Category Archives: SIP

Apr 30

Cisco Jabber cannot Call out to PSTN

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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

Nov 20

Sip Calls being Rejected – CUBE

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When calling through to any of the DID number range for a customer, it was found that only a small percentage of calls were successful. The failed calls would either play an ISP announcement or just ring continuously until the timer expired.

This customer had a newly provisioned SIP Trunk to the ITSP and all was working well until this point. No changes had been made by the Internal IT. After tracing successful and failed inbound call attempts it was found the ITSP was sending additional information in the SIP SDP. The information being sent was the QoS SDP Parameters, the local CUBE was not equipped to handle/negotiate these parameters therefore the call negotiation process would fail.

Resolution was to provide the below information to the ITSP to strip the QoS SDP Parameters.

INVITE Captured during trace.

Received:
INVITE sip:2XXXXXXXX@X.X.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 203.52.1.167:5060;branch=z9hG4bKhs0g8n00eogroamlbun0.1
From: <sip:2XXXXXXXX@X.X.X.X;user=phone>;tag=866784654-1439255743666-
To: “Name”<sip:2XXXXXXXX@domain.com.au;user=phone>
Call-ID: BW111543666110815-468528800@10.83.154.184
CSeq: 220849754 INVITE
Contact: <sip:2XXXXXXXX@203.52.1.167:5060;transport=udp>
Supported: 100rel
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Accept: application/media_control+xml,application/sdp,multipart/mixed
Max-Forwards: 29
Content-Type: application/sdp
Content-Length: 641

v=0
o=BroadWorks 3668715 1 IN IP4 X.X.X.X
s=-
c=IN IP4 X.X.X.X
t=0 0
a=sendrecv
a=media-release:hngl5rp9pujvivh05pvu6ibgr83pv898756s7s8auvmg62o6u724a2ur96v3nd0v1350ln1-6
a=media-release-con-addr:d6v1k1hv6hvh1002o9j0
m=audio 18064 RTP/AVP 8 0 18 96 97
c=IN IP4 X.X.X.X
b=RR:3000
b=RS:1000
a=curr:qos local none
a=curr:qos remote none
a=des:qos mandatory local sendrecv
a=des:qos optional remote sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-set=7;max-red=0
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=maxptime:40

Oct 20

Caller-ID SIP – Displaying Names for PSTN Callers

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Calling Party’s Name is not passed through carriers, only the Calling Party’s phone number is passed through. This results in having the caller-id display as a phone number on an Internal Phone. In some cases I’ve had requests to setup a handful of key PSTN/Mobile phone numbers to resolve from phone number to display name. Below is how an example configuration piece on a Cisco IOS Gateway that I configure to resolve phone numbers to display names.


voice class sip-profiles 10
request INVITE sip-header Remote-Party-ID modify “(<sip:00408842426@.*>)(.*)” “\”Ben Morgan\” \1\2″

dial-peer voice 9500 voip
description Mobile-Name-Conversion
destination-pattern ^4…      *** Internal Extension Range
session protocol sipv2
session target ipv4:192.168.0.1    **** CUCM Subscriber Host
voice-class codec 1
voice-class sip profiles 10    **** References above SIP Profile
dtmf-relay rtp-nte
no vad

May 15

SIP Trunk to ITSP – 400 Service Unavailable

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When calling outbound via SIP Trunk to ITSP you receive a 400 Service Unavailable-No Ports Available error after receiving the TRYING packet.

SIP Trace - CUCM

This error may indicate an issue with Early Offer being disabled. Most ITSP’s expect the SDP in the initial SIP INVITE. If the SDP is missing, the ITSP can return the above error.

Simple fix is to enable Early Offer on Cisco CUCM. Below is a quick how to.

  1. Navigate to the SIP Profile configuration page and select “Mandatory (insert mtp if needed) for the Early Offer Support field.
  2. Navigate to the SIP Trunk configuration page and select “MTP Required”.
  3. Reset SIP Trunk.

CUCM Sip Trunk and SIP Profiles