Vidura Senadeera
2009-Apr-16 11:17 UTC
[asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call
> > 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<jabber%3Atzafrir.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<gordon%2Basterisk 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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090416/af95324c/attachment.htm
Alex Balashov
2009-Apr-16 11:22 UTC
[asterisk-users] Can Asterisk bridge between a SIP client and a Cisco Call
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 > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090416/c75082ac/attachment.htm