12-14-2012 04:03 PM - edited 03-16-2019 02:44 PM
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
!
12-14-2012 06:21 PM
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
12-14-2012 10:36 PM
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;priv4001>
acy=off
From: "SiteC Phone1" <4001>;tag=16C51C8-26394001>
To: <2001>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>4001>
Call-Info: <12.0.0.2:5060>;method="NOTIFY;Event=telephone-event;Duration=20012.0.0.2:5060>
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=16C51C8-26394001>
Allow-Events: presence
Content-Length: 0
To: <2001>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=16C51C8-26394001>
Allow-Events: presence
Content-Length: 0
To: <2001>;tag=5011123522001>
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=16C51C8-26394001>
To: <2001>;tag=5011123522001>
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#
12-15-2012 01:42 AM
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
12-15-2012 03:34 PM
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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