Hello,
I'm facing an issue here were i can't dial out towards the ITSP from my Cisco UBE
I've been assigned a number range to test, and i have been trying to dial out. The call does not go through, it hits the CUBE from the CUCM but then it seems to die.
CUBE#
*Apr 20 15:10:27: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:+3222299024@10.90.33.249:5060 SIP/2.0
Via: SIP/2.0/TCP 10.3.26.11:5060;branch=z9hG4bK33e9260b6c009
From: "Test" <sip:+46104909610@10.3.26.11>;tag=959365~5eaf8f8c-6191-4e99-9578-b6d8606e956f-23800485
To: <sip:+3222299024@10.90.33.249>
Date: Mon, 20 Apr 2015 13:11:26 GMT
Call-ID: bea79880-5341fafe-b6a0-b1a030a@10.3.26.11
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <sip:10.3.26.11:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Cisco-Guid: 3198654592-0000065536-0000034173-0186254090
Session-Expires: 1800
P-Asserted-Identity: "Test" <sip:+46104909610@10.3.26.11>
Remote-Party-ID: "Test" <sip:+46104909610@10.3.26.11>;party=calling;screen=yes;privacy=off
Contact: <sip:+46104909610@10.3.26.11:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 386
v=0
o=CiscoSystemsCCM-SIP 959365 1 IN IP4 10.3.26.11
s=SIP Call
c=IN IP4 10.110.251.179
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24588 RTP/AVP 0 8 116 18 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:116 iLBC/8000
a=ptime:20
a=maxptime:60
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Apr 20 15:10:27: //52/BEA798800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.3.26.11:5060;branch=z9hG4bK33e9260b6c009
From: "Test" <sip:+46104909610@10.3.26.11>;tag=959365~5eaf8f8c-6191-4e99-9578-b6d8606e956f-23800485
To: <sip:+3222299024@10.90.33.249>
Date: Mon, 20 Apr 2015 13:10:27 GMT
Call-ID: bea79880-5341fafe-b6a0-b1a030a@10.3.26.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.3.M4
Content-Length: 0
*Apr 20 15:10:46: //52/BEA798800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 10.3.26.11:5060;branch=z9hG4bK33e9260b6c009
From: "Test" <sip:+46104909610@10.3.26.11>;tag=959365~5eaf8f8c-6191-4e99-9578-b6d8606e956f-23800485
To: <sip:+3222299024@10.90.33.249>;tag=E759CC84-188D
Date: Mon, 20 Apr 2015 13:10:27 GMT
Call-ID: bea79880-5341fafe-b6a0-b1a030a@10.3.26.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.3.M4
Reason: Q.850;cause=3
Content-Length: 0
*Apr 20 15:10:46: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:+3222299024@10.90.33.249:5060 SIP/2.0
Via: SIP/2.0/TCP 10.3.26.11:5060;branch=z9hG4bK33e9260b6c009
From: "Test" <sip:+46104909610@10.3.26.11>;tag=959365~5eaf8f8c-6191-4e99-9578-b6d8606e956f-23800485
To: <sip:+3222299024@10.90.33.249>;tag=E759CC84-188D
Date: Mon, 20 Apr 2015 13:11:26 GMT
Call-ID: bea79880-5341fafe-b6a0-b1a030a@10.3.26.11
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0
The invite is received from Call Manager (10.3.26.11) and relayed by the CUBE, but it seems that he is relaying it back Call Manager??
The dial peers are setup like this:
CUBE#sh run | s dial-peer
dial-peer voice 1 voip
description Outbound All
translation-profile incoming SIP-IN
preference 1
destination-pattern +.T
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/2
voice-class sip bind media source-interface GigabitEthernet0/2
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description VoIP Dial Peer to CUCM Sub
preference 1
destination-pattern 2.......
session protocol sipv2
session target ipv4:10.3.26.10
incoming called-number +T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1.90
voice-class sip bind media source-interface GigabitEthernet0/1.90
dtmf-relay rtp-nte
no vad
dial-peer voice 101 voip
description VoIP Dial Peer to CUCM Pub
preference 2
destination-pattern 2.......
session protocol sipv2
session target ipv4:10.3.26.11
incoming called-number +T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1.90
voice-class sip bind media source-interface GigabitEthernet0/1.90
dtmf-relay rtp-nte
no vad
GI0/2 is a point to point VLAN with the provider router in it
and Gi0/1.90 is straight over the internet the Call Manager.
I have attached 2 other outputs
- debug voip dialpeer all
- debug voip ccapi inout
It seems that he says incoming dial peer 100? Outgoing call from Callmanager to CUBE. But it says that the outgoing dial-peer = 1, that seems OK?
Could someone assist me in this?
Thanks
