SIP Trunk between CME and CUCM

[fuhrersk8]

                   Hello guys!

I am trying to establish a SIP trunk between a CME 7 and a CUCM 8.0.3 but i am getting fast busy when calling from CME to CUCM. Test directory number is 6367. On the CME i get the following:

3A5D : 379082 -687102836ms.16874 +-1 +30 pid:30 Originate 6367

dur 00:00:00 tx:0/0 rx:0/0 39  (bearer capability not authorized (57))

IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off

  media inactive detected:n media contrl rcvd:n/a timestamp:n/a

  long duration call detected:n long dur callduration :n/a timestamp:n/a

The dial-peer on CME is:

dial-peer voice 30 voip

description SIP Trunk to 6XXX

translation-profile outgoing Spain

preference 1

destination-pattern 6...

session protocol sipv2

session target ipv4:10.34.6.12:5060

dtmf-relay sip-notify rtp-nte

no vad

I see the bearer capability not authorized error on the call history brief. what should I be looking forward on this?

Thanks in advanced!

[Chris Deren · Accepted solution]

Please provide full CME config, IOS version and "debug ccsip messages".

On CUCM side ensure the CSS on the SIP trunk is configured with access to partition of the DN you are calling.

Is the SIP trunk assigned to DP that uses CUCM Group where 10.34.6.12 CUCM server is listed?

If IOS is 15.1.2T or newer ensure toll fraud prevention is not blocking you, "debug voice ccapi inout" would show that.

Chris

[fuhrersk8]

Hello Chris and thanks for reply. IOS version is 12.4. Since I only control the CME, I cannot debug on CUCM side. I am trying to make sure the CME side is cleared from any misconfiguration. Below is the relevant CME config:

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
sip
   outbound-proxy dns:sbcent.centixvoip.net
!
dial-peer voice 30 voip
description SIP Trunk to Spain 6XXX
translation-profile outgoing Spain
preference 1
destination-pattern 6...
session protocol sipv2
session target ipv4:10.34.6.12:5060
dtmf-relay sip-notify rtp-nte
codec g711ulaw
no vad
!
telephony-service
max-ephones 40
max-dn 114
ip source-address 10.110.250.1 port 2000
system message EDICIONES SM P.R.
url services http://10.110.250.2/voiceview/common/login.do
url authentication http://10.110.250.2/voiceview/authentication/authenticate.do
cnf-file location flash:
load 7915-12 B015-1-0-3.SBN
load 7942 SCCP42.8-4-2S.loads
load 7962 SCCP42.8-4-2S.loads
time-zone 15
date-format dd-mm-yy
voicemail 999
max-conferences 8 gain -6
moh music-on-hold.au
multicast moh 239.10.16.4 port 2000 route 10.110.250.1 10.32.9.178
web admin system name admin password c1sc0123
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 9T
secondary-dialtone 9
directory last-name-first
directory entry 1 100 name Sheila Verdiales
create cnf-files version-stamp 7960 Feb 27 2013 18:03:14
!

[Chris Deren]

Can you provide the requested debug from CME?

Can you answer the questions?

Chris

[fuhrersk8]

Hi Chris. Have not performed any debug yet since this is a production voice gateway and i am connected remotely to it. I was provided with screen shots of CUCM configuration and I see that MTP is set as required. Will this be an issue?

Also, under SIP Information, the SUBSCRIBE Calling Search Space was not set. Is this the CSS that need to be configured as you mentioned?

Thanks again Chris. Below screen shots I was provided with:

[fuhrersk8]

Hello guys. Problem solved by configuring the outbound proxy on the specific dial peer to voerride the global one. It was a mismatch. Thanks all for your help and support.  

[islam.kamal]

hello

first check the below link

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a00808b6ca6.shtml

voice service voip

sip

bind all source-interface loopback 0

dial-peer voice 3120 voip

destination-pattern 312301....$

session target ipv4:x.x.x.x (pub)

session protocol sipv2

dtmf-relay sip-notify rtp-nte

codec g711ulaw

!

dial-peer voice 3121 voip

destination-pattern 312301....$

session target ipv4:x.x.x.x (sub)

session protocol sipv2

dtmf-relay sip-notify rtp-nte

codec g711ulaw

thank you

please rate if this will help

[fuhrersk8]

Hello Kamal and thanks for the reply;

     In my config I have not binded the sip service to any interface. In my case, the voice gateway is connected to a SIP server to the PSTN and the SIP dial-peer that points to the CUCM is routed through another interface. Does binding the SIP service to an interface will affect my connection to the PSTN SIP server?

Thnaks again for reply.

[Ayodeji Okanlawon]

I suggest you provide the debugs that Chris requested..Unless you understand how sip bind works dont apply any bind command yet. As chris mentioned you need to provide your full show run and

debug ccsip messages

debug voip ccapi inout

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"

[fuhrersk8]

I understand. As soon as I am able to collect traces will post them. Thanks again guys!