obe@SoftX3000
Server: Cisco-SIPGateway/IOS-15.2.2.T2
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 167

v=0
o=CiscoSystemsSIP-GW-UserAgent 4018 4052 IN IP4 172.29.40.90
s=SIP Call
c=IN IP4 172.29.40.90
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.40.90

Feb 26 14:53:16.407: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:172.29.40.90:5060 SIP/2.0
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKpdpstoobfppcpaou44hthh4osT18936
Call-ID: isbc4auh2a7ubhpc2ahca77aa2hdp7a7db4k@SoftX3000
From: <sip:172.29.40.90:5060>;tag=sbc0805odhpc2pk
To: <sip:172.29.40.90>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

Feb 26 14:53:16.407: //97783/07A737DE9994/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.200.7.157:5060;branch=z9hG4bKpdpstoobfppcpaou44hthh4osT18936
From: <sip:172.29.40.90:5060>;tag=sbc0805odhpc2pk
To: <sip:172.29.40.90>;tag=6BECB3AC-180B
Date: Thu, 26 Feb 2015 14:53:16 GMT
Call-ID: isbc4auh2a7ubhpc2ahca77aa2hdp7a7db4k@SoftX3000
Server: Cisco-SIPGateway/IOS-15.2.2.T2
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 167

v=0
o=CiscoSystemsSIP-GW-UserAgent 9949 9129 IN IP4 172.29.40.90
s=SIP Call
c=IN IP4 172.29.40.90
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 172.29.40.90

Note: SIP trunk is configured correctly between Webex and CUCM. SIP trunk work very well for our incoming and outgoing calls.

Thanks

[mohamed sebaey]

Hello 

any help please.

[dpetrovi · Accepted solution]

Hi Mohamed, 

There is most likely something wrong with your SIP configuration on CWMS/CUCM.

What is the size of your CWMS deployment (50 250, 800, or 2000 users)?

Do you have High Availability (HA)?

Based on this information, I may be able to provide you with a summarized information on what you should have configured on CUCM to have this work. 

This is the official documentation for configuring SIP integration between CUCM and CWMS: http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_5/Planning_Guide/Planning_Guide/Planning_Guide_chapter_0111.html

I hope this will help

-Dejan

[ahmed-hassan1110]

Dear Gents,

I have CWMS 2.5 and I'm facing the same issue as the call to the call in number directly disconnects after entering the meeting ID whether the call is from internal IP phone extension or from external source through the PSTN trunks.

moreover the Call me feature fails either to a PSTN number or even when I choose call me on internal extension number and put IP phone extension number, in both case the call fails.

any idea regarding my case ?

[dpetrovi]

Hi Ahmed,

Please, look into this document that explains in details how to configure CUCM integration with CWMS: http://www.cisco.com/c/en/us/td/docs/collaboration/CWMS/2_5/Planning_Guide/Planning_Guide/Planning_Guide_chapter_0111.html

Depending on your system size and if you have HA, there might be more configuration needed on CUCM side.

Make sure your CUCM can handle + sign as all outdials to external numbers from CWMS will have a + sign before country code.

Also, make sure you have a correct CSS configured in Incoming Calls section of SIP trunk, so that the calls can be allowed to the appropriate destinations.

As for indials, most likely you have an issue with SIP route pattern and SIP trunk configurations (make sure to specify Re-routing CSS in SIP trunks that includes the Route Partition defined in SIP route Pattern).

I hope these will be good pointers for you to complete the configuration successfully.

-Dejan

[ahmed-hassan1110]

Hi Dejan,

could you please clarify more how the CUCM can handle the + ? by a real example for a route pattern, also how this can affect even the calls to internal extension numbers ?

[dpetrovi]

HI Ahmed,

You would need to create a translation pattern on CUCM side. I can't provide you with a real example as it is very specific to each environment. I would advise you reach out to CUCM TAC to help you out once everything else is configured.

As for the call back to internal number, + sign is not used. If that is not working, we would need to review your entire configuration. 

What is the size of your system? Do you have HA?

-Dejan

[Terry Cheema]

Hi Ahmed,

CUCM can handle the plus sign. You just need to have a route pattern for plus prefixed with a backslash, for example below Route pattern will handle all calls starting with a plus:

\+!

You can have more specific ones:

\+61!

\+612!

etc.

-Terry

[islam.kamal · Accepted solution]

Hello

Regarding access code problem , this means no rerouting to webex . Please double check your SIP trunk on your CUCM. Error for call me , you have to know the below

1) the route pattern should be match +.

2) CSS should be match the RP.

3) Try to dial a number outside your country and collect logs . To dial any number inside your country , you have to make sure that you discard " + and your country code number ".

thanks

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