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3CX SIP Trunk Cisco UCM Help

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edujim

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Hello,

I am fairly new to 3CX, and know some stuff in Cisco UCM, I know this topic has been brought up before in past Threads but i dont see any posts/replies to any of them after 2010-2011.

I have successfully created a SIP Trunk between 3cx v11 to Cisco UCM 7.1.

I've got two concerns:

1. when making a call 3CX extension to Cisco extension, It only shows SIP Port # on the display. I dont see the 3CX extension that is calling on cisco phone. (Cisco to 3CX works).
Here is the log:

16-10-2012 12:17:29.366 Leg L:27.2[Line:10000>>1505] is terminated: Cause: BYE from PBX
16-10-2012 12:17:29.366 [CM503008]: Call(C:27): Call is terminated
16-10-2012 12:17:29.365 Leg L:27.1[Extn] is terminated: Cause: BYE from 10.5.80.110:51593
16-10-2012 12:17:25.654 Currently active calls - 1: [27]
16-10-2012 12:17:20.757 [CM503007]: Call(C:27): Line:10000>>1505 has joined, contact <sip:[email protected]:5060>
16-10-2012 12:17:20.755 [CM503007]: Call(C:27): Extn:6504 has joined, contact <sip:[email protected]:51593>
16-10-2012 12:17:20.753 L:27.2[Line:10000>>1505] has joined to L:27.1[Extn]
16-10-2012 12:17:15.869 [CM505003]: Provider:[CCM] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Cisco-CUCM7.1] PBX contact: [sip:[email protected]:5060;transport=TCP]
16-10-2012 12:17:15.809 [CM503025]: Call(C:27): Calling T:Line:10000>>1505@[Dev:sip:[email protected]:5060] for L:27.1[Extn]
16-10-2012 12:17:15.757 [CM503027]: Call(C:27): From: Extn:6504 ("Eddie Jimenez" <sip:[email protected]:5060>) to T:Line:10000>>1505@[Dev:sip:[email protected]:5060]
16-10-2012 12:17:15.757 [CM503004]: Call(C:27): Route 1: from L:27.1[Extn] to T:Line:10000>>1505@[Dev:sip:[email protected]:5060]
16-10-2012 12:17:15.757 Line limit check: Current # of calls for line Lc:10000(@CCM[<sip:[email protected]:5060>]) is 0; limit is 24
16-10-2012 12:17:15.757 Call(C:27): Call from Extn:6504 to 1505 matches outbound rule 'Extension to Cisco'
16-10-2012 12:17:15.755 [CM503001]: Call(C:27): Incoming call from Extn:6504 to <sip:[email protected]:5060>

2. In order to make a successful call from Cisco to 3cx i had to add/change Source ID on 3cx
Request Line URI: User Part Custom Field Custom Value "6504" which is my extension on 3cx
based myself on this article: http://www.3cx.com/blog/docs/source-identification-issues/

Here is Activity Log when it wasnt working

15-10-2012 17:22:39.843 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
Invite-UNK Recv Req INVITE from 10.178.50.3:56350 tid=2d756f8e0c [email protected]:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.178.50.3:5060;branch=z9hG4bK2d756f8e0c
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=tcp>
To: <sip:[email protected]>
From: "Eduardo Jimenez"<sip:[email protected]>;tag=29a56480-65c8-4771-b8ff-c11fbf058889-38300632
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 180
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
Call-Info: <sip:10.178.50.3:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Date: Mon, 15 Oct 2012 22:22:51 GMT
Supported: timer, resource-priority, replaces, Geolocation
User-Agent: Cisco-CUCM7.1
Allow-Events: presence, kpml
P-Asserted-Identity: "Eduardo Jimenez" <sip:[email protected]>
Content-Length: 0
Remote-Party-ID: "Eduardo Jimenez" <sip:[email protected]>;party=calling;screen=yes;privacy=off

Here is my question: what changes or additions do i need to make for the other 3cx extensions to work, when someone from cisco extension dials them.

Any help is greatly appreciated
Eddie
 
Have you create correct trusted sip trunk on Cisco .
Have you create correct Traslation Pattern for incoming call on Cisco and correct Partition ?
 
sip trunk has been created
I didnt see any of the other threads mention that i needed a translation pattern
i created a route pattern that sends all calls with extension 6XXX thru sip trunk
 
Hi,
We had a project where we needed to changeover one phone at a time meaning that the old phone system had to be retained until all were changed over. To make calls to the old Asterisk based system we used a bridge. This might work with one end slave, the other master. Another method, if retaining trunks, is to use a constant trusted parameter such as the IP address of the other PBX.
 
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