[asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call

Thu Apr 16 06:22:39 CDT 2009

Sounds like the real question is: can Asterisk originate and receive  
SIP calls?

The answer is yes. :-)

--
Sent from mobile device

On Apr 16, 2009, at 7:17 AM, Vidura Senadeera <vidurased at gmail.com>  
wrote:

> Hi,
>
>  You can achieve this by integrate CCM and asterisk using SIP trunk.
>
> In CCM you can create SIP trunk, After creating SIP trunk in between  
> CCM and asterisk, you have to configure dialplan on CCM to pass the  
> calls to asterisk.
>
> One the caller id comes to Asterisk you have to use extension.conf  
> to route the calls.
> You can also try with freepbx GUI to configure inbound route, it  
> makes your life easy.
>
>
> -- 
> Thanks & Regards,
> Vidura Senadeera,
> Sri Lanka.
> msn/yahoo/skype Ids - vidurased
>
> ======================================
> Message: 16
> Date: Fri, 10 Apr 2009 00:06:50 -0600
> From: Shocky <shocky1 at users.sourceforge.net>
> Subject: [asterisk-users] Can Asterisk bridge between a SIP client and
>        a       Cisco Call Manager server?
> To: asterisk-users at lists.digium.com
> Message-ID: <200904100006.51201.shocky1 at users.sourceforge.net>
> Content-Type: text/plain;  charset="us-ascii"
>
> Hi,
>
> This is probably outside what Asterisk is intended for, but I'm  
> hoping it can
> help.
>
> I need to make and receive calls through a Cisco Call Manager server  
> that I
> have no control over. I have to use a Cisco soft phone (Cisco IP
> Communicator), which only runs on Windows. But I'm on Linux. CCM is
> apparently capable of supporting SIP and H.323 interfaces, but they  
> won't
> provide this option for me. Right now I'm using a VMWare XP guest to  
> run the
> soft phone, but this is painful (especially with some VPN  
> complications
> thrown in).
>
> I've read that Asterisk supports SCCP, at least somewhat. I'm  
> wondering if I
> could set up Asterisk on my desktop machine to route calls between a  
> SIP
> client such as Kphone or Ekiga and the CCM server. Would this be  
> possible?
>
> I heard that one of the problems in interfacing with CCM over SCCP  
> is the use
> of proprietary codecs. Would this be a problem in my case?
>
> If there's a chance it can be made to work, I'll give it a try. If  
> I'd be
> wasting my time, please let me know.
>
> Thanks,
>
> Shocky
> --
> These are my opinions. Get your own.
>
>
>
> ------------------------------
>
> Message: 17
> Date: Fri, 10 Apr 2009 10:07:38 +0300
> From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
> Subject: Re: [asterisk-users] MeetMe not working - was before
> To: asterisk-users at lists.digium.com
> Message-ID: <20090410070738.GS3227 at xorcom.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote:
> > When I dial the extension of a meetme conference room, I get a  
> message that
> > states "is not a valid conference".  The meetme app was working  
> before.
> >
> > I am getting this error on the CLI:
> > app_meetme.c:800 build_conf: Unable to open pseudo device
> >
> > I have Asterisk  1.4.23.1 and zaptel-1.4.11
>
> Elsewhere you mentioned you also have dahdi installed. What is the
> output of:
>
>  ls /usr/include/dahdi
>
> I suspect Asterisk was built vs. dahdi whereas Zaptel was actually
> running.
>
> Actual tests:
>
>  dahdi_test
>
> vs.
>
>  zttest
>
> --
>               Tzafrir Cohen
> icq#16849755              jabber:tzafrir.cohen at xorcom.com
> +972-50-7952406           mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com  iax:guest at local.xorcom.com/tzafrir
>
>
>
> ------------------------------
>
> Message: 18
> Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST)
> From: Gordon Henderson <gordon+asterisk at drogon.net>
> Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client
>        and a Cisco Call Manager server?
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.64.0904101032040.23406 at unicorn.drogon.net>
> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
>
> On Fri, 10 Apr 2009, Shocky wrote:
>
> > Hi,
> >
> > This is probably outside what Asterisk is intended for, but I'm  
> hoping it can
> > help.
> >
> > I need to make and receive calls through a Cisco Call Manager  
> server that I
> > have no control over. I have to use a Cisco soft phone (Cisco IP
> > Communicator), which only runs on Windows. But I'm on Linux. CCM is
> > apparently capable of supporting SIP and H.323 interfaces, but  
> they won't
> > provide this option for me. Right now I'm using a VMWare XP guest  
> to run the
> > soft phone, but this is painful (especially with some VPN  
> complications
> > thrown in).
> >
> > I've read that Asterisk supports SCCP, at least somewhat. I'm  
> wondering if I
> > could set up Asterisk on my desktop machine to route calls between  
> a SIP
> > client such as Kphone or Ekiga and the CCM server. Would this be  
> possible?
> >
> > I heard that one of the problems in interfacing with CCM over SCCP  
> is the use
> > of proprietary codecs. Would this be a problem in my case?
> >
> > If there's a chance it can be made to work, I'll give it a try. If  
> I'd be
> > wasting my time, please let me know.
>
> I've never looked at SCCP, but if it does work then you could use the
> console phone built into asterisk rather than IP plumb it into a
> soft-phone... So asterisk is essentially acting as an SCCP soft-phone
> itself. No GUI though, but if you're happy typing commands... :)
>
> Gordon
>
>
>
>
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