<html><body bgcolor="#FFFFFF"><div>Sounds like the real question is: can Asterisk originate and receive SIP calls? </div><div><br></div><div>The answer is yes. :-)<br><br>--<div>Sent from mobile device</div></div><div><br>On Apr 16, 2009, at 7:17 AM, Vidura Senadeera <<a href="mailto:vidurased@gmail.com">vidurased@gmail.com</a>> wrote:<br><br></div><div></div><blockquote type="cite"><div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Hi,</blockquote><div><br></div><div> You can achieve this by integrate CCM and asterisk using SIP trunk.</div>
<div><br></div><div>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.</div><div><br></div><div>One the caller id comes to Asterisk you have to use extension.conf to route the calls.</div>
<div>You can also try with freepbx GUI to configure inbound route, it makes your life easy.</div><div><br></div><div><br>-- <br>Thanks & Regards,<br>Vidura Senadeera,<br>Sri Lanka.<br>msn/yahoo/skype Ids - vidurased<br>
</div><div> </div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">======================================<br>
Message: 16<br>
Date: Fri, 10 Apr 2009 00:06:50 -0600<br>
From: Shocky <<a href="mailto:shocky1@users.sourceforge.net"><a href="mailto:shocky1@users.sourceforge.net">shocky1@users.sourceforge.net</a></a>><br>
Subject: [asterisk-users] Can Asterisk bridge between a SIP client and<br>
a Cisco Call Manager server?<br>
To: <a href="mailto:asterisk-users@lists.digium.com"><a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></a><br>
Message-ID: <<a href="mailto:200904100006.51201.shocky1@users.sourceforge.net"><a href="mailto:200904100006.51201.shocky1@users.sourceforge.net">200904100006.51201.shocky1@users.sourceforge.net</a></a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
Hi,<br>
<br>
This is probably outside what Asterisk is intended for, but I'm hoping it can<br>
help.<br>
<br>
I need to make and receive calls through a Cisco Call Manager server that I<br>
have no control over. I have to use a Cisco soft phone (Cisco IP<br>
Communicator), which only runs on Windows. But I'm on Linux. CCM is<br>
apparently capable of supporting SIP and H.323 interfaces, but they won't<br>
provide this option for me. Right now I'm using a VMWare XP guest to run the<br>
soft phone, but this is painful (especially with some VPN complications<br>
thrown in).<br>
<br>
I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I<br>
could set up Asterisk on my desktop machine to route calls between a SIP<br>
client such as Kphone or Ekiga and the CCM server. Would this be possible?<br>
<br>
I heard that one of the problems in interfacing with CCM over SCCP is the use<br>
of proprietary codecs. Would this be a problem in my case?<br>
<br>
If there's a chance it can be made to work, I'll give it a try. If I'd be<br>
wasting my time, please let me know.<br>
<br>
Thanks,<br>
<br>
Shocky<br>
--<br>
These are my opinions. Get your own.<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 17<br>
Date: Fri, 10 Apr 2009 10:07:38 +0300<br>
From: Tzafrir Cohen <<a href="mailto:tzafrir.cohen@xorcom.com"><a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a></a>><br>
Subject: Re: [asterisk-users] MeetMe not working - was before<br>
To: <a href="mailto:asterisk-users@lists.digium.com"><a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></a><br>
Message-ID: <<a href="mailto:20090410070738.GS3227@xorcom.com"><a href="mailto:20090410070738.GS3227@xorcom.com">20090410070738.GS3227@xorcom.com</a></a>><br>
Content-Type: text/plain; charset=us-ascii<br>
<br>
On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote:<br>
> When I dial the extension of a meetme conference room, I get a message that<br>
> states "is not a valid conference". The meetme app was working before.<br>
><br>
> I am getting this error on the CLI:<br>
> app_meetme.c:800 build_conf: Unable to open pseudo device<br>
><br>
> I have Asterisk 1.4.23.1 and zaptel-1.4.11<br>
<br>
Elsewhere you mentioned you also have dahdi installed. What is the<br>
output of:<br>
<br>
ls /usr/include/dahdi<br>
<br>
I suspect Asterisk was built vs. dahdi whereas Zaptel was actually<br>
running.<br>
<br>
Actual tests:<br>
<br>
dahdi_test<br>
<br>
vs.<br>
<br>
zttest<br>
<br>
--<br>
Tzafrir Cohen<br>
icq#16849755 <a href="mailto:jabber%3Atzafrir.cohen@xorcom.com">jabber:<a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a></a><br>
+972-50-7952406 mailto:<a href="mailto:tzafrir.cohen@xorcom.com"><a href="mailto:tzafrir.cohen@xorcom.com">tzafrir.cohen@xorcom.com</a></a><br>
<a href="http://www.xorcom.com" target="_blank"><a href="http://www.xorcom.com">http://www.xorcom.com</a></a> <a href="http://iax:guest@local.xorcom.com/tzafrir" target="_blank">iax:<a href="mailto:guest@local.xorcom.com">guest@local.xorcom.com</a>/tzafrir</a><br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 18<br>
Date: Fri, 10 Apr 2009 10:33:36 +0100 (BST)<br>
From: Gordon Henderson <<a href="mailto:gordon%2Basterisk@drogon.net"><a href="mailto:gordon+asterisk@drogon.net">gordon+asterisk@drogon.net</a></a>><br>
Subject: Re: [asterisk-users] Can Asterisk bridge between a SIP client<br>
and a Cisco Call Manager server?<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com"><a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a></a>><br>
Message-ID: <<a href="mailto:Pine.LNX.4.64.0904101032040.23406@unicorn.drogon.net"><a href="mailto:Pine.LNX.4.64.0904101032040.23406@unicorn.drogon.net">Pine.LNX.4.64.0904101032040.23406@unicorn.drogon.net</a></a>><br>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed<br>
<br>
On Fri, 10 Apr 2009, Shocky wrote:<br>
<br>
> Hi,<br>
><br>
> This is probably outside what Asterisk is intended for, but I'm hoping it can<br>
> help.<br>
><br>
> I need to make and receive calls through a Cisco Call Manager server that I<br>
> have no control over. I have to use a Cisco soft phone (Cisco IP<br>
> Communicator), which only runs on Windows. But I'm on Linux. CCM is<br>
> apparently capable of supporting SIP and H.323 interfaces, but they won't<br>
> provide this option for me. Right now I'm using a VMWare XP guest to run the<br>
> soft phone, but this is painful (especially with some VPN complications<br>
> thrown in).<br>
><br>
> I've read that Asterisk supports SCCP, at least somewhat. I'm wondering if I<br>
> could set up Asterisk on my desktop machine to route calls between a SIP<br>
> client such as Kphone or Ekiga and the CCM server. Would this be possible?<br>
><br>
> I heard that one of the problems in interfacing with CCM over SCCP is the use<br>
> of proprietary codecs. Would this be a problem in my case?<br>
><br>
> If there's a chance it can be made to work, I'll give it a try. If I'd be<br>
> wasting my time, please let me know.<br>
<br>
I've never looked at SCCP, but if it does work then you could use the<br>
console phone built into asterisk rather than IP plumb it into a<br>
soft-phone... So asterisk is essentially acting as an SCCP soft-phone<br>
itself. No GUI though, but if you're happy typing commands... :)<br>
<br>
Gordon<br><br>
</blockquote></div><br><br clear="all"><br>
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