Re: [asterisk-users] H.323 video support
On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: Hi list! I asked this in this list some time ago, and now I was searching for evolution about this subject, but I found nothing. Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta brunch? If not, is this in the roadmap for 1.6 brunch? Regards. Diego. When and where is the 1.6 brunch? ;-) Steve T ___ -- 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
Re: [asterisk-users] H.323 video support
Yes, you are right... sorry for my fast and poor English. I rewrite my questions: Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta branch? If not, is this in the roadmap for 1.6 branch? Regards. 2008/5/23 Steve Totaro [EMAIL PROTECTED]: On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: Hi list! I asked this in this list some time ago, and now I was searching for evolution about this subject, but I found nothing. Nowadays, what is the state for H.323 video support? Is there support in the 1.6 beta brunch? If not, is this in the roadmap for 1.6 brunch? Regards. Diego. When and where is the 1.6 brunch? ;-) Steve T ___ -- 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 ___ -- 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
Re: [asterisk-users] H.323 video support
Remind me to pick on your poor Spanish next time I see you for a mid-morning meal. :) Steve Totaro wrote: On Fri, May 23, 2008 at 4:05 AM, Diego Moreno [EMAIL PROTECTED] wrote: When and where is the 1.6 brunch? ;-) ___ -- 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
Re: [asterisk-users] H.323
Hi, I have used h323, oh323 and ooh323. My experience is that ooh323 does not work properly, i dont recommend it. I dont know why, but the sound is bad, with sound breaks. I also need to put some wait (2) functions after the answer( ) or playback( ) functions, it think that asterisk takes some time to stablish the ooh323 channel (maybe it is due to other reason, i dont know exactly) but during this time no sound is played, so the first seconds of conversation or playback are cutted. ooh323 did not work for me at all. oh332 worked fine in asterisk 1.2 (i did not try in 1.4, i guess it is fine). h323 works fine in asterisk 1.4. it is the one i am using now, and i have no problems with it. bye now 2007/8/2, Rurouni Alucard [EMAIL PROTECTED]: Hi there, I have use the H.323 module that comes with asterisk-addons and i consider it (so far) VERY stable for my needs. Im talking about 10,000 minutes at month , + or - , and never had a crash or something bad about it. Personally, i recommend it, -- J. P. rakh at slackware-es dot com bilal ghayyad wrote: Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --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 ___ --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 ___ --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
Re: [asterisk-users] H.323
Hi, I'm using H323 in asterisk 1.4.9 work well On 8/3/07, yonoko molomo [EMAIL PROTECTED] wrote: Hi, I have used h323, oh323 and ooh323. My experience is that ooh323 does not work properly, i dont recommend it. I dont know why, but the sound is bad, with sound breaks. I also need to put some wait (2) functions after the answer( ) or playback( ) functions, it think that asterisk takes some time to stablish the ooh323 channel (maybe it is due to other reason, i dont know exactly) but during this time no sound is played, so the first seconds of conversation or playback are cutted. ooh323 did not work for me at all. oh332 worked fine in asterisk 1.2 (i did not try in 1.4, i guess it is fine). h323 works fine in asterisk 1.4. it is the one i am using now, and i have no problems with it. bye now 2007/8/2, Rurouni Alucard [EMAIL PROTECTED]: Hi there, I have use the H.323 module that comes with asterisk-addons and i consider it (so far) VERY stable for my needs. Im talking about 10,000 minutes at month , + or - , and never had a crash or something bad about it. Personally, i recommend it, -- J. P. rakh at slackware-es dot com bilal ghayyad wrote: Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --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 ___ --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 ___ --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 -- Alessandro R. ___ --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
Re: [asterisk-users] H.323
Hi there, I have use the H.323 module that comes with asterisk-addons and i consider it (so far) VERY stable for my needs. Im talking about 10,000 minutes at month , + or - , and never had a crash or something bad about it. Personally, i recommend it, -- J. P. rakh at slackware-es dot com bilal ghayyad wrote: Hi List; Did any one tried the H.323 module? How much it is stable and work fine? Regards, ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460 Ready for the edge of your seat? Check out tonight's top picks on Yahoo! TV. http://tv.yahoo.com/ ___ --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 ___ --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
Re: [asterisk-users] H.323 IP Phones and H.323 Traffic
You need install the asterisk h323 drivers. You can get them in the asterisk-addons. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, June 22, 2007 9:49 PM Subject: [asterisk-users] H.323 IP Phones and H.323 Traffic Hi List; I saw sip.conf and iax.conf but I do not see a files for H.323 IP Phones, does that mean Asterisk does not support H.323 IP Phones? Also, what if Asterisk need to talk with another IP PBX that support H.323, so the IP Trunk in that case should be H.323 IP Trunk, does Asterisk support such thing? Regards Bilal Ghayad Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --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 ___ --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
Re: [asterisk-users] H.323 an IAX
On 24 Jul 2006, at 14:24, Asif Ali wrote: Hi I have a problem with the NAT using H.323 and am thinking of employing IAX as a workwround. I have a scenario in my mind which I am not sure is gonna work or not, neaways here it goes. I want my IAX clients to connect to Asterisk which will be interconnected with H.323 terminating gateways. Now my clients will place calls using IAX and Asterisk will establish connection with H. 323 gateways but WOULD NOT relay media, i.e. I dont want to proxy media and media would be between endpoint to endpoint. Can Asterisk do this protocol conversion transparently? Is this practically possible? and would I be able to achieve NAT tarversal(at client end) using above mentioned scenaio? as I have read about IAX that it does NAT traversal. My Asterisk would run on public IP Any suggestion or help in this regard will be highly appreciated. Thanx in advance No, you cant mix and match IAX signaling and H323 media. One of the reasons that IAX does NAT so nicely is that it uses the same 'connection' (ie UDP port) for control and media, but that also means it does not do RTP media. On the otherhand if you are prepared to have asterisk proxy the media as well as the control, then yes you can do what you need. Tim Panton www.mexuar.com ___ --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] H.323 soft phone known to be run with asterisk.
I had problems with sjphone ... same version as yours. Finally, i managed to solve it by: - in sjphone, media channels settings: untick Use remote codec preferences and Open audio streams after remote opened ... it was trial-error ... now it works (to Echo and Sip-H323 call). - in asterisk, h323.conf ... the codec configuration ... i commented all lines related to it ... ;disallow=all ;allow=all ;allow=gsm ;disallow=g723.1 (just in case) (again, trial-error) Cesc On 6/21/06, Pawel [EMAIL PROTECTED] wrote: Hallo group members Could You tell me a h.323 soft phone that runs well with asterisk. I tried the following so far, but in general I cannot compile them (fc.3) or I cannot configure them to run with asterisk: http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure http://cphone.sourceforge.net/ - cannot compile http://www.ekiga.org/ - cannot compile http://www.openh323.org/ - cannot compile Greetings. ___ --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] H.323 soft phone known to be run with asterisk.
