01-25-2017 12:13 AM - edited 03-17-2019 09:18 AM
Hi all,
I have an issue about sip trunk configuration with ITSP.
Here is my Topology
SIP/SCCP phones ----> CUCME/CUE router ----> Firewall -----> EdgeRouter------> Internet----------> ITSP
I have attached CME router configuration file.
I have made outgoing call from my SCCP phone with no success.
Here is debug ccsip messages output for outgoing calls:
SIP Call messages tracing is enabled
un-vr#
Jan 25 07:46:27.522: //30502/34CF873F86F4/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:096134848@81.89.216.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK52C10D5
From: "Ruzanna Zohrabyan" <sip:129@81.89.216.4>;tag=1E41B6C4-1678
To: <sip:096134848@81.89.216.4>
Date: Wed, 25 Jan 2017 07:46:27 GMT
Call-ID: 36501804-E20911E6-86F98E92-7CF6C55E@192.168.10.2
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0886015807-3792245222-2264174226-2096547166
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1485330387
Contact: <sip:129@192.168.10.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 718 7174 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 16822 RTP/AVP 0 101
c=IN IP4 192.168.10.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Jan 25 07:46:28.022: //30502/34CF873F86F4/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:096134848@81.89.216.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK52C10D5
From: "Ruzanna Zohrabyan" <sip:129@81.89.216.4>;tag=1E41B6C4-1678
To: <sip:096134848@81.89.216.4>
Date: Wed, 25 Jan 2017 07:46:28 GMT
Call-ID: 36501804-E20911E6-86F98E92-7CF6C55E@192.168.10.2
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0886015807-3792245222-2264174226-2096547166
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1485330388
Contact: <sip:129@192.168.10.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 718 7174 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 16822 RTP/AVP 0 101
c=IN IP4 192.168.10.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Jan 25 07:46:29.022: //30502/34CF873F86F4/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:096134848@81.89.216.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK52C10D5
From: "Ruzanna Zohrabyan" <sip:129@81.89.216.4>;tag=1E41B6C4-1678
To: <sip:096134848@81.89.216.4>
Date: Wed, 25 Jan 2017 07:46:29 GMT
Call-ID: 36501804-E20911E6-86F98E92-7CF6C55E@192.168.10.2
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0886015807-3792245222-2264174226-2096547166
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1485330389
Contact: <sip:129@192.168.10.2:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 246
v=0
o=CiscoSystemsSIP-GW-UserAgent 718 7174 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 16822 RTP/AVP 0 101
c=IN IP4 192.168.10.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
When I am calling to my given number from ITSP, I hear message from provider PBX ("The number is unavailable")
but I am registered with SP:
un-vr#show sip-ua register status
Line peer expires(sec) reg survival P-Associ-URI
================================ ========== ============ === ======== ============
60488111 -1 1934 yes normal
Solved! Go to Solution.
01-25-2017 02:23 AM
Hi Rozanna,
Can you confirm with the provider that the number you are sending in the Invite is correct ?
From: "Ruzanna Zohrabyan" <sip:129@81.89.216.4>;tag=1E41B6C4-1678
To: <sip:096134848@81.89.216.4>
What is the reply you are getting from the provider for the invite sent. Can you provide the complete ccsip debugs to check that.From the info you provided, it seems like the provider is not accepting the number you sent in the invite and they dont know to route the call for the number sent.
HTH
Rajan
Pls rate all useful posts
01-25-2017 02:23 AM
Hi Rozanna,
Can you confirm with the provider that the number you are sending in the Invite is correct ?
From: "Ruzanna Zohrabyan" <sip:129@81.89.216.4>;tag=1E41B6C4-1678
To: <sip:096134848@81.89.216.4>
What is the reply you are getting from the provider for the invite sent. Can you provide the complete ccsip debugs to check that.From the info you provided, it seems like the provider is not accepting the number you sent in the invite and they dont know to route the call for the number sent.
HTH
Rajan
Pls rate all useful posts
01-27-2017 02:48 AM
Actually 129 is our local extension, it must be from 129@192.168.100.1
I don't know why it has been changed?
01-27-2017 02:52 AM
There is no other messages, I have pasted all output from the debug ccsip messages command.
01-27-2017 07:30 PM
Hi Ruzanna,
As there is no reply back to the INVITE from ITSP (TRYING for example), this means there is no SIP messages forwarded to CME. Check firewall configurations if is forwarding incoming SIP traffic to CME either by one-to-one NATing (if you have dedicated public IP address for CME) or port forwarding (forward 5060 and UDP media).
Thanks,
BH
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