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Highlighted
Beginner

SIP Trunk between CME 9.1 & CUCM 8.6

hello all,

I need to configure sip trunk between CUCM 4.1 at HQ & CME 9.1 at Branch

I've VPN connectivity from HQ to Branch. & i can ping from CME to CUCM ip address & its VG

I have configured SIP trunk into CUCM & i have done following configuration into CME. Incoming call (from CUCM to CME) is working but outgoing calls (from CME to CUCM) is not working.

Please verify & let me know if i need to do additional configuration related to SIP into CME ?

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

h323

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class h323 1

  h225 timeout tcp establish 3

  h225 timeout setup 3

dial-peer voice 9033 voip

description *** Outoing call to CUCM ***

corlist outgoing call-national

translation-profile incoming 2

destination-pattern 9033

session protocol sipv2

session target ipv4:192.168.0.10 <---- CUCM IP Address

session transport udp

voice-class codec 1 

voice-class h323 1

dtmf-relay sip-notify

no vad

dial-peer voice 9034 voip

description *** Incoming call from CUCM ***

incoming called-number .

voice-class codec 1 

voice-class h323 1

dtmf-relay h245-alphanumeric

no vad

sip-ua

sip-server ipv4:192.168.0.10

telephony-service

sdspfarm units 2

sdspfarm transcode sessions 3

sdspfarm tag 1 EXTRA-CONF

sdspfarm tag 2 EXTRA-XCODE

no privacy

conference hardware

no auto-reg-ephone

max-ephones 110

max-dn 400

ip source-address 172.16.246.5 port 2000

timeouts interdigit 5

time-zone 42

voicemail 2000

max-conferences 4 gain -6

call-forward pattern .T

dn-webedit

time-webedit

transfer-system full-consult

transfer-pattern .T

secondary-dialtone 9

create cnf-files version-stamp 7960 Dec 08 2012 07:17:27

!

4 REPLIES 4
Highlighted
Hall of Fame Master

SIP Trunk between CME 9.1 & CUCM 8.6

Why is your incoming dial-peer H323 not SIP?  Is the SIP trunk defined as SIP trunk on CUCM or H323 trunk?

Please post "debug ccsip messages" from CME for the failed call.

Chris

Highlighted
Beginner

SIP Trunk between CME 9.1 & CUCM 8.6

Alright,

I changed my incoming dial-peer to

dial-peer voice 9034 voip

description *** Incoming call from CUCM ***

session protocol sipv2

session target sip-server

incoming called-number .

dtmf-relay sip-notify

It is configured as SIP Trunk into CUCM not H.323 Trunk

Below are the debug ccsip msgs:

Dec 15 06:39:13.597: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:2001@142.100.64.12:5060 SIP/2.0

Via: SIP/2.0/TCP 12.0.0.2:5060;branch=z9hG4bKBCF7

Remote-Party-ID: "SiteC Phone1" <4001>;party=calling;screen=no;priv

acy=off

From: "SiteC Phone1" <4001>;tag=16C51C8-2639

To: <2001>

Date: Sat, 15 Dec 2012 06:39:13 GMT

Call-ID: F9738560-45B811E2-807AA3C1-B35D922A@12.0.0.2

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE:  1800

Cisco-Guid: 4178737364-1169691106-2155127745-3009253930

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF

Y, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1355553553

Contact: <4001>

Call-Info: <12.0.0.2:5060>;method="NOTIFY;Event=telephone-event;Duration=200

0"

Expires: 180

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 239

v=0

o=CiscoSystemsSIP-GW-UserAgent 1516 7769 IN IP4 12.0.0.2

s=SIP Call

c=IN IP4 12.0.0.2

t=0 0

m=audio 17290 RTP/AVP 0 8 18

c=IN IP4 12.0.0.2

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

Dec 15 06:39:13.821: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Date: Sun, 30 Sep 2012 14:36:46 GMT

From: "SiteC Phone1" <4001>;tag=16C51C8-2639

Allow-Events: presence

Content-Length: 0

To: <2001>

Call-ID: F9738560-45B811E2-807AA3C1-B35D922A@12.0.0.2

Via: SIP/2.0/TCP 12.0.0.2:5060;branch=z9hG4bKBCF7

CSeq: 101 INVITE

Dec 15 06:39:13.881: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 503 Service Unavailable

Date: Sun, 30 Sep 2012 14:36:46 GMT

Warning: 399 "Routing failed: ccbid=15 tcpindex=4 socket=12.0.0.2:5060'

From: "SiteC Phone1" <4001>;tag=16C51C8-2639

Allow-Events: presence

Content-Length: 0

To: <2001>;tag=501112352

Call-ID: F9738560-45B811E2-807AA3C1-B35D922A@12.0.0.2

Via: SIP/2.0/TCP 12.0.0.2:5060;branch=z9hG4bKBCF7

CSeq: 101 INVITE

Dec 15 06:39:13.893: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:2001@142.100.64.12:5060 SIP/2.0

Via: SIP/2.0/TCP 12.0.0.2:5060;branch=z9hG4bKBCF7

From: "SiteC Phone1" <4001>;tag=16C51C8-2639

To: <2001>;tag=501112352

Date: Sat, 15 Dec 2012 06:39:13 GMT

Call-ID: F9738560-45B811E2-807AA3C1-B35D922A@12.0.0.2

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

SC#

SC#

SC#

Highlighted
VIP Mentor

Re:SIP Trunk between CME 9.1 & CUCM 8.6

Hi,

Looking at traces your call to extension 2001 is sent to a 142.x.x.x. I also do not see any dial-peer that matched this number. Your cucm is 192.x.x.x IP address. Are you nating this address


Sent from Cisco Technical Support Android App

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Highlighted
Beginner

Re:SIP Trunk between CME 9.1 & CUCM 8.6

I fixed the issue aokanlawon:

i configured bind source control & media pointing to the interface 12.0.0.2

Thanx for your help.

I am having one more issue for the same client & i have posted at following:

https://supportforums.cisco.com/thread/2188217

Let me know if you can help me in this

Regards,

Anis

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