SIP Calls on CUCM failing

[Valentine Sondoyi]

My setup is as below

CUCM------------------SIP/H323-----------Voice Gateway(2951)---------------------------E1/ITSP

The E1 operation is whats in use. We now adding a SIP trunk from Telco.

Have configured VG as h323 gateway which is working well with the E1 setup. Also configured a SIP trunk from CUCM to VG. When using H323, can make calls on ITSP sip trunk but only getting one way audio, Can hear called party but they cant hear us. When routing the call via the SIP trunk, am getting "your call cannot be completed as dialed..etc" from that lovely lady on CUCM.

Below is my SIP codec and dial-peer information.

Interface gig0/1 is terminating ITSP sip link, gig0/0 is for CUCM connection.

======================

voice service voip
 no ip address trusted authenticate
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 supplementary-service h450.12
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet0/1
  bind media source-interface GigabitEthernet0/1
  min-se 2000 session-expires 2000
  header-passing
  registrar server expires max 600 min 60
  asserted-id pai
  no silent-discard untrusted

=================================

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729br8
!

===============================

sip-ua
 retry invite 2
 retry response 2
 retry bye 2
 retry cancel 2
 registrar 1 ipv4:10.37.37.254:5060 expires 600

==========================================

dial-peer voice 102 pots

translation-profile outgoing RTEL-OUT

destination-pattern 7T

port 0/0/0:15

!

dial-peer voice 199 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 100 voip

destination-pattern 8...

session target ipv4:192.168.51.14

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 101 voip

destination-pattern 3...

session target ipv4:192.168.51.14

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 103 pots

translation-profile outgoing RTEL-OUT

destination-pattern 9T

port 0/0/0:15

!

dial-peer voice 104 voip

description RTEL SIP CALLS

translation-profile outgoing RTEL-SIP-OUT

destination-pattern .T

session protocol sipv2

session target ipv4:10.37.37.254

session transport udp

voice-class codec 1

voice-class sip bind control source-interface GigabitEthernet0/1

voice-class sip bind media source-interface GigabitEthernet0/1

dtmf-relay rtp-nte

no vad

!

dial-peer voice 198 voip

translation-profile incoming RTEL-SIP-IN

destination-pattern [3,8]...

session target ipv4:192.168.51.14

incoming called-number .T

voice-class codec 1

voice-class sip bind control source-interface GigabitEthernet0/1

voice-class sip bind media source-interface GigabitEthernet0/1

dtmf-relay rtp-nte

!

dial-peer voice 105 voip

session protocol sipv2

session transport udp

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

===========================================

Have attached some debugs

[Vivek Batra]

Hi,

Since you are using SIP trunk on Telco side, I suggest you to use SIP in between CUCM and gateway for better inter-networking.

For failed call when you use SIP between CUCM and gateway, please share the output of debug voice ccapi inout and debug ccsip messages.

- Vivek

[Valentine Sondoyi]

have run debug but nothing showing on router. Not even able to show any output apart from the usual setup messages with telco side.

[Vivek Batra]

That probably means you have something wrong in CUCM configuration.

Can you please cross verify related configuration in CUCM viz partition/CSS, route group/list and SIP trunk configuration etc....

- Vivek

[Valentine Sondoyi]

Thanks Vivek. That part is sorted now. Can see the calls hitting the VG but i think there is something wrong with my dial-peers or interface binding for SIP.

Since interfaces gi0/0 and gi0/1 are both pointting to SIP trunks (ITSP and CUCM) I am not sure which one I should bind. The debug message shows below

Mar  4 13:16:07.100: //-1/5D22D8000000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x15DEEA78
State of The Call        : STATE_DEAD
TCP Sockets Used         : YES
Calling Number           : 3299
Called Number            : 0787606460
Source IP Address (Sig  ): 10.37.37.15
Destn SIP Req Addr:Port  : 192.168.51.14:0
Destn SIP Resp Addr:Port : 192.168.51.14:52430
Destination Name         : 192.168.51.14

*Mar  4 13:16:07.100: //-1/5D22D8000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 38
Disconnect Cause (SIP)   : 422

The source should be the CUCM - 192.168.51.14, and destination should be the ITSP - 10.37.37.15

How can I change this with the dial-peers?

[Vivek Batra]

That will be better if you could have shared complete SIP logs, but anyway you had mentioned before that gig 0/0 is the interface towards CUCM and gig 0/1 towards service provider. In that case, you should bind inbound dial peer with gig 0/0 and outbound dial peer with gig 0/1.

- Vivek

[Valentine Sondoyi]

Full log attached.

[Valentine Sondoyi]

Getting one way audio after binding interfaces in the dial-peer as above. Should interfaces be bound under voip service config?

[Vivek Batra]

From SIP traces, I can say that dial peer towards CUCM is bind with 192.168.50.5 and towards provider is bind with 10.37.37.15. If this is as per your requirement, you should be good to go.

1. CUCM is using TCP on call leg to router whereas on provider leg, router is using UDP. I suggest you to use UDP on both call legs. In CUCM, go to SIP Trunk Security Profile and change the outgoing transport type to UDP.

2. As per SIP traces, RTP packets from phone to provider will flow from 192.168.51.14 to 192.168.50.5 to 10.20.34.4.

RTP packets from provider to phone will flow from 10.20.34.4 to 10.37.37.15 to 192.168.1.242.

You should check for proper routing between these networks.

- Vivek