Hallo Cesc Cesc writes: I had problems with sjphone ... same version as yours. Finally, i managed to solve it by: - in sjphone, media channels settings: untick Use remote codec preferences and Open audio streams after remote opened ... it was trial-error ... now it works (to Echo and Sip-H323 call). - in asterisk, h323.conf ... the codec configuration ... i commented all lines related to it ... ;disallow=all ;allow=all ;allow=gsm ;disallow=g723.1 (just in case) (again, trial-error) Cesc On 6/21/06, Pawel [EMAIL PROTECTED] wrote: Hallo group members Could You tell me a h.323 soft phone that runs well with asterisk. I tried the following so far, but in general I cannot compile them (fc.3) or I cannot configure them to run with asterisk: http://www.pacphone.com/downloads/PacPhoneSetup117.exe - cannot configure http://www.sjlabs.com/softphone/SJphone-289a.exe - cannot configure http://cphone.sourceforge.net/ - cannot compile http://www.ekiga.org/ - cannot compile http://www.openh323.org/ - cannot compile Greetings. ___ --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 Your suggestions helped. Thanks a lot! Greetings ___ --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] H.323 ( HW PBX to *)
Hi, I'm trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with other PBX. The port use to connect is TCP 1720 but I can't configure this port on my * box. I'm using a H.323.conf file sample to activate the port but the * isn't listening there. Somebody have any idea or tip? This is mi H.323.conf Hi, just checking the basics here: have you opened 1720 on the firewall? which are you using - oh323 or ooh323 - as in the inaccesnetworks oh323 or asterisk-addons ooh323. when you do a 'help' on cli do you see *h323? is asterisk seeing * *h323 loaded as a module? if yes to above, what a bout a h323 debug. -- 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] H.323 call '.....' cleared,reason 8 (Transport failure)
Cenk,
Are you sure that remote will handle H245
tunneling? If the remote does not know how to do that, you will get transport
failure. I would suggest doing FastStart instead and
see if you are getting the same results. Of course, you can verify that the
remote can handle faststart as well.
Alex
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Cenk Yabas
Sent: Saturday, April 02, 2005
6:20 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] H.323
call '.' cleared,reason 8 (Transport failure)
I
installed the oh323 channel driver and registered to the gate keeper
succesfully.
I come
through the GK, ring the dialed number forabout 0.5 seconds
andloose the line.I contacted the GKand they report that they
receive the correct dialstring to route the call but the call is ended by my
side.
The
dialstring looks like this:
exten
= _.,1,Dial(OH323/${EXTEN},60,r)
I use the
following channel driver:
asterisk-oh323-0.7.1
openh323-Janus_patch4-src
pwlib-Janus_patch4-src
and the
message on asterisk console looks like this:
--
Registered with gatekeeper '[EMAIL PROTECTED]'.
--
Executing Dial(SIP/2000-c9fc, OH323/0012029361212|60|r)
in new stack
-- H.323
call to 0012029361212 with codec(s) g729
-- Called
0012029361212
-- H.323
call 'ip$localhost/2209' cleared, reason 8 (Transport failure)
--
OH323/L2209 is circuit-busy
-- Hungup
'OH323/L2209'
==
Everyone is busy/congested at this time (1:0/1/0)
== Auto
fallthrough, channel 'SIP/2000-c9fc' status is 'CONGESTION'
--
Executing Dial(SIP/2000-c9fc, OH323/h|60|r) in new
stack
-- H.323
call to h with codec(s) g729
-- Called
h
-- Hungup
'OH323/L2210'
== Spawn
extension (local, h, 1) exited non-zero on 'SIP/2000-c9fc'
-- H.323
call 'ip$localhost/2210' cleared, reason 1 (Cleared by local user)
My oh323
configuration:
Configuration
of OpenH323 channel driver
--
Version: 0.7.1
Listening on address: 0.0.0.0:1720
Gatekeeper used: [EMAIL PROTECTED]
(Registered)
FastStart/H245Tunnelling/H245inSetup: OFF/ON/ON
Supported formats in pref. order: g7290
Jitter buffer limits (min/max): 20-100 ms
TCP port range: 5000 - 31000
UDP (RAS) port range: 5000 - 31000
UDP (RTP) port range: 1 - 2
IP Type-of-Service value: 0
User input mode: 2
Max number of inbound H.323 calls: 10
Max number of outbound H.323 calls: 10
Max number of simultaneous H.323 calls: 20
Max call rate (ingress direction): 1.00/30
What might be the problem?
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Re: [Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)
Cenk Yabas wrote: Thanks to Yves's commitment I was able to configure oh323 channel, cleared the codec issue, registered to Gatekeeper, placed a call, but receive this message on the console. What might be the problem? Asterisk Ready. *CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'. -- Executing Dial(SIP/2000-5a52, OH323/193.192.100.92/0212441) in new stack -- H.323 call to 193.192.100.92/0212441 with codec(s) g729 -- Called 193.192.100.92/0212441 -- H.323 call 'ip$localhost/5502' cleared, reason 15 (Call ended due to security checks) The gatekeeper has cleared the call. I guess because a password is required or the one provided is not correct. What version of the channel driver do you use? Michael. ___ 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
Re: [Asterisk-Users] H.323
On Tue, 2005-02-01 at 14:09 +0900, Kuniyoshi Murata wrote: Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and Open H.323 for Asterisk ? I can't tell you the exact differences, but oh323 seems more stable to me, and more actively maintained also. It looks like the native h323 support doesn't have such a high priority amongst * developers. I've been using oh323 for maybe two months now in my home setup, and it works OK. With native h323 I even had troubles get it compiling. Let me point out though that neither h323 nor oh323 support video. The only channel driver I know about which supports video is SIP, and I can't comment on the quality since I didn't test that myself. But if you need video with h323, asterisk won't work for you. Regards, Bruno. ___ 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
Re: [Asterisk-Users] h.323 Type=User
Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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
RE: [Asterisk-Users] h.323 Type=User
Thanks !! I will try!! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Soren Rathje Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h.323 Type=User Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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
RE: [Asterisk-Users] h.323 Type=User
Now it is working... Thanks! -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Soren Rathje Enviado el: Martes, 21 de Diciembre de 2004 02:30 p.m. Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h.323 Type=User Sebastian Nocetti wrote: is h323 per user based working??? I have setup this: [User1] type=user host=xx.xx.xx.xx context=international incominglimit=30 But all calls from xx.xx.xx.xx are not routed to context international, it is working? I am using chan_h323 I'm using current CVS 21-12-04 and found that my Siemens OptiPoint 300a w/ h323 version 2.5.32 (and no GK) have some issues with openh323, it sends the h323-id but chan_h323 is not able to attach a context to it except for the default context... I found that by adding userbyalias = no to h323.conf it now associate device/context by IP address and not Name... It works for me but it is apparent to me that the h323 stack in the phone is pure crap.. :-) /Soren ___ 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 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 --- Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.803 / Virus Database: 546 - Release Date: 2004-11-30 ___ 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
Re: [Asterisk-Users] H.323 trunking
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323 trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] H.323 trunking
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with Q.931 cause)
-- Hungup 'OH323/L11490'
== No one is available to answer at this time
Dec 7 16:48:25 WARNING[1687569]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
what is the meaning of *reason 24*. Is there a problem
with my codec?
thx in advance...
ps: 61 is a prefix. All the extensions 61xxx
should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] H.323 trunking
hi michael,
thx for the answer, but now i have the following
error:
Executing Dial(SIP/2004-b1cf,
OH323/192.168.204.130) in new stack
-- H.323 call to 192.168.204.130 with codec ALAW
-- Called 192.168.204.130
-- H.323 call 'ip$localhost/11490' cleared, reason
24 (Call ended with Q.931 cause)
-- Hungup 'OH323/L11490'
== No one is available to answer at this time
Dec 7 16:48:25 WARNING[1687569]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
what is the meaning of *reason 24*. Is there a problem
with my codec?
thx in advance...
