I need to pick all the Asterisk and Cisco People a little. My company has a Cisco Call Manager 3.3, configured via h323 gateways. I have remote users that I want to place a SIP Server on the external WAN and be able to connect their phones to the system and be able to get calls and call people in the office going through the Cisco Call Manager and the h323 router. My only problem is that Cisco Call Manager 3.3 does not support sip trunking. Is there anyway this can be done. Please shed some light on this topic. Thanks. Goran
Peder @ NetworkOblivion
2005-Oct-14 09:09 UTC
[Asterisk-Users] Asterisk/Cisco Call Manager 3.3
You can use H.323 on Asterisk and setup CM to use an H.323 gateway to Asterisk, or setup a gatekeeper and have both ends talk to the gatekeeper. I have redundant CM boxes, so I HAD to use a gatekeeper and set it in proxy mode because I had media path issues (the call initiated from one box, but for some reason the media path on outside calls came from the other box and we had one way audio). If there is just one CM box, then you probably don't need a gatekeeper. gorand@dvvti.com wrote:> I need to pick all the Asterisk and Cisco People a little. > > My company has a Cisco Call Manager 3.3, configured via h323 gateways. I > have remote users that I want to place a SIP Server on the external WAN > and be able to connect their phones to the system and be able to get calls > and call people in the office going through the Cisco Call Manager and the > h323 router. My only problem is that Cisco Call Manager 3.3 does not > support sip trunking. Is there anyway this can be done. > > Please shed some light on this topic. > > Thanks. > > Goran > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Network stuff you didn't know.... http://www.networkoblivion.com
> > > Message: 13 > Date: Fri, 14 Oct 2005 09:58:37 -0500 (CDT) > From: <gorand@dvvti.com> > Subject: [Asterisk-Users] Asterisk/Cisco Call Manager 3.3 > To: <asterisk-users@lists.digium.com> > Message-ID: <4408.69.215.189.2.1129301917.squirrel@www.dvvti.com> > Content-Type: text/plain; charset=iso-8859-1 > > I need to pick all the Asterisk and Cisco People a little. > > My company has a Cisco Call Manager 3.3, configured via h323 gateways. I > have remote users that I want to place a SIP Server on the external WAN > and be able to connect their phones to the system and be able to get calls > and call people in the office going through the Cisco Call Manager and the > h323 router. My only problem is that Cisco Call Manager 3.3 does not > support sip trunking. Is there anyway this can be done. > > Please shed some light on this topic. > > Thanks. > > Goran > > Goran-Speaking from experience, you have a tough road ahead of you. The only way to accomplish this is via h.323 trunks under Cisco and Asterisk. There are a few known good configurations- I can really only speak to one, as it eventually worked for me- but others may have different and perfectly reasonable advice. First, some prerequisites: 1. Asterisk 1.2 or CVS HEAD. Do NOT try this with any of the 1.0X series- you will be able to call from CCM to Asterisk, but not from Asterisk to CCM. 2. An H323 Gatekeeper. GnuGK works, but does occasionally bonk out. CCM will send RRQ requests to the gatekeeper at a rate of 10x per second, and eventually, GnuGK loses it. An IOS gatekeeper seems to be much better. 3. chan_h323 set up and running properly. There's whole readme files on the prerequisites for this- read them, follow the directions closely- and call on JerJer *LAST* if you value your life. 4. A Gatekeeper controlled Trunk on CCM. The tricky bits here are the significant digits, and the technology prefix. CCM does *NOT* register the tech prefix or it's extensions with the gatekeeper- so you'll have to config the gatekeeper to know where to send the call, and you'll have to configure your CCM dialplan to act accordingly. Set this up slowly. Get a working Asterisk box that's able to handle softphones or hardphones as an island PBX, then configure the H323 trunk- you'll save some frustration of trying to configure both simultaneously. Find me on the IRC channel if you need specific questions answered- or email me directly. I can optionally configure it for you for a fee- I'm based in the US, and judging from your accent, I'd say you aren't- I can do this remotely if needed. I won't charge you for questions answered. :) -pbd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051014/49ea78c6/attachment.htm