SIP in higher education

[Stephen Kingham <>]

Hi

Can I presume you have the CCM calling the SER SIP Phone?


Dinesh wrote:

> Hi Stephen,
>
> I tried to configure this.  But I am having some problem to talk to the ser
> (from ccm4.0).  What is the string you used to talk or dial a extension from
> ser to ccm 4.0? I assumed you used sip trunk between the ccm4.0 and ser.
>  


The string is the dialed digits. 
CCM ip address 192.168.63.242
Dialed digits 3575

In SER
rewritehost("192.168.63.242");
then route call with
t_relay();

On the call manager make sure the SIP Trunk on the CCM has a Calling 
Search Space (CSS) that is allowed to make calls to extensions, ie the 
CCS has the necessary Partitions added.  - Please excuse the Cisco Call 
Manager speak ;-).

You might need to look at an ngrep trace (try ngrep -p -q -W byline port 
5060)


Good luck and let me know how you go as I can probably help a bit more 
that these comments.

I guess we should document this with some screen shots for the Cookbook.

Stephen

> Regards,
>
> Dinesh.
>
>
> -----Original Message-----
> From: Stephen Kingham [mailto:] 
> Sent: Wednesday, June 08, 2005 2:47 AM
> To: 
>
> Subject: Re: [sip.edu] Cisco Call Manager 4.0 + Asterisk/SER ?
>
> Hi Denesh
>
> I have had it working with the CCM 4.0 talking to SER SIP Express Router.
>
> Asterisk is good for services like Voice Mail, IVR, and H.323-SIP 
> Gateway (well sort of) but it is not a SIP Server.
>
> To help with fault find may I suggest you use "ngrep" on the unix server 
> you have your SIP Services running.  You get to see all the SIP 
> messages.  Here is the command line I use:
>
> ngrep -p -q -t -W byline port 5060
>
>
> Stephen
>
> Dinesh wrote:
>
>  
>
> > Hello all,
> >
> >
> >
> > I am trying to make my ccm 4.0 talk to sip.edu with the help of 
> > asterisk.  But I am all lost about how to hook it up.  I have got help 
> > from reading the sip.edu website, that you can do it with the cisco 
> > sip proxy server 2.0.  But getting the software is way to 
> > expensive(50K).  I have made my ccm 4.0 talk to a asterisk box, and 
> > was interested in getting a few pointers of how to interface the two 
> > (sip.edu and asterisk/ser to call manager).  Any help is appreciated.
> >
> >
> >
> > Thanks in advance,
> >
> >
> >
> > Regards,
> >
> >
> >
> > Dinesh Birlasekaran
> > Network Engineer,
> > ComIT, Institute of Molecular and Cell Biology
> > 61 Biopolis Drive, Singapore 138673
> > HP : 92962676 DID : 65869804 Fax : 67791117 Email : 
> >  <mailto:>
> > WWW: www.imcb.a-star.edu.sg <http://www.imcb.a-star.edu.sg>
> >
> >
> >
> >    
>
>  

--
Stephen Kingham, MIT, BSc, E&C Cert
Project Manager and Consulting Engineer 
mailto:
Telephone +61 2 6222 3575 (office)
         +61 419 417 471 (mobile)
sip:

Voice and Video over IP
for The Australian Academic Research Network (AARNet)
http://www.aarnet.edu.au