--- Michael Manousos [EMAIL PROTECTED]
wrote:
See below.
Nardis Dome wrote:
Hi,
Could someone help me on configuring a H.323
trunk.
I am trying to set up the following scenario:
[SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)]
I am using the following versions:
Linux CentOS 3.3/2.4.21-.EL.co
asterisk 1.0.1
pwlib_1.5.2
openh323_1.12.2
asterisk-oh323-0.6.3b
Calling from Asterisk (2004) to the H.323phone
(61-8004) gives me the following error
-- Executing Dial(SIP/2004-8350,
H323/192.168.204.130) in new stack
Dec 7 13:45:19 WARNING[1032209]: channel.c:1901
ast_request: No channel type registered for 'H323'
Dec 7 13:45:19 NOTICE[1032209]: app_dial.c:742
dial_exec: Unable to create channel of type 'H323'
== Everyone is busy/congested at this time
Dec 7 13:45:29 WARNING[1032209]: pbx.c:1933
ast_pbx_run: Timeout, but no rule 't' in context
'default'
[general]
static=yes
writeprotect=no
;Trunk=Modem/g1
[default]
exten = 2004,1,NoOp( call for ${EXTEN})
exten = 2004,2,Dial(SIP/${EXTEN},10,tr)
exten = 2004,3,Congestion
exten = 2005,1,NoOp( call for ${EXTEN})
exten = 2005,2,Dial(SIP/${EXTEN},10,tr)
exten = 2005,3,Congestion
exten = _61,1,Dial,H323/192.168.204.130
Change this into:
exten = _61,1,Dial,OH323/192.168.204.130
ps: 61 is a prefix. All the extensions 61xxx
should be
routed to the H.323 trunk.
thx for your feedback
Michael.
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Re: [Asterisk-Users] h.323 debian sarge problem - Could not open sound channel
Jeremy McNamara wrote: Mészáros Mihály wrote: Please if you can please help me to solve this problem. Help yourself and READ THE README. Hello Jeremy! I read it already! ;-) thx! But i didn't find a word about that chan-h323 what decoder encoder use. It use the libopenh323 or other in built encoder ? I have problem (as you can see in my trace) in opening libopenh323 encoder. My question was can i override function OpenAudioChannel original is in libopenh323 - h323ep.cxx function in chan-h323 MyH323EndPoint can i ignore opening sound device ? Or chan-h323 use it somehow? Regards, Misi Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 debian sarge problem - Could not open sound channel
Mészáros Mihály wrote: Please if you can please help me to solve this problem. Help yourself and READ THE README. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 call problemm (no sound)
[EMAIL PROTECTED] wrote: Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs and consile output: we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 call problemm (no sound)
I am having the same problem and not using the NuFone h323 but the Asterisk-OH323... Inbound to sip from h323 seems to be a problem with audio... Robert A. Huddleston, KF4BYY IT Support Analyst Cavalier Telephone LLC. (Cell) 804.400.3686 [EMAIL PROTECTED] -Original Message- From: Senad Jordanovic [mailto:[EMAIL PROTECTED] Sent: Monday, September 20, 2004 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] H.323 call problemm (no sound) [EMAIL PROTECTED] wrote: Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs and consile output: we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 call problemm (no sound)
Could You please tell me which exactly version of H.323 (or source files date or so) I need for latest cvs I tried last versions before march (cvs checkout -D 2004-03-01) and before april (cvs checkout -D 2004-04-01), replace files in channels/h323 directory of last CVS with files from theese versions. H323 compiles but asterisk gives an error: chan_h323.c: In function `load_module': chan_h323.c:1975: warning: passing arg 7 of `h323_callback_register' from incompatible pointer type chan_h323.c:1975: error: too many arguments to function `h323_callback_register' make[1]: *** [chan_h323.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Monday, September 20, 2004 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] H.323 call problemm (no sound) [EMAIL PROTECTED] wrote: Hi all, I'm having trouble with H.323 outbound calls, * connects but there is no sound in both ways. I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729 licenses installed and this is onli one call. I tested my * with another ITSP over SIP and G.729 codec and there was all ok Here is my configs and consile output: we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Jeremy McNamara wrote: Michael Manousos wrote: The performance of the oh323 channel driver is limited by OpenH323. asterisk-oh323 uses the (more complete) RTP implementation offered by the library, and not that of Asterisk. Of course there are pros (adaptive jitter buffer, RTCP implementation) and cons (lower performance). It's up to the user to select the one that performs better for his application. flamePut the crack pipe down./flame I won't bite. We all know what you have done. We have gone over this before, asterisk-oh323 is limited by the method you implemented to buffer the audio around. Jeremy McNamara Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Scott Stingel a écrit : Hi- In answer to your questions: Someone on Friday had said that disabling Fast Start corrected the audio problem with H.323, so yesterday I tried to disable it in ~/asterisk/channels/h323/ast_h323.cpp. Today, I noticed that Jeremy (NuFone) uploaded a new version of this file with the same fix: Change the line: BOOL noFastStart; To: BOOL noFastStart = TRUE; Unfortunately, this made no difference for connections from my customer's Cisco 5300, so I decided to abandon the built-in h323 in favour of oh323. Maybe you'll have better luck with the original code. I updated the to the cvs-27/06/04, applied the changes above and it works. I'm not using any cisco devices but the GNUgk [...] -- dash ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Tommy, Still waiting from you whether the CDRs are recorded with cdr_csv. This is working just fine for me. Michael. T. Chan wrote: Hi, Scott. Are you telling me that this native h.323 has been hardcoded with fast start? Can you tell me where in ast_h323.cpp it is that you disabled this faststart? Have you tried using the Stable cvs of the Asterisk. Can you let me know which version of the OH323 are you using ? Is it the 0.6.2A? Which version of the Pwlib and OpenH323 you used, is it the newest version as stated? Did you apply the patch? I tried using this driver, but I have problem with cdr_mysql, it is not recording cdr. Please share your information, thanks alot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Sunday, June 27, 2004 6:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Audio problem UPDATE Update on this problem: I gave up on the native h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to edit the Makefile as stated. Some load testing to following this week, but I'm encouraged! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get audio
working. (yes, I tried disabling FastStart in ast_h323.cpp - no change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323, as
stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi Brian-
I think you have to use 0.6.2a not 0.6.2. Also, you might try the new
version from today: 0.6.3.
And just checking, in your Makefile, that you set ASTERISKSRCDIR =
/usr/src/asterisk. (maybe this is a 0.6.2a thing)
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Wilkins
Sent: Monday, June 28, 2004 8:15 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad
to say that everything seems to work so far. Not only does audio
work, but even the handshaking is now working in both OpenPhone and
even NetMeeting (for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and
OpenH323, as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you were
able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with asterisk-oh323
0.6.2. I get the following errors when trying to compile the oh323 wrapper
for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get audio
working. (yes, I tried disabling FastStart in ast_h323.cpp - no change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323, as
stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Michael:
Yes I did.
