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 <[email protected]>
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 <[email protected]>
Subject: [asterisk-users] Can Asterisk bridge between a SIP client and
a Cisco Call Manager server?
To: [email protected]
Message-ID: <[email protected]>
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 <[email protected]>
Subject: Re: [asterisk-users] MeetMe not working - was before
To: [email protected]
Message-ID: <[email protected]>
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:[email protected]
+972-50-7952406 mailto:[email protected]
http://www.xorcom.com iax:[email protected]/tzafrir
------------------------------
Message: 18
Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST)
From: Gordon Henderson <[email protected]>
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
<[email protected]>
Message-ID: <[email protected]>
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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