CME 7.1 - Debug outbound calls via SIP

[james-worley]

Hi there,

I am having real trouble setting up outbound SIP calls from my Call Manager Express!

I am using a 2801 running IOS 'c2801-adventerprisek9-mz.124-24.T.bin'. I have around 8 phones running SCCP internally. The Call manager has been working fine for a while now, we recently wanted to make PSTN calls and subscribed to SIP SP for in and outbound calls. After much reading of documentation and forums I have got inbound calls from the SP working. However I can not get outbound calls to work and no amount of playing appears to be working..

My config:

voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol cisco
stun
sip
  bind control source-interface Loopback10
  bind media source-interface Loopback10
  session transport tcp
  registrar server expires max 300 min 60
  transport switch udp tcp
!
!
!
voice class codec 1
codec preference 3 g729r8
codec preference 4 g711alaw
codec preference 5 g711ulaw
!

!
dial-peer voice 2001 voip
description * Outbound calling to Landlines via Gradwell.com *
  destination-pattern 0[1-3].........
voice-class codec 1
  session protocol sipv2
session target dns:sip.trunk.gradwell.com
session transport tcp
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
  no vad
!
dial-peer voice 2011 voip
description * Outbound calling to Mobiles via Gradwell.com *
  destination-pattern 07.........
voice-class codec 1
  session protocol sipv2
session target dns:sip.trunk.gradwell.com
session transport tcp
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
  no vad
!   
!
sip-ua
nat symmetric role passive
nat symmetric check-media-src
no remote-party-id
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
timers connect 100
!
!
telephony-service
sdspfarm transcode sessions 32
conference hardware
fxo hook-flash
max-ephones 24
max-dn 64
ip source-address 192.168.70.1 port 2000
calling-number initiator
no service local-directory
service phone displayOnWhenIncomingCall 1
service phone specialNumbers 999
system message PMD Inc.
cnf-file location flash:
cnf-file perphone
network-locale GB
network-locale 1 GB
network-locale 2 GB
network-locale 3 GB
network-locale 4 GB
load 7960-7940 P00308010200
load 7941 SCCP41.8-5-4S
load 7942 SCCP42.8-5-4S
load 7961 SCCP61.8-5-4S
load 7962 SCCP62.8-5-4S
load 7970 SCCP70.8-5-4S
time-zone 21
time-format 24
date-format dd-mm-yy
keepalive 15
max-conferences 4 gain -6
moh flash:/MOH/music-on-hold.au
multicast moh 239.195.255.255 port 16834 route 172.17.2.2 172.17.2.6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!

!
ephone-dn  59  dual-line
number 0844xxxxxx no-reg both
!

My SP has not provided a username and password of the service and instead chooses to do authentication based on my source IP address and CLID.

[paolo bevilacqua]

Use "debug ccsip message" with "term mon" to see what the problem is.

[james-worley]

thanks for the reply. I have tried 'debug ccsip messages' already and do not get any responces. I was thinking this indicates the problem is before the SIP messages are created????  I have enabled 'debug ccsip all' and get the following:

Jan 29 2010 14:06:29.700 GMT: //48/4B307EB28095/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x69153F8C
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 0844xxxxxxx
Called Number            : 07xxxxxxxxx
Source IP Address (Sig  ): 192.168.70.1
Destn SIP Req Addr:Port  : 212.11.94.100:5060
Destn SIP Resp Addr:Port : 212.11.94.100:5060
Destination Name         : sip.trunk.gradwell.com

Jan 29 2010 14:06:29.704 GMT: //48/4B307EB28095/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.70.1
Source IP Port    (Media): 17490
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0

Jan 29 2010 14:06:29.704 GMT: //48/4B307EB28095/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 38
Disconnect Cause (SIP)   : 503

I understand SIP code 503 is a server unavaliable. I have placed a call with the SP who just tell me that the server is up and its not their problem.

[paolo bevilacqua]

That means, DP is not configuered correctly, and no call is attepted.

[james-worley]

After one of those 'oh you say your using TCP' with the serivce provider the support guy was kind enough to tell me that they only support SIP over UDP. I removed all the TCP related commands and the outbound SIP trunk works.

Many thanks to everyone who posted a reply.

James :-)

[Tommer Catlin · Accepted solution]

Funny or not, it seems to be the biggest issue with the SIP trunk setup these days.  You would think the provider was using TCP and turns out its UDP or vice versa.

Gotta love the forums!