Re: [Asterisk-Users] Pickup() h323
Hamid Hashemi wrote: I did try it again without success. I did check the debug logs and there is nothing special there about any errors. Following the logs it says that the connection is established but no Voice and no Tone. here is my scenario : I have a SIP phone which make a SIP call to asterisk with G729 Codec. The Asterisk then make an H323 call to the external peer with G729 codec again and it should make bridge between these 2 calls ( 1 incomming and 1 outgoing ) I checked it with OH323 with the same scenario and it is working well. But for H323 I couldn't make the call. Any idea ? Without debug there not even a chance that ANYONE can assist you. Jeremy McNamara ___ --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] Pickup() h323
I did try it again without success. I did check the debug logs and there is nothing special there about any errors. Following the logs it says that the connection is established but no Voice and no Tone. here is my scenario : I have a SIP phone which make a SIP call to asterisk with G729 Codec. The Asterisk then make an H323 call to the external peer with G729 codec again and it should make bridge between these 2 calls ( 1 incomming and 1 outgoing ) I checked it with OH323 with the same scenario and it is working well. But for H323 I couldn't make the call. Any idea ? Hamid Hashemi wrote: Ok I will try it again and will let the list know about the result. Jeremy McNamara wrote: Hamid Hashemi wrote: Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . Yes, chan_h323 will pass thru any codec as long as your configuration and the 'far-end' allows the proper codec. Also, you cannot have any Dial options (Ttr etc) and you cannot play any prompts. Just Dial,H323/your_location If you have trouble turn on H.323 debug and have debug on the console line in logger.conf. Jeremy McNamara ___ --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 -- Regards = / Seyyed Hamid Reza/WINDOWS FOR NOW !!/ / Hashemi Golpayegani / Linux for future , FreeBSD for ever / /Morva System Co. / - / / Network Administrator/ [EMAIL PROTECTED] , ICQ# : 42209876 / ___ --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] Pickup() h323
Pavel Jezek wrote: Hello Jeremy, do you think, that adding features to original h323 channel is perspective? is still maintained or will be replaced eg. with ooh323 (from asterisk add-ons)? anyway I'm currently using original h323, it working prety fine for me (with ooh323/oh323 I had problem with callerid between h323-asterisk)... chan_h323 is very much supported, just nobody has bothered to give me any valid information on what needs to be fixed. I have totally removed H.323 from my operation, so I no longer utilize chan_h323 for anything. Thus it is now up to the community to report issues they find. Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Jeremy McNamara ___ --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] Pickup() h323
On 04/06/06 04:41 Dan Austin said the following: Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. ok, i'm not getting into a my H323 is better than yours argument, but we've been struggling to get OOH323 working with OHPHONE. symptoms are that calls from SIP -- OhPhone work fine, but when OhPhone calls SIP, the call is hungup the moment the SIP phone answers. any clues why ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Pickup() h323
Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . _Hamid Jeremy McNamara wrote: Pavel Jezek wrote: Hello Jeremy, do you think, that adding features to original h323 channel is perspective? is still maintained or will be replaced eg. with ooh323 (from asterisk add-ons)? anyway I'm currently using original h323, it working prety fine for me (with ooh323/oh323 I had problem with callerid between h323-asterisk)... chan_h323 is very much supported, just nobody has bothered to give me any valid information on what needs to be fixed. I have totally removed H.323 from my operation, so I no longer utilize chan_h323 for anything. Thus it is now up to the community to report issues they find. Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Jeremy McNamara ___ --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
Re: [Asterisk-Users] Pickup() h323
Ok I will try it again and will let the list know about the result. Jeremy McNamara wrote: Hamid Hashemi wrote: Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . Yes, chan_h323 will pass thru any codec as long as your configuration and the 'far-end' allows the proper codec. Also, you cannot have any Dial options (Ttr etc) and you cannot play any prompts. Just Dial,H323/your_location If you have trouble turn on H.323 debug and have debug on the console line in logger.conf. Jeremy McNamara ___ --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 -- Regards = / Seyyed Hamid Reza/WINDOWS FOR NOW !!/ / Hashemi Golpayegani / Linux for future , FreeBSD for ever / /Morva System Co. / - / / Network Administrator/ [EMAIL PROTECTED] , ICQ# : 42209876 / ___ --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] Pickup() h323
Hamid Hashemi wrote: Hi , Is your chan_h323 driver can support codec g729 or g723 in bridge mode ( I mean no transcoding just bridging ) ? I did try it without success however both chan_oh323 and chan_ooh323 can support it without any problem . Yes, chan_h323 will pass thru any codec as long as your configuration and the 'far-end' allows the proper codec. Also, you cannot have any Dial options (Ttr etc) and you cannot play any prompts. Just Dial,H323/your_location If you have trouble turn on H.323 debug and have debug on the console line in logger.conf. Jeremy McNamara ___ --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] Pickup() h323