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323,
as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Ok,
I got it all to work finally. I removed everything and started from
scratch. I also got the latest version of asterisk from the CVS. I built
PWLib, then applied the patch to oh323 1.13.5 then built oh323, and finally
built and installed the wrapper (0.6.3). I just started up Asterisk and
everything is working fine. Thanks for all the help -
On Yaum al-Ithnain 10 Jumaada al-Awal 1425 11:28 am, Michael Manousos wrote:
Did you apply to the OpenH323 the included patch BEFORE configuring the
library (openH323)?
Also, try to use the latest version (0.6.3) if you are running current
Asterisk CVS code.
Michael.
Brian Wilkins wrote:
Sorry this has nothing to do with your audio issue, but I noticed you
were able to build oh323 1.13.5 and pwlib 1.6.6 sucessfully with
asterisk-oh323 0.6.2. I get the following errors when trying to compile
the oh323 wrapper for asterisk:
-- snippet of errors --
In file included from asteriskaudio.cxx:37:
wrapper_misc.hxx:61: parse error before `{'
wrapper_misc.hxx:71: parse error before `protected'
In file included from asteriskaudio.cxx:38:
asteriskaudio.hxx:41: parse error before `{'
asteriskaudio.hxx:48: destructors must be member functions
asteriskaudio.hxx:55: parse error before `protected'
asteriskaudio.hxx:57: syntax error before `;'
asteriskaudio.hxx:61: parse error before `}'
asteriskaudio.hxx:69: parse error before `{'
asteriskaudio.hxx:76: destructors must be member functions
asteriskaudio.hxx:78: syntax error before `('
asteriskaudio.hxx:79: syntax error before `('
asteriskaudio.hxx:80: parse error before `'
--end snippet--
In my makefile, I have set the following settings :
PWLIBDIR=/usr/src/pwlib
OPENH323DIR=/usr/src/openh323
ASTERISKINCDIR=/usr/src/asterisk/include
ASTERISKMODDIR=/usr/lib/asterisk/modules
OH323WRAPLIBDIR=/usr/local/lib
Both pwlib and openh323 build sucessfully, but when I try to build
asterisk-oh323 I get those errors. Any clues?
Regards,
Brian Wilkins
--
Heritage Communications Corporation
Melbourne, FL USA 32935
On Yaum al-Ahad 09 Jumaada al-Awal 1425 06:20 pm, Scott Stingel wrote:
Update on this problem:
I gave up on the native h.323 because, like others, I couldn't get
audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no
change)
So I went and got the OH323 code from www.inaccessnetworks.com. Glad to
say that everything seems to work so far. Not only does audio work, but
even the handshaking is now working in both OpenPhone and even NetMeeting
(for the first time).
Notes to others who want to try OH323:
* The installation is a bit more complicated than h323. Follow the
instructions in the ReadMe file exactly.
* You must choose and install the proper versions of PWLib and OpenH323,
as stated.
* Don't forget to edit the Makefile as stated.
Some load testing to following this week, but I'm encouraged!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
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--
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http://www.hcc.net
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Re: [Asterisk-Users] H.323 Audio problem UPDATE
Michael Manousos wrote: The performance of the oh323 channel driver is limited by OpenH323. asterisk-oh323 uses the (more complete) RTP implementation offered by the library, and not that of Asterisk. Of course there are pros (adaptive jitter buffer, RTCP implementation) and cons (lower performance). It's up to the user to select the one that performs better for his application. flamePut the crack pipe down./flame We have gone over this before, asterisk-oh323 is limited by the method you implemented to buffer the audio around. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi, Scott. Are you telling me that this native h.323 has been hardcoded with fast start? Can you tell me where in ast_h323.cpp it is that you disabled this faststart? Have you tried using the Stable cvs of the Asterisk. Can you let me know which version of the OH323 are you using ? Is it the 0.6.2A? Which version of the Pwlib and OpenH323 you used, is it the newest version as stated? Did you apply the patch? I tried using this driver, but I have problem with cdr_mysql, it is not recording cdr. Please share your information, thanks alot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Sunday, June 27, 2004 6:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Audio problem UPDATE Update on this problem: I gave up on the native h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to edit the Makefile as stated. Some load testing to following this week, but I'm encouraged! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi- In answer to your questions: Someone on Friday had said that disabling Fast Start corrected the audio problem with H.323, so yesterday I tried to disable it in ~/asterisk/channels/h323/ast_h323.cpp. Today, I noticed that Jeremy (NuFone) uploaded a new version of this file with the same fix: Change the line: BOOLnoFastStart; To: BOOLnoFastStart = TRUE; Unfortunately, this made no difference for connections from my customer's Cisco 5300, so I decided to abandon the built-in h323 in favour of oh323. Maybe you'll have better luck with the original code. I don't have the option of going back to the so-called STABLE version, since there are now many fixes and updates in the current HEAD that effect me. Anyway, so far I'm happy with Michael Manousos' oh323 drivers. Here's the full link: http://www.inaccessnetworks.com/projects/asterisk-oh323 Suggest that you try it and see if it works for you, if you have any problems with the built-in h323 code. The web site provides a lot of info about versions and what is supported. I'm running the latest oh323 0.6.2a, which requires PWLib 1.6.6 and OpenH323 1.13.5. Don't use any other versions of these - might not work Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Chan Sent: Sunday, June 27, 2004 4:47 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H.323 Audio problem UPDATE Hi, Scott. Are you telling me that this native h.323 has been hardcoded with fast start? Can you tell me where in ast_h323.cpp it is that you disabled this faststart? Have you tried using the Stable cvs of the Asterisk. Can you let me know which version of the OH323 are you using ? Is it the 0.6.2A? Which version of the Pwlib and OpenH323 you used, is it the newest version as stated? Did you apply the patch? I tried using this driver, but I have problem with cdr_mysql, it is not recording cdr. Please share your information, thanks alot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel Sent: Sunday, June 27, 2004 6:20 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Audio problem UPDATE Update on this problem: I gave up on the native h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even NetMeeting (for the first time). Notes to others who want to try OH323: * The installation is a bit more complicated than h323. Follow the instructions in the ReadMe file exactly. * You must choose and install the proper versions of PWLib and OpenH323, as stated. * Don't forget to edit the Makefile as stated. Some load testing to following this week, but I'm encouraged! Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Audio problem UPDATE
Scott Stingel wrote: Some load testing to following this week, but I'm encouraged! This is where you are going to be discouraged with that other H.323 driver. I guarantee it. Disabling fast-start has solved the problems for quite a few other ppl using 5300s, so you must be doing something really nasty with them to still not work. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Audio problem UPDATE
Hi, Scott, I am very interested in knowing the result of your loading test, please share after you have done it. Are you using Asterisk as a pass-through (kinda softswitch) or do have have digium hardware and use it as an endpoint, because I believe the maxiumum number of channels you can run stably could be different, please share, thanks. TC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jeremy McNamara Sent: Sunday, June 27, 2004 9:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] H.323 Audio problem UPDATE Scott Stingel wrote: Some load testing to following this week, but I'm encouraged! This is where you are going to be discouraged with that other H.323 driver. I guarantee it. Disabling fast-start has solved the problems for quite a few other ppl using 5300s, so you must be doing something really nasty with them to still not work. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.707 / Virus Database: 463 - Release Date: 6/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile
RE: [Asterisk-Users] H.323 - NO AUDIO IN BOTH DIRECTIONS
