<font face="arial" size="2">Hi!<br /><br />
You were right! changing the Calling Search Space for the SIP trunk worked!.<br /><br />
Many thanks for your help Eric. Now I'm fine tuning the application, but the communication with Cisco works pretty fione so far.<br /><br /><br />Alyed </font>
<br />
<br />
<font face="Tahoma, Arial, Sans-Serif" size="2">
<hr align="center" size="2" width="100%" />Return-Path: <end1r@comcast.net> Wed Oct 11 10:57:00 2006<br />Received: from rwcrmhc15.comcast.net [204.127.192.85] by maila11.webcontrolcenter.com with SMTP;<br /> Wed, 11 Oct 2006 10:57:00 -0700<br /></font>
<br />
<br />
<br />I
think SIP is working on your CM becuase it sends a 404 message, I think
your problem is the CM cant find the endpoint in its database.<br /><br />I
am running CM 4.1(3)sr1, and under where you enter the device name
there is a section called, "Call Routing Information". There you should
see a field called "Calling Search Space".<br /><br />You should select the
Calling Search space that the destination endpoints are in. Or if you
want to create a new calling search space, and then give them access to
each other you can do that.<br /><br />In the attachment, I circled the calling search space field I see on my Add NEW SIP TRUNK PAGE.<br /><br />Hope this helps.<br /><br /><br /><br /><br /><br /> -------------- Original message ----------------------<br />From: "Alyed Tzompa" <alyed.tzompa @simitel.com=""><br />> <br />> Many thanks for your interest :)<br />> <br />> In the Cisco the only thing I've donde so far is enable a SIP trunk and pointing <br />> it towards the IP the Asterisk has.<br />> <br />> For this I mean:<br />> <br />> Device --> Trunk --> Add a new Trunk<br />> <br />> Seleccted for Trunk Type "SIP trunk"<br />> <br />> Selected for Device Protocol "SIP"<br />> <br />> Entered for <br />> Device Name "asterisk"<br />> <br />> Entered for <br />> Description "asterisk"<br />> <br />> Selected the appropriate device for Device Pool<br />> <br />> Entered for Destination Address "192.168.1.20" (Asterisk's IP)<br />> <br />> Selected for Outgoing Transport Type "UDP"<br />> <br />> Entered the appropriate routing patterns<br />> <br />> clicked on "Insert"<br />> <br />> in sip.conf I'm using the same config shown in th VoIP-info wiki:<br />> <br />> [cisco]<br />> <br />> type=friend<br />> <br />> context=cisco-test<br />> <br />> host=192.168.1.100<br />> <br />> disallow=all<br />> <br />> allow=ulaw<br />> <br />> allow=alaw<br />> <br />> nat=no<br />> <br />> canreinvite=yes<br />> <br />> qualify=yes <br />> <br />> Regards,<br />> <br />> Alyed <br />> <br />> ----------------------------------------<br />> Return-Path: <end1r @comcast.net=""> Tue Oct 10 16:24:58 2006<br />> Received: from rwcrmhc13.comcast.net [216.148.227.153] by <br />> maila11.webcontrolcenter.com with SMTP;<br />> Tue, 10 Oct 2006 16:24:58 -0700<br />> <br />> I can check for you tomorrow and send you screen shots.<br />> <br />> How do you have the asterisk configured on the CM. Please send me some <br />> configuration information.<br />> <br />> -Eric<br />> -------------- Original message ----------------------<br />> From: "Alyed Tzompa" <br />> > <br />> > Do you know where in the Cisco menu can I<br />> > check out this "calling search spaces" you point out? I don't have<br />> > access to it right now so want to know if this is a configurable menu<br />> > in the web interface or will need to ask for expert (and expensive)<br />> > Cisco assistance.<br />> > <br />> > Alyed <br />> > <br />> > ----------------------------------------<br />> > Return-Path: Tue Oct 10 11:04:09 2006<br />> > Received: from rwcrmhc11.comcast.net [216.148.227.151] by <br />> > maila11.webcontrolcenter.com with SMTP;<br />> > Tue, 10 Oct 2006 11:04:09 -0700<br />> > <br />> > Looks<br />> > like the CallManager is unable to find the endpoint in its database.<br />> > Make sure asterisk trunk on the Call manager is in the same "calling<br />> > Search Space" as the phones are in, or make sure there is access<br />> > between the "calling search spaces"<br />> > <br />> > -Eric<br />> > <br />> > -------------- Original message ----------------------<br />> > From: "Alyed Tzompa" <br />> > > <br />> > > Hi!<br />> > > <br />> > > I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've <br />> followed <br />> > > the info in <br />> > > <br />> > > <br />> > <br />> http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integrat<br />> > > ion <br />> > > <br />> > > but still not able to make Asterisk communicate with Cisco. I keep on <br />> > receiving <br />> > > --- <br />> > > SIP/2.0 400 Bad Request - 'Malformed/Missing URL' <br />> > > --- and --- <br />> > > <br />> > > SIP/2.0 404 Not Found --- <br />> > > messages everytime I send a call. Had play a lot with the way <br />> > > SIP messages are sent to the Cisco, but always been unseccessful. <br />> > > <br />> > > I'm begining to think this is more of a Cisco config problem than<br />> > > Asterisk, has anyone had a similar problem??? I'm not a Cisco expert so<br />> > > dun't know if I need to "enable" SIP messageing/reception in the Cisco.<br />> > > <br />> > > Regards,<br />> > > <br />> > > Alyed <br />> > > <br />> > > <br />> > > <br />> > <br />> > <br />> > <br />> <br />> <br />> <br /><br /><br /><br /><br /><br /></end1r></alyed.tzompa>