RE: [asterisk-users] Cisco CAll Manger and H323
On Fri, 2006-09-29 at 20:26 -0700, Dan Austin wrote: > Greg wrote: > > 4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. > > Anything over 4.0 supports SIP trunking. > > While it is true that CCM 4.0 and up supports SIP trunking, it is not > all rainbows an butterflies. The 4.X series implimentation of SIP > requires a MTP, as the SCCP endpoints do not support standards based > DTMF. Additionally only ulaw is supported. > > In 5.X the SCCP endpoints now support RFC2833, but if you have Unity > or IOS gateways, switching everything to SIP is not trivial. So for > some implimentations a decent H323 channel driver is still the best > option for integration. That is true, but the CCM itself is an MTP as long as you start the MoH or Media Streaming services on it.. I will say that I have not tried the 323 channels in a while (about 8 months or so), but once we switched to SIP, we have had zero issues whereas before, every 2-3 weeks, * would hiccup requiring a restart.. Our CCM phones are all Skinny since we develop products for CCM - only using * for vmail, meetme, IVR, etc.. All of our remote (all Cisco) phones use SIP connected directly to *.. 5.x CCM is much more SIP capable moreso than just the phones, but Cisco has once again done it and sends all kinds of proprietary INVITES to enable some more features on their phones.. Kinda dissapointing... I love the phones, but hate the company.. -Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco CAll Manger and H323
Greg wrote: > 4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. > Anything over 4.0 supports SIP trunking. While it is true that CCM 4.0 and up supports SIP trunking, it is not all rainbows an butterflies. The 4.X series implimentation of SIP requires a MTP, as the SCCP endpoints do not support standards based DTMF. Additionally only ulaw is supported. In 5.X the SCCP endpoints now support RFC2833, but if you have Unity or IOS gateways, switching everything to SIP is not trivial. So for some implimentations a decent H323 channel driver is still the best option for integration. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CAll Manger and H323
4.1.3 supports SIP trunks - I would HIGHLY recommend you move to that. Anything over 4.0 supports SIP trunking. -Greg On Thu, 2006-09-28 at 19:32 +0200, Yusuf wrote: > Hi, > > I recently had to hook up to Cisco Call Manager 4.1.3, and it only > supports H323. SO I used ooh323, and a strange thing happens. When a > Cisco IP user calls from his phone, the call gets sent from Call Manager > to Asterisk, but the phone will ring once only, then it seems asterisk > will drop the call, and int he debug it says: "stopped from reciving > frames from OOH323/cisco , bridging is being stopped". > > What is wrong? > > What RTP ports must I be using? > > thanks, > yusuf > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco CAll Manger and H323
Thanks Dan, that was awesome, and really made sense about what was really happening. :) Will try a newer. BTW: I did get it to successfully route inbound calls to asterisk with oh323, and DTMF and transfers worked fine. > Yusuf wrote: >> Hi Dan, > >> I used asterisk 1.2.10 with asterisk-addons 1.2.3. >> I did two successfull calls, but with dtmf=rfc2833, dtmf was not >> sending at all. Then when I made some changes, I could not get any >> calls to go through. The call would just hangup after first ring. > Call Manager's support for RFC2833 is 'lacking'. It works reasonably > well in 5.X for SIP, but forget about using it with H323. I've tested > all four options with Call Manager, and only q931keypad and h245signal > worked. I'd recommend using h245signal. > >> Did you get calls going in both ways, inbound and outbound to > asterisk. >> O got two calls going from CAllmanger to asterisk only, other would > not >> work. > > Calls work both ways, although 99.99% of my calls are inbound, since > we use Asterisk for conferencing only at this point. > > Here are a couple of ideas to try: > 1. Set the Call Manager H323 gateway to 'Require MTP' > 2. Set DTMF to h245 signal > > What is likely happening is that with Asterisk asking for RFC2833, > CCM tries to invoke a MTP. I am not sure in which Asterisk-Addons > version it was added, but I wrote support for Empty Terminal Capability > sets for chan_ooh323. If that feature is not in the version you have, > (chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on > the CCM gateway you'll see a problem like you have. > > I should also point out that if you are not running chan_ooh323 0.5 or > newer and do get Asterisk to accept calls with out forcing an MTP, calls > will be dropped anytime a CCM endpoint uses hold or transfer features. > > If you do have 0.5 or newer, then changing the DTMF migth be enough. > > Hope this helps, > Dan > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > > thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco CAll Manger and H323
