Cisco Jabber cannot Call out to PSTN

Migrated to SIP Carrier and experience an issue where Deskphones could call out to the PSTN, however the Cisco Jabber softphones could not. The annunciator message was played from the carrier network. The remote device (being the device across the PSTN Network) would ring once.

Tracing the SIP messages, I could see the carrier was sending back a SIP reason of “Q.850;cause=41″. Looked up the code which is “Temporary Network Failure – Try again”. So not too much help there..

I analysed the SDP being sent to the carrier from the Cisco Jabber softphone vs the Deskphone and found the video and content sharing attributes were being passed out to the Telco. This Telco connection is just an audio PSTN service, so it would not support video.

I created a new Device Pool for the SIP Trunk to the CUBE, along with a new Region and set the Video to “None”. This effectively disables Video, hence any endpoint including Cisco Jabber establishing call, will not send the Video/Content media attributes. Telco is now happy and calls proceed through the PSTN.

FYI, have pasted below the differences in the SDP.

SDP – Cisco Jabber

v=0
o=CiscoSystemsSIP-GW-UserAgent 9138 8218 IN IP4 192.168.241.20
s=SIP Call
c=IN IP4 192.168.241.20
t=0 0
m=audio 17206 RTP/AVP 8 0 18 101
c=IN IP4 192.168.241.20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 51372 RTP/AVP 31
c=IN IP4 192.168.241.20
m=application 17458 RTP/AVP 125
c=IN IP4 192.168.241.20

SDP – Standard IP Phone (No Camera)

v=0
o=CiscoSystemsSIP-GW-UserAgent 7438 4222 IN IP4 192.168.241.20
s=SIP Call
c=IN IP4 192.168.241.20
t=0 0
m=audio 18034 RTP/AVP 8 0 18 101
c=IN IP4 192.168.241.20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

a=ptime:20

One thought on “Cisco Jabber cannot Call out to PSTN

  1. This tip – to drop the video SDP out from the UCM – works for a lot of SIP carriers who of course never tell you up front about these things.

    If you’re like me and you like putting your CUBE to work with multiple ITSP (not always a good idea) and you don’t want to drop this from the UCM you can also create a SIP profile on the CUBE and apply it to the global SIP configuration or dial peer to remove this, but still allow you to also use the CUBE to say, interconnect with a branch or peer and permit video calling.

    request INVITE sdp-header Video-Attribute remove
    request INVITE sdp-header Video-Media modify “m=video(.*)” “”
    request INVITE sdp-header Video-Bandwidth-Info remove

Leave a Reply

Your email address will not be published. Required fields are marked *


You may use these HTML tags and attributes: <a href="" title=""> <abbr title=""> <acronym title=""> <b> <blockquote cite=""> <cite> <code> <del datetime=""> <em> <i> <q cite=""> <strike> <strong>