SIP Trunk between CME 9.1 & CUCM 8.6

anis_cisco · ‎12-14-2012

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

!

Chris Deren · ‎12-14-2012

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

anis_cisco · ‎12-14-2012

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#

Ayodeji Okanlawon · ‎12-15-2012

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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anis_cisco · ‎12-15-2012

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