I have the same problem here. I have to servers working with identical (same) configurations, the old one is working just perfect and the new one I got, is not working (no voice in both directions). Im trying to fix this problem with digium, we are exchanging emails so if I get a solution Im gona reply it here. Best Regards Hekuran Doli FYI I am experiencing the same problem. I have complied asterisk from the latest CVS The call connects with no audio or DTMF to either end. I tested with ulaw and g729 with no success. -Michael On Fri, 2004-06-25 at 10:55, Scott Stingel wrote: Just checking that you have installed the proper versions of both OpenH323 and PWLib, as mentioned in ~/asterisk/channels/h232/README, and have rebuilt asterisk after those installations as specified? If so, then you are having the same problem I'm experiencing: no audio on H.323. I'm also connecting through a Cisco 5300. I'm just generating audio in one direction: outbound from asterisk - I hear nothing. This used to work I'm pretty sure... There is an outstanding bug report covering H.323 problems (#1334), not sure what the current status is. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com http://www.evtmedia.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Friday, June 25, 2004 7:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NO AUDIO IN BOTH DIRECTIONS hello all, I am having a trouble with Audio using h.323 channel... I am doing this Call comes into cisco 5300 and is sent to Asterisk, asterisk catch call with h.323 driver and send call to a SoftSwitch that routes the call, I can see log debug telling me, CALLED XXX, and then RINGING, and I can hear ring tones... but when call is answered, I DONT HEAR ANYTHING... I am using lastest ASTERISK download somebody can help me to solve this problem thanks..!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Michael K. Rodriguez Dialmex LLC Director of Network Operations 200 S. 10th Suite 1209 McAllen, TX 78501 (956) 994-0014 x107 office (956) 682-8521 fax (956) 239-0627 mobile ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and cause code 'user busy'
This is a known architectural issue that does not appear to have been resolved yet. See bug http://bugs.digium.com/bug_view_page.php?bug_id=0001337 The problem is the mapping of the various internal states of different Asterisk channels on to the Q.931 states. Asterisk currently does not wait for 'alerting' to be confirmed on the outgoing channel before 'ALERTING' is sent out on the ISDN line. This means that if the outgoing channel turns out to be 'busy', it is already too late to reject the incoming call with a 'SUBSCRIBER BUSY' cause as this is not valid in the ALERTING phase. Our switches here in the UK retuern 'ringback tone' to the calling party as soon as 'ALERTING' is received, and if it is then rejected with a 'busy' clearing cause you get a message 'The other party has hung up'. I spoke to Markster at length about this but I am not sure I really made the issue clear. Anyone else got any comments on this issue? I think it is a fairly major issue... Rgds Tim Jan Baumann wrote: Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried different Apps there (Hangup, Busy, Congestion) They deliver different cause codes to the H.323/ISDN side (normal call clearing or call rejected) but none of them returns 'user busy' as expected. In Zaptel with Q.931 PRI (euroisdn) you can do exten = 123,102,SetVar(PRI_CAUSE=17) exten = 123,103,Hangup to explicitely set the RELEASE cause code. Is something similiar also possible with H.323? Thank you and regards, Jan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Seg faulting
Ive placed a bounty on my bug. See http://bugs.digium.com/bug_view_page.php?bug_id=0001334 From: Derek Samford Sent: Wednesday, April 07, 2004 4:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323 Seg faulting Can someone take a look, tell me if this is a bug, a possible resources issue, or my own damn fault? http://bugs.digium.com/bug_view_page.php?bug_id=0001381 Thanks, Derek
Re: [Asterisk-Users] H.323 ASN.1 Vulnerabilities: Request for official patch!
See the existing discussion on this Ditto. IT DOES NOT WORK. Compiles, but no calls go through. I asked you to post your exact versions of all components, but I don't believe you did this. I have not been able to get it to work with Asterisk 0.7.2. Just because *YOU* got it to work on your particular system does not mean the problem is solved. If there is a way to get it to work reliably: 1. Please post complete details 2. Someone update asterisk.org with correct information. I believe it is correct that there is no official response on this from Asterisk to what many people consider a critcal security issue. Read the archives is nice, but really, the default Asterisk should be fixed. And the fix needs to be tested on a variety of systems, too. I tried your exact version of pwlib, and have not been able to get a *SINGLE* call to work. See the existing discussion on this Ahem. I posted pretty thorough details on what wasn't working ... Please respond so that the discussion can -- uh -- exist ... -T.i.A., Jim [Apologies for bandwidth-wasting inclusion below -- I'm reposting since someone thinks this discussion has been settled ...] On Thu, Feb 26, 2004 at 09:18:45AM +0800, [EMAIL PROTECTED] wrote: In the Makefile inside asterisk/channels/h323 directory, there's a line like this: CFLAGS += -I$(PWLIBDIR)/include/ptlib/unix -I$(PWLIBDIR)/include try to use -I$(PWLIBDIR)/include ONLY, it should work. I've compiled it with pwlib 1_6_2, which works fine leo Sigh. I am having a very rough time here. Could you please post exactly which versions of Asterisk and OpenH323 you used? When I use your advice above I get a successful build, but I haven't got a single call to actually *work* through H.323. Here are my results (all trials are Asterisk 0.7.2): OpenH323 1.13.0 / Pwlib 1.6.0: Asterisk segfaults when it gets an H.323 call. OpenH323 1.13.2 / Pwlib 1.6.3: Channel won't load, there's an unresolved symbol. OpenH323 1.13.2 / Pwlib 1.6.2: Asterisk appears to be fully stable. As far as Asterisk is concerned, everything works: calls are made, answered, bridged, all looks fine from the console. But nothing is actually making it *back* through H.323 from the Asterisk end. When I call Asterisk through H.323, Asterisk thinks things are fine, but from the calling end it thinks no one answered. When I call from the Asterisk end, I never hear anything that sounds like an answer. Now this looks *VERY* familiar. It sure is like the H.323 problems I had right at first until I caught on to using *only* G.711 A-law. Once I started making sure everyone was on ALAW, H.323 starting working fine (except for DTMF, but that's a subject for a new thread ...) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Phone w/ Asteisk
On Saturday 06 December 2003 19:01, Greg Boehnlein wrote: Hello, I have a friend that is asking if he can use his Ericsson 3413 H.323 IP phone with Asterisk. I can't seem to find any reference to this phone on the Wiki... you can either use chan_h323 or chan_oh323 (the latter is contrib stuff - google it up). Both needs pwlib and openh323. RTFM first and if it doesn't work, send an email to the list or ask on IRC. I know people are running h.323 in production (or so I've heard), but as (AFAIK) there still are some unsolved issues, YMMW. roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
LOOKING FOR SPECIFIC INFORMAITON ( was Re: [Asterisk-Users] H.323 Phone w/ Asteisk)
Roy Sigurd Karlsbakk wrote: I know people are running h.323 in production (or so I've heard), but as (AFAIK) there still are some unsolved issues, YMMW. Why not list out the specific problems and they can be addressed if they are still a problem? You are quick to bash my channel driver but you never seem to ever offer any SPECIFIC information. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: LOOKING FOR SPECIFIC INFORMAITON ( was Re: [Asterisk-Users] H.323 Phone w/ Asteisk)