Yusuf wrote: > Hi Dan, > I used asterisk 1.2.10 with asterisk-addons 1.2.3. > I did two successfull calls, but with dtmf=rfc2833, dtmf was not > sending at all. Then when I made some changes, I could not get any > calls to go through. The call would just hangup after first ring. Call Manager's support for RFC2833 is 'lacking'. It works reasonably well in 5.X for SIP, but forget about using it with H323. I've tested all four options with Call Manager, and only q931keypad and h245signal worked. I'd recommend using h245signal. > Did you get calls going in both ways, inbound and outbound to asterisk. > O got two calls going from CAllmanger to asterisk only, other would not > work. Calls work both ways, although 99.99% of my calls are inbound, since we use Asterisk for conferencing only at this point. Here are a couple of ideas to try: 1. Set the Call Manager H323 gateway to 'Require MTP' 2. Set DTMF to h245 signal What is likely happening is that with Asterisk asking for RFC2833, CCM tries to invoke a MTP. I am not sure in which Asterisk-Addons version it was added, but I wrote support for Empty Terminal Capability sets for chan_ooh323. If that feature is not in the version you have, (chan_ooh323 release 0.5 or newer), and you are not forcing an MTP on the CCM gateway you'll see a problem like you have. I should also point out that if you are not running chan_ooh323 0.5 or newer and do get Asterisk to accept calls with out forcing an MTP, calls will be dropped anytime a CCM endpoint uses hold or transfer features. If you do have 0.5 or newer, then changing the DTMF migth be enough. Hope this helps, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco CAll Manger and H323
Hi Dan, I used asterisk 1.2.10 with asterisk-addons 1.2.3. I did two successfull calls, but with dtmf=rfc2833, dtmf was not sending at all. Then when I made some changes, I could not get any calls to go through. The call would just hangup after first ring. Did you get calls going in both ways, inbound and outbound to asterisk. O got two calls going from CAllmanger to asterisk only, other would not work. Dan Austin wrote: I've used chan_ooh323 with Call Manager version 3.3, 4.0, 4.1 and now 5.0 with great success. Which version of Asterisk-addons are you using and which version of Asterisk? I have a very simple config. I seem to remember an issue if bindaddr was not set, or left to 0.0.0.0, but I might be thinking of another channel. I have faststart set to no and dtmfmode set to h245signal with all other settings using the defaults. Let me know if this doesn't help, I can try to provide more details. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: Thursday, September 28, 2006 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco CAll Manger and H323 Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: "stopped from reciving frames from OOH323/cisco , bridging is being stopped". What is wrong? What RTP ports must I be using? thanks, yusuf -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco CAll Manger and H323
I've used chan_ooh323 with Call Manager version 3.3, 4.0, 4.1 and now 5.0 with great success. Which version of Asterisk-addons are you using and which version of Asterisk? I have a very simple config. I seem to remember an issue if bindaddr was not set, or left to 0.0.0.0, but I might be thinking of another channel. I have faststart set to no and dtmfmode set to h245signal with all other settings using the defaults. Let me know if this doesn't help, I can try to provide more details. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: Thursday, September 28, 2006 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco CAll Manger and H323 Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: "stopped from reciving frames from OOH323/cisco , bridging is being stopped". What is wrong? What RTP ports must I be using? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco CAll Manger and H323
Hi, I recently had to hook up to Cisco Call Manager 4.1.3, and it only supports H323. SO I used ooh323, and a strange thing happens. When a Cisco IP user calls from his phone, the call gets sent from Call Manager to Asterisk, but the phone will ring once only, then it seems asterisk will drop the call, and int he debug it says: "stopped from reciving frames from OOH323/cisco , bridging is being stopped". What is wrong? What RTP ports must I be using? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