I have exactly oposite experience: I'm using in two sites one with callmanager (v4), one with gnugk gatekeeper and ci$co h323 gateways with ooh323 - caller id name is not go through, when calling from asterisk to callmanager, sending dtmf from callmanger to asterisk isn't funtional with oh323 - caller id isn't working, dtmf OK, but problems with oh323 registering to two gatekeepers, also seems, that oh323 isn't good maintained, look at bugtracker and last release date :-( original h.323 is working prety good for me on site with callmanager and on site with dual gatekeepers (except pickup via this channel, as I wrote at beginning, but maybe this be implemented soon ;-) PJ Dan Austin wrote: Better is subjective in this case. There's no doubt that chan_ooh323 has some warts. On the other hand it has NO external library requirements, and works out of the box with Cisco's Call Manager. One could argue that Call Manager is crap. Fine, that doesn't change the fact some of us are stuck with it. Chan_h323 did not work with CCM, and a query/bug report was dismissed, basically stating that Cisco was F'd up and the channel would not be updated to work with it unless funded. (fair, but not helpful) Chan_oh323 worked with CCM, but suffered from the external library requirements. Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. So for me chan_ooh323 is a 'better' H.323 channel driver. 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 ___ --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] Pickup() h323
On 04/06/06 04:41 Dan Austin said the following: Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. Dinesh wrote: ok, i'm not getting into a my H323 is better than yours argument, but we've been struggling to get OOH323 working with OHPHONE. symptoms are that calls from SIP -- OhPhone work fine, but when OhPhone calls SIP, the call is hungup the moment the SIP phone answers. any clues why ? I did not intend to get that ball rolling. Each of the H.323 channels have at least one area that they work better than the others. The key is to find out which one works better for your needs. As to your problem, I have nothing specific, but I suspect codec/bearer capability issues, which is one area of weakness in ooh323. We only use it for trunking to CCM and have no H.323 clients. I have had excellent results with running a debug build of chan_ooh323 and submitting the logs to the ooh323 developer list. 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] Pickup() h323
Dan Austin wrote: Chan_h323 did not work with CCM, and a query/bug report was dismissed, basically stating that Cisco was F'd up and the channel would not be updated to work with it unless funded. (fair, but not helpful) chan_h323 most certainly does work with CCM. I have customers using it all day (and night) long. Jeremy McNamara ___ --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] Pickup() h323
Dan Austin wrote: Chan_h323 did not work with CCM, and a query/bug report was dismissed, basically stating that Cisco was F'd up and the channel would not be updated to work with it unless funded. (fair, but not helpful) chan_h323 most certainly does work with CCM. I have customers using it all day (and night) long. OK, so I should have said the last time I tried to use it. The problem was one-way audio, and when I asked if it was a known issue, or there were any work arounds I was told it was a known issue and the work around was to not use CCM. I do not recall seeing a note that the issue had been addressed and I did the only thing I could at the time, which was to look elsewhere for H.323. Again I am not trying to knock any of the H.323 channels. They each have their strengths and the system administrator should use the one that works best for them. And in case it is not perfectly clear, I appreciate your efforts on both the H.323 and SCCP channels. 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] Pickup() h323
Dan Austin wrote: Again I am not trying to knock any of the H.323 channels. They each have their strengths and the system administrator should use the one that works best for them. The one thing nobody ever realizes is there there is very very limited interoperability in H.323.It is not any one particular channel drivers fault, it is the underlying H.323 stack and the recommendation passed down from the ITU - Which is not a specification, hence the problems everyone has with H.323. There was too much left open for interpretation. Again, everyone, stop blaming any one particular channel driver in Asterisk and start blaming H.323 itself. If for some crazy reason you still must run H.323, then you will always run into problems, no matter what H.323-based gear or software you utilize. Jeremy McNamara ___ --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] Pickup() h323
Jeremy McNamara wrote: Digium paid for ooh323, for whatever reasons that is beyond me, but it has proven to be no better than any H.323 channel driver, so I hope they got their money back. Better is subjective in this case. There's no doubt that chan_ooh323 has some warts. On the other hand it has NO external library requirements, and works out of the box with Cisco's Call Manager. One could argue that Call Manager is crap. Fine, that doesn't change the fact some of us are stuck with it. Chan_h323 did not work with CCM, and a query/bug report was dismissed, basically stating that Cisco was F'd up and the channel would not be updated to work with it unless funded. (fair, but not helpful) Chan_oh323 worked with CCM, but suffered from the external library requirements. Chan_ooh323 just worked. The code is, to a infrequent programmer, easy to read, extend and fix bugs. So for me chan_ooh323 is a 'better' H.323 channel driver. 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] Pickup() h323
Hello Jeremy, do you think, that adding features to original h323 channel is perspective? is still maintained or will be replaced eg. with ooh323 (from asterisk add-ons)? anyway I'm currently using original h323, it working prety fine for me (with ooh323/oh323 I had problem with callerid between h323-asterisk)... PJ Jeremy McNamara wrote: Pavel Jezek wrote: Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? I am going to say its not currently supported. If you provide me (offlist) some info on how to support it, I will find the time to attempt an implementation. Turn on H.323 debug and see what you get. Jeremy McNamara ___ --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] Pickup() h323
Hello,
I can use directed call pickup using pickup application (between sip,
iax, skinny cals),
but unable to pickup call that is ringing on phone behind h323 gateway
(using original h323 channel in asterisk), is this even suported?