On Sat, 6 Dec 2003, Jeremy McNamara wrote: Roy Sigurd Karlsbakk wrote: I know people are running h.323 in production (or so I've heard), but as (AFAIK) there still are some unsolved issues, YMMW. Why not list out the specific problems and they can be addressed if they are still a problem? You are quick to bash my channel driver but you never seem to ever offer any SPECIFIC information. Jeremy, I just got the NuFone H323 channel driver compiled in and will be doing some testing with Erricson H323 phones. I'll be happy to report back to you on my progress. Since I've never played with H323 anything, I may need some pointers on what type of information you will need! ;) -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE: Follow the instructions on line below and do NOT issue a make clean install in asterisk/channels/h323 as indicated elsewhere, just issue a make and then in /usr/src/asterisk (or wherever you source is), issue a make install and this will work. To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. On Friday, November 7, 2003, at 02:51 AM, Anton L. Kapela wrote: Jeremy McNamara said: 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. You haven't read the README And I quote: To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. I suspect that he indeed did read the README. In fact, I just (for kicks) cvs checked out a fresh copy of libpri, zaptel, and asterisk this evening. I checked my h.323 channel README, and I see: Example commands to make sure everything gets cleaned and then rebult in proper order: cd /path/to/pwlib make clean opt cd /path/to/openh323 make clean opt cd asterisk/channels/h323 make clean install For some reason, doing as instructed on various sites (digium.com, for one) you'll be pulling stale CVS code. Or, for some reason, your updates to what's in CVS aren't actually working. I wish I understood CVS more to better research this. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Look again, this time with the cvs code. Jeremy McNamara Paul Cheng wrote: THERE IS AN INCONSISTENCY IN THE README FILE THAT IS OUT OF DATE: Follow the instructions on line below and do NOT issue a make clean install in asterisk/channels/h323 as indicated elsewhere, just issue a make and then in /usr/src/asterisk (or wherever you source is), issue a make install and this will work. To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. On Friday, November 7, 2003, at 02:51 AM, Anton L. Kapela wrote: Jeremy McNamara said: 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. You haven't read the README And I quote: To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. I suspect that he indeed did read the README. In fact, I just (for kicks) cvs checked out a fresh copy of libpri, zaptel, and asterisk this evening. I checked my h.323 channel README, and I see: Example commands to make sure everything gets cleaned and then rebult in proper order: cd /path/to/pwlib make clean opt cd /path/to/openh323 make clean opt cd asterisk/channels/h323 make clean install For some reason, doing as instructed on various sites (digium.com, for one) you'll be pulling stale CVS code. Or, for some reason, your updates to what's in CVS aren't actually working. I wish I understood CVS more to better research this. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 and G729: Another sad tale
Dear all, I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile on asterisk/channels/h323. I also donwload pwlib and openh323 from nufone.net/downloads, and did following things: 1. /pwlib, make clean, make both 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. any one can help on this. Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 and G729: Another sad tale
Hi all, The errror message is as follows: [EMAIL PROTECTED] h323]# make install install -m 755 chan_h323.so /usr/lib/asterisk/modules install: cannot stat `chan_h323.so': No such file or directory make: *** [install] Error 1 Please advise if you could. Regards, George Lin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of G Lin Sent: Thursday, November 06, 2003 8:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H.323 and G729: Another sad tale Dear all, I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile on asterisk/channels/h323. I also donwload pwlib and openh323 from nufone.net/downloads, and did following things: 1. /pwlib, make clean, make both 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. any one can help on this. Thanks, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
G Lin wrote: Dear all, I just fresh CVS the asterisk code, and uncomment the G729 in the Makefile on asterisk/channels/h323. I also donwload pwlib and openh323 from nufone.net/downloads, and did following things: 1. /pwlib, make clean, make both make opt Anything else just wastes massive time. 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. You haven't read the README And I quote: To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Jeremy McNamara said: 2. /openh323, make clean, make opt 3. /asteriks/channels/h323, make clean, make install, and it is got error about no chan_h323.o exists. and the make install is failed. You haven't read the README And I quote: To compile this code: Issue a make in the asterisk/channels/h323 directory, then go back to the Asterisk source top level directory and issue a make install. I suspect that he indeed did read the README. In fact, I just (for kicks) cvs checked out a fresh copy of libpri, zaptel, and asterisk this evening. I checked my h.323 channel README, and I see: Example commands to make sure everything gets cleaned and then rebult in proper order: cd /path/to/pwlib make clean opt cd /path/to/openh323 make clean opt cd asterisk/channels/h323 make clean install For some reason, doing as instructed on various sites (digium.com, for one) you'll be pulling stale CVS code. Or, for some reason, your updates to what's in CVS aren't actually working. I wish I understood CVS more to better research this. --Tk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. I believe that when you use up2date on both RH8 and RH9, you end up with the same version of Kernel. So how do you differentiate RH8 and RH9 in terms of this issue? Or do you not use up2date to get the latest kernel and source? On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
I can also confirm chan_h323 and g.729 work well to 5300s, but we had to build on RH8 not RH9. Haven't tried 5300 to Asterisk except via SIP which works fine--even to i4l interfaces. On Friday, October 31, 2003, at 01:57 AM, Jeremy McNamara wrote: John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? Yes, g729r8 If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. Others seem to have massive issues with chan_h323 and G.729, but i've dealt a dozen or so 5300s of which I haven't had any trouble whatsoever, with nothing other than the code that is currently in the cvs. However, I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. If Asterisk is going to be encoding G.729, yes you will need licenses from Digium. Jeremy McNamara P.S. I'm biased and cannot comment about that other driver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
John Todd wrote: I've done some reviewing of the archives for G729 and H323 experiences. The landscape of that query isn't pretty - lots of pleas for help, and nor do I see too many answers. I have a pending bid that requires some data before I can implement * on this particular solution. My question is perhaps a slightly differently worded one than has been asked before, but it may be the case that it is the same question as I have seen already posted (with no 'definitive' answer): Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? Yes, g729r8 If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. Others seem to have massive issues with chan_h323 and G.729, but i've dealt a dozen or so 5300s of which I haven't had any trouble whatsoever, with nothing other than the code that is currently in the cvs. However, I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. If Asterisk is going to be encoding G.729, yes you will need licenses from Digium. Jeremy McNamara P.S. I'm biased and cannot comment about that other driver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Quoting Jeremy McNamara [EMAIL PROTECTED]: Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? IMHO as for today No, For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300. Calls were dropped from cisco side after two udp packets from cisco sent. I have only terminated calls from Asterisk to the 5300, never from the 5300 to Asterisk. Outgoing calls from * to cisco with g729 from digium works fine. But I didnt test it with large volume. regards izo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? IMHO as for today No, For incomig I couldnt even get it working with g711 and ciscos 72xx and as5300. Calls were dropped from cisco side after two udp packets from cisco sent. I've had incoming working with G711, i can't recall if I had it working with G729. I found changing the payload to 20 frames fixed it, although Jeremy tells me I'm crazy and it works without doing that. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 and G729: Another sad tale
Hello, Can g729 calls of type g729r8 or g729br8 from a Cisco AS5300 be terminated on Asterisk systems and sent out Zap interfaces? A while ago, I only manage to get g729 call works when terminating in Cisco AS5300 from Asterisk but was unable to terminate call in Asterisk from Cisco AS53000 using g729. If the answer is Yes, then are there any specific patches I will need? Which of the two H323 drivers works? Both? Of course, I assume that the G729 licenses from Digium are required for each active channel. not sure about patches, however if you plan to use chan_h323, it is best to get the CORRECT versions of pwlib and openh323 and follow the exact installation instructions. One important thing about these libraries with chan_h323 is DO NOT 'make install' pwlib and openh323 hth ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 - SIP gateway
Olaf Menzel wrote:
But I want to transmit the original callerid as defined in sip.conf
via the H.323 gatekeeper to a H.322 phone. How to manage this ??
