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New Member

Jabber cannot do voice calls to PSTN or to other jabber user


I'm trying to troubleshoot why jabber for windows 9.7.2 is not able to establish calls to PSTN dialed by DN and to other jabber user called by URI, not DN.


We have CUCM 10 with IM and Presence 10. IM and Presence is newly installed. I have created 2 x Cisco Unified Client Services Framework devices for 2 users. Users are able to connect through jabber for windows, IM, do direct DN-to-DN calls (including video), but not able to call outside to PSTN and by URI to other users.


- jabber for windows - call DN to DN between internal users - working with video

- jabber for windows - call DN to PSTN through SIP-to-SIP connection (SIP trunk to GW, SIP trunk from GW to ISP) - not working CUCM returning me Q.850 cause 65 to jabber immediately when other side pick up phone, even jabber hear ringing with music from ISP side. - Cause: 65(0x41)[Bearer capability not implemented]

- jabber for windows - in search field I search for second user, I can see presence, when I click call button, call is immediately dropped - don't know reason still not troubleshooted.


communication between GW 2801 and ISP is OK, they are acknowledge codec and everything is fine.


EDIT: I tried to connect jabber for windows directly to CUCM TFTP servers without IM and Presense server. I have same behavior, when outgoing call is picked up it is immediately dropped by CUCM and send Q.850 cause 65 to jabber.


EDIT2: any other SIP device configured same way as jabber CSF is working. just jabber not.


EDIT3: when I selected "Media Termination Point required" on jabber device in CUCM, then voice is working normally. But if possible I do not want this and make voice without MTP. How?



Everyone's tags (1)
Cisco Employee

Try to disable video and then

Try to disable video and then test another time to PSTN

Have you enabled URI dialing??



if this helps, please rate
New Member

ok I have disabled video in

ok I have disabled video in jabber client. same, not working. call is immediately after pickup on other side dropped.

New Member

I found it, i have enabled

I found it, i have enabled "Allow Presentation Sharing using BFCP" on SIP profile assigned to trunk pointing to voice gateway. This causes that SDP from gateway didn't come.

New Member

Hi Tibor, 

Hi Tibor, 

did this fixed your issue? we have where the call cut off after 16 sec ? , was reading about MTU size blocked by ASA ? not sure from here 

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