thx
PJ
exten = _*7.,1,Pickup(${EXTEN:2})
console log, when trying o pickup ringing line 324 (h323), from skinny
phone (953)
-- Executing Pickup(SCCP/953-0004, 324) in new stack
== Spawn extension (default, *7324, 1) exited non-zero on
'SCCP/953-0004'
-- SCCP: Asterisk request to hangup channel SCCP/953-0004
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Re: [Asterisk-Users] Pickup() h323
Pavel Jezek wrote:
Hello,
I can use directed call pickup using pickup application (between sip,
iax, skinny cals),
but unable to pickup call that is ringing on phone behind h323 gateway
(using original h323 channel in asterisk), is this even suported?
thx
PJ
exten = _*7.,1,Pickup(${EXTEN:2})
console log, when trying o pickup ringing line 324 (h323), from skinny
phone (953)
-- Executing Pickup(SCCP/953-0004, 324) in new stack
== Spawn extension (default, *7324, 1) exited non-zero on
'SCCP/953-0004'
-- SCCP: Asterisk request to hangup channel SCCP/953-0004
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The directed pickup application needs a CDR record on the channel, so be
sure that the H323 channel driver is creating one. You may need to set
amaflags to have it happen. I remember this being found before by
someone else, and that being there way to make it work.
Joshua Colp
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Re: [Asterisk-Users] Pickup() h323
Yes, my asterisk generates cdr when call from sip/iax/sccp to h323
master.csv
,953,324,default,
953,SCCP/953-001a,H323/ccm-6,Dial,H323/[EMAIL PROTECTED],2006-04-03
16:36:44,,2006-04-03 16:37:05,21,0,NO ANSWER,DOCUMENTATION
and my h323.conf is quite simple:
[general]
port = 1720
bindaddr = 192.168.40.4
disallow=all
allow=g729
allow=alaw
[ccm]
type=peer
host=192.168.40.7
noFastStart=yes
PJ
Joshua Colp wrote:
Pavel Jezek wrote:
Hello,
I can use directed call pickup using pickup application (between sip,
iax, skinny cals),
but unable to pickup call that is ringing on phone behind h323
gateway (using original h323 channel in asterisk), is this even
suported?
thx
PJ
exten = _*7.,1,Pickup(${EXTEN:2})
console log, when trying o pickup ringing line 324 (h323), from
skinny phone (953)
-- Executing Pickup(SCCP/953-0004, 324) in new stack
== Spawn extension (default, *7324, 1) exited non-zero on
'SCCP/953-0004'
-- SCCP: Asterisk request to hangup channel SCCP/953-0004
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The directed pickup application needs a CDR record on the channel, so
be sure that the H323 channel driver is creating one. You may need to
set amaflags to have it happen. I remember this being found before by
someone else, and that being there way to make it work.
Joshua Colp
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___
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Re: [Asterisk-Users] Pickup() h323
Pavel Jezek wrote: Hello, I can use directed call pickup using pickup application (between sip, iax, skinny cals), but unable to pickup call that is ringing on phone behind h323 gateway (using original h323 channel in asterisk), is this even suported? I am going to say its not currently supported. If you provide me (offlist) some info on how to support it, I will find the time to attempt an implementation. Turn on H.323 debug and see what you get. Jeremy McNamara ___ --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