How about Dial,H323/[EMAIL PROTECTED]/${CALLERIDNUM} ?
However you will have issues with the gatekeeper if it is expecting a
specific H.323 ID from your endpoint.
Jeremy McNamara
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Re: [Asterisk-Users] H.323 - SIP gateway
It works with a third party gatekeeper, which is good enough (get asterisk to register with a gatekeeper). There's 20 lines in asterisk where Jeremy started making the gatekeeper functionally, currently rapped with #if 0 #endif I doubt it will ever exist, for the short term anyway. - Original Message - From: Roy Sigurd Karlsbakk [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Wednesday, October 15, 2003 6:22 PM Subject: Re: [Asterisk-Users] H.323 - SIP gateway You shouldn't treat asterisk as a gatekeeper (because it ain't) On your H.323 equipment, set asterisk up as a gateway. ok? I've repeatedly heard Jeremy brag about chan_h323 working as a gatekeeper, although I've never understood how to set it up like this roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 - SIP gateway
I am trying to configure * to route calls from SIP extension to an externeal H.323 gatekeeper and vice versa. The route from * to the gatekeeper is a simple ENUM call and work fine: [outbound][outbound] exten = _3XXX,1,Dial,H323/[EMAIL PROTECTED] One Snom100 phone is defined in sip.conf: [snom] type=friend host=dynamic dtmfmode=rfc2833 mailbox=1000 context=local callerid=Operator Office 1000 in extensions.conf it is defined as extension 1000: [local] include = voicemail include = outbound include = inbound ; SIP Phone Operator Office exten = 1000,1,Dial,SIP/snom|30|tr exten = 1000,2,Voicemail,u1000 exten = 1000,102,Voicemail,b1000 When I call a H.323 telephone behind the gatekeeper this telephone shows a callerid root as name and the Asterisk IP address without the original extension 1000. I can define an alias in h323.conf but every call to the gatekeeper will have this callerid: [olaf-snom] type=h323 e164=1000 context=local But I want to transmit the original callerid as defined in sip.conf via the H.323 gatekeeper to a H.322 phone. How to manage this ?? -- I have defined a inbound gatekeeper in h323.conf: [Corponet] type=user host=10.3.1.100 context=inbound incominglimit=4 What else has to be in the extensions.conf if a H.323 phone want to call me at: [EMAIL PROTECTED] ?? regards Olaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 - SIP gateway
Hi! I have done that but it doesn't work because I need also the port 1720 to make the comunication. Port 1719 is only used to the RAS messages and 1720 is used to make the communication. Thanks a lot for your help Regards, Mireia Quoting CW_ASN - Gus [EMAIL PROTECTED]: Or, if you must use 1719, try to change h323.conf: [general] port = 1719 bindaddr = 0.0.0.0 Regards, Gus - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:23 PM Subject: Re: [Asterisk-Users] H.323 - SIP gateway h323 runs on port 1720. Your gatekeeper is trying to contact the wrong port number. On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote: Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 - SIP gateway
You shouldn't treat asterisk as a gatekeeper (because it ain't) On your H.323 equipment, set asterisk up as a gateway. Hi! I have done that but it doesn't work because I need also the port 1720 to make the comunication. Port 1719 is only used to the RAS messages and 1720 is used to make the communication. Thanks a lot for your help Regards, Mireia Quoting CW_ASN - Gus [EMAIL PROTECTED]: Or, if you must use 1719, try to change h323.conf: [general] port = 1719 bindaddr = 0.0.0.0 Regards, Gus - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, October 14, 2003 12:23 PM Subject: Re: [Asterisk-Users] H.323 - SIP gateway h323 runs on port 1720. Your gatekeeper is trying to contact the wrong port number. On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote: Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323-SIP Gateway
There is no need to use oh323. If you look in /usr/src/asterisk/channels/h323 then you will find that there is already an h323 implemenatation present (chan_h323). You just need to follow the instructions and it works great. Tan telappliant.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bryan Nolen Sent: 03 October 2003 14:55 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] H.323-SIP Gateway Basically you just need to make sure that the (o)h323 channel is compiled. Personally I use the chan_oh323 driver (google it). Its very easy, just like setting up an normal extentions (see handbook + voip-info.org + google) -Bryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, 3 October 2003 11:47 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] H.323-SIP Gateway Hi everybody! I am trying to do a H.323-SIP Gateway and someone have told me that asterisk would help me. Has this software this functionality? If it has, so what must I do to make that everything works ok? Thanks a lot for your answers! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 04:02, Jeremy McNamara wrote: You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr That doesn't work when you use H323 directly. As in Dial(H323/ip$12.34.56.78|120|r) ... Works fine with OH323 though. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/bq4/2TEAILET3McRAmUkAJ97wvMuj+O5D1E3Uu9PBZ5G4bIt+QCgiFAA Ufbc9xNFKXOExxoXia0Qits= =iRbu -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
I have found that mixing the Dial() format with | can cause problems. Does Dial(H323/ip$12.34.56.78,120,r) work as expected? On Mon, 2003-09-22 at 03:09, Tais M. Hansen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 04:02, Jeremy McNamara wrote: You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr That doesn't work when you use H323 directly. As in Dial(H323/ip$12.34.56.78|120|r) ... Works fine with OH323 though. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/bq4/2TEAILET3McRAmUkAJ97wvMuj+O5D1E3Uu9PBZ5G4bIt+QCgiFAA Ufbc9xNFKXOExxoXia0Qits= =iRbu -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 22 September 2003 10:16, Eric Wieling wrote: I have found that mixing the Dial() format with | can cause problems. Does Dial(H323/ip$12.34.56.78,120,r) work as expected? Doesn't change anything. Here's a better explanation of the problem. Using chan_h323, it doesn't matter which tech I choose to dial. It doesn't make the ringing sound on the h323 endpoint. I.e. h323 ep - chan_h323 asterisk 1 chan_iax2 - chan_iax2 asterisk 2 If 'asterisk 1' has a extension like this: exten = 1234,1,Dial(IAX2/user:[EMAIL PROTECTED]/[EMAIL PROTECTED],30,r) And 'asterisk 2': exten = 1234,1,Wait(10) exten = 1234,2,Answer() exten ... Dialing 1234 on the h323 endpoint would send the call to 'asterisk 2' but during those 10 seconds wait on 'asterisk 2', there's no indication on my h323 endpoint that it's actually ringing. Using chan_oh323 instead of the native h323, the problem magically disappears. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/brqt2TEAILET3McRAkEwAJ9Qa23Gmet470GBhU7NHQm6gXgWsQCfTbO6 mXEHmNtd7xgiQ4B8LrDuANY= =VKte -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h.323 - success
You have to enable ring indications exten = whatever,1,Dial,CAPI/22545070:b98013356|300|Tr Jeremy McNamara Roy Sigurd Karlsbakk wrote: hi seems like things are closing in to something that might look like success. I have one problem left: I don't get ring indicator when I dial out from the h.323 phone... Sound is good, so it doesn't look like a codec problem. I'm using chan_capi with early B3. I also use gnugp to route the calls from the phones to asterisk, as the dlink dph-100h requires this. Debug output follows: Any ideas? roy DEBUG - *CLI exten b4: 98013356 -- Executing Dial(H323/ip$10.47.0.1:39307/29476, CAPI/22545070:b98013356|300|T) in new stack -- Called 22545070:b98013356 us: 0.0.0.0:6124 them: 0.0.0.0:0 info: 0.0.0.0:6124 us: 0.0.0.0:6124 them: 0.0.0.0:0 info: 0.0.0.0:6124 -- CAPI[contr1/22545070]/8 is ringing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Support
chan_h323 is built into asterisk. Check the /usr/src/asterisk/channels/h323 directory for more info. - Original Message - From: Phillip Britt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 02, 2003 1:12 PM Subject: [Asterisk-Users] H.323 Support Hi, I am currently using Asterisk and want to add H.323 support for talking to our gateway routers, which use gnkgk Is the package Asterisk-oh323 the right thing to use, or are there better ways of achieving h.323 support in Asterisk. Thanks, Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 channel problems
Jeremy == Jeremy McNamara [EMAIL PROTECTED]: Jeremy What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing Jeremy newer, nothing older if u want this to work. don't you Jeremy understand? Well, I was trying to find out (politely) about some things. Please allow me to paste from my previous E-mail: 1. Perhaps it's worth trying to report the bugs to distribution maintainers if indeed the distribution-specific installs of openh323 are this buggy? 2. Briefly, do I have a chance of reporting this bug with my versions of libraries, or is chan_h323 completely unsupported if I use anything other than 1.11.7? There was also an implicit question 3. Perhaps the docs haven't been updated and openh323 isn't this problematic anymore? You couldn't have answered question #2 any clearer. Also thanks to Brian West for his informative followup. --J. pgp0.pgp Description: PGP signature
Re: [Asterisk-Users] H.323 channel problems
What part of IN OTHER WORDS: Run Open H.323 v1.11.7, nothing newer, nothing older if u want this to work. don't you understand? Jeremy McNamara Jan Rychter wrote: I have hit a problem where chan_h323 sometimes doesn't hang up properly and stays stuck in the Up state, with asterisk consuming 100% of CPU: *CLI show channels Channel (ContextExtensionPri ) State Appl. Data H323/ip$127.0.0.1:30008/21552 (local 123 1 ) Up (None)(None) 1 active channel(s) *CLI show ch channel channels *CLI show channel H323/ip$127.0.0.1:30008/21552 -- General -- Name: H323/ip$127.0.0.1:30008/21552 Type: H323 UniqueID: 1061946140.22 Caller ID: Jan DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 8 WriteFormat: 1024 ReadFormat: 1024 1st File Descriptor: 26 Frames in: 47575 Frames out: 94850 Time to Hangup: 0 -- PBX -- Context: local Extension: 123 Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (None) Stack: -1 Blocking in: ast_waitfor_nandfds *CLI That's after hanging up (in gnomemeeting) on a H.323 call that is then bridged to IAX2. Now, before I go running to the bugtracker, I'd like to ask some general questions. The H.323 channel readme says: NOTICE: Whatever you do, DO NOT USE distrubution specific installs of Open H.323 and PWLib. In fact you should check to make sure your distro didn't install them for you without your knowledge. Check everything out of CVS. If you dont know how to deal with cvs, learn. Also, if you are not using the listed versions of Open H.323 or PWlib you are on your own, sorry. And: Some chan_h323 users have reported success and others have reported dramatic failures when using newer versions of Open H.323. We haven't personally tested this and will not be able to assist you if you have 'issues'. Sorry. IN OTHER WORDS: Run Open H.323 v1.11.7 nothing newer nothing older if u want this to work. How does this relate to my bug? I'm using openh323-1.12 and pwlib-1.5.0 that I compiled myself. Do they have problems? Does this mean I am on my own? Perhaps it's worth trying to report the bugs to distribution maintainers if indeed the distribution-specific installs of openh323 are this buggy? The requirement of using this particular version of openh323 is a problem for those of us who also use other H.323 software (such as gnomemeeting) which specifically requires newer libraries. Briefly, do I have a chance of reporting this bug with my versions of libraries, or is chan_h323 completely unsupported if I use anything other than 1.11.7? many thanks, --J. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 Gateway Connection
Hi Justin, Try: exten=242,1,Dial(h323/[EMAIL PROTECTED]) Regards, Szymon Czyz Justin Eckhouse [EMAIL PROTECTED] wrote: Hi, I'm trying to setup Asterisk to allow users to dial out to the PSTN using a remote box supporting h.323. I'm using chan_h323.so, and I'm able to make outbound calls to a client like netmeeting with a line like this: exten = 242,1,Dial(h323/xxx.xxx.xxx.xxx) And I'm able to receive incoming calls to asterisk. However I'm not sure how to route calls to the remote h.323 gateway. In my nave state I've tried something like this (xxx is the IP of the h.323 gw): exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE) When I dial 244, nothing happens, this appears in the console: -- Called xxx.xxx.xxx.xxx == No one is available to answer at this time Any pointers in the right direction would be greatly appreciated. Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H.323 Gateway Connection
exten = _91XX,1,Dial(H323/${EXTEN:[EMAIL PROTECTED])
${EXTEN:1} will grab all the digits you sent in 91XX and the :1, in
${EXTEN:1}, tells it to drop the first digit.
Michael
I'm trying to setup Asterisk to allow users to dial out to the PSTN using a
remote box supporting h.323. I'm using chan_h323.so, and I'm able to make
outbound calls to a client like netmeeting with a line like this:
exten = 242,1,Dial(h323/xxx.xxx.xxx.xxx)
And I'm able to receive incoming calls to asterisk. However I'm not sure
how
to route calls to the remote h.323 gateway. In my naïve state I've tried
something like this (xxx is the IP of the h.323 gw):
exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE)
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Re: [Asterisk-Users] H.323 Gateway Connection
Justin Eckhouse wrote: exten = 244,1,Dial(h323/xxx.xxx.xxx.xxx/PSTN-NUMBER-HERE) This is bad... if you use this kind of exten line PSTN-NUMBER-HERE will be the H.323ID Asterisk will use to make the call. exten = 244,1,Dial(h323/[EMAIL PROTECTED]) is the proper format. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 support
Julio Tommasi wrote: Have any body succesfully compiled the files in asterisk-oh323-0.2.tar.gz ? This is a very, very old version. Try the latest one (0.5.1) from http://www.inaccessnetworks.com/projects/asterisk-oh323 Michael. I have the following errors: +for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG -I/usr/include -I/usr/include/crypto -I/root/pwlib/include/ptlib/unix -I/root/pwlib/include -I/root/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o wrapper.o cc1plus: warning: changing search order for system directory /usr/include cc1plus: warning: as it has already been specified as a non-system directory wrapper.cxx: In constructor `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int, int, int, short unsigned int)': wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function) wrapper.cxx:563: (Each undeclared identifier is reported only once for each function it appears in.) wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 I have the latest versions of PWlib and openh323. May be I need the same versions that appear in the README file, but I can't get them. Does any body have these versions ? Thanks Julio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
