RE: [asterisk-users] H323-to-SIP proxy
T.38 pass-through should work fine on the SIP leg. (With Asterisk 1.40) There are a few bugs but you can get past them. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] H323-to-SIP proxy What about the SIP leg? - Mensaje Original - De: "Michelle Dupuis" <[EMAIL PROTECTED]> Para: "Asterisk Users Mailing List - Non-Commercial Discussion" Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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 ___ --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] H323-to-SIP proxy
What about the SIP leg? - Mensaje Original - De: "Michelle Dupuis" <[EMAIL PROTECTED]> Para: "Asterisk Users Mailing List - Non-Commercial Discussion" Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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 ___ --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] H323-to-SIP proxy
T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --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] H323 to SIP - One way voice
Which H.323 channel driver are you using, and could you post a log or debug of a session. Craig - Original Message - From: "Andrei U" <[EMAIL PROTECTED]> To: Sent: Thursday, February 08, 2007 2:41 AM Subject: [asterisk-users] H323 to SIP - One way voice Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you, Andrei U ___ --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] h323 to sip ringing indication
On Saturday 20 May 2006 16:31, Roman Yeryomin wrote:: > Hello all! > > I have a problem with ringing indication when calling from h323 (oh323+open > phone client) to sip users. The phone rings and users can talk to each > other with no problems but the calling h323 user hear silence unless sip > user picks up the phone. > Calling to pstn no problems. Calling from sip to that open phone client > also no problems. > I tried latest ooh323 and oh323... no difference > Also passing "r" option to dial doesn't help. > > Does anyone know where could be the problem? > > Roman That's strange, but it's working now... I didn't change anything.. ___ --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] H323 to SIP
Farhad Ibragimov wrote: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me Asterisk is perfectly documented everywhere on the net. Maybe the first place to visit in order to have working asterisk is www.asterisk.org.Second place is www.voip-info.org If any question arises feel free to email me privately. Tofik Suleymanov ___ --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] H323 to SIP
On Sun, 7 May 2006 19:58:26 +0500, "Farhad Ibragimov" <[EMAIL PROTECTED]> wrote: > Thanks > Try reading this URL (spanish language): http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 With the page instructions I can call from and to H.323 to every registred SIP/IAX2/H.323 device with my Asterisk box. Good luck, > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alberto > Sagredo > Sent: Sunday, May 07, 2006 7:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] H323 to SIP > > You could begin with: > > http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation > > http://www.voip-info.org/wiki/view/Asterisk+H323+channels > > http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels > > and much more. > > You need to install chan_h323 module and configure as well as you need > in your application, (if you need gatekeeper functionality maybe you > need to try before GNUGK), and later via extensions make wherever you > need. > > Its a little complicated and you need how to work with asterisk before > doing all this things. > > Regards > > Farhad Ibragimov escribió: >> I dont have practice to work with Asterisk but I see that is a great > soft. >> If you have any idea or some config files can you help me >> >> >> -Original Message- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Alberto >> Sagredo >> Sent: Sunday, May 07, 2006 7:34 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] H323 to SIP >> >> You could make a H323 to SIP transport. Before to do that, you need to >> have installed and working both chan protocolos on Asterisk. >> >> aFarhad Ibragimov escribió: >> >>> Hi all >>> >>> I have installed station which support only H323 protocol. I want to >>> install SIP telephone. Is it possible to call SIP telephone throught >>> my station >>> >>> > >>> >>> ___ >>> --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 >> >> >> ___ >> --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 > > > ___ > --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 -- Guillermo V. Salas M Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 1er Piso Teléfono: 262 8071 Celular : 09 985 5138 Manta - Manabí - Ecuador ___ --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] H323 to SIP
Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: > I dont have practice to work with Asterisk but I see that is a great soft. > If you have any idea or some config files can you help me > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alberto > Sagredo > Sent: Sunday, May 07, 2006 7:34 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] H323 to SIP > > You could make a H323 to SIP transport. Before to do that, you need to > have installed and working both chan protocolos on Asterisk. > > aFarhad Ibragimov escribió: > >> Hi all >> >> I have installed station which support only H323 protocol. I want to >> install SIP telephone. Is it possible to call SIP telephone throught >> my station >> >> >> >> ___ >> --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 > > > ___ > --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 ___ --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] H323 to SIP
You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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 ___ --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] H323 to SIP
I dont have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: > > Hi all > > I have installed station which support only H323 protocol. I want to > install SIP telephone. Is it possible to call SIP telephone throught > my station > > > > ___ > --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 ___ --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] H323 to SIP
You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --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] H323 to SIP
BJ, You were exactly right! The context was screwed up in the oh323.conf which points to the extensions.conf...now the issue I have is as follows: TDM > h323 > * > SIP Endpoint (private IP) > rings once and goes dead H323 Endpoint > * > H323 > TDM > no audio Comuniquémonos, Inc. / SA Jeromy Grimmett CEO [EMAIL PROTECTED] 1212 South Hampton Drive Alexandria, LA 71301 tel: +593 (4) 287 3854 fax: (501) 646-0680 mobile: +593 (9) 366 6521 IM: MSN: [EMAIL PROTECTED] http://www.comuniquemonos.com -Original Message- From: BJ Weschke [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 17, 2005 1:55 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You need to post your extensions.conf and oh323.conf for further assistance. It sounds like though that the h.323 endpoints are sending a call to you and since you didn't define a default extension/context for them to go to, they are trying to go to extension 's' in the default context, but this isn't defined either. On 5/17/05, Jeromy Grimmett <[EMAIL PROTECTED]> wrote: > > Hi all, > > Of course I am a newbie, so please bear with me... > > I'm having a lot of trouble getting things to work properly and I am > sure it is a configuration issue somewhere, I'm just not sure > where...I have been all through my extensions.conf and cannot seem to > see a problem. > > SIP Endpoint (w/ private IP) > Asterisk > H323 (public IP) > TDM > > works perfect > > SIP Endpoint (w/private IP) > Asterisk > H323 Endpoint (w/ private IP) > > works perfect > > H323 Endpoint (w/ private IP) > Asterisk > H323 (public IP) > TDM > fails with this message: > > Inbound H.323 call 'ip$192.168.6.176:10270/2258' detected. Channel > OH323/R2258 created and attached for inbound H.323 call > 'ip$192.168.6.176:10270/2258'. May 17 21:46:33 WARNING[29317]: > pbx.c:1890 ast_pbx_run: Channel 'OH323/R2258' sent into invalid > extension 's' in context 'default', but no invalid handler > Call 'ip$192.168.6.176:10270/2258' cleared. > Call 'ip$192.168.6.176:10270/2258' without owner has already been cleared > (1). > Cancelled scheduled release of call 'ip$192.168.6.176:10270/2258'. > > H323 ATA 186 (w/ private IP) > Asterisk > SIP Softphone (w/ private > IP) > fails with this message: > > Inbound H.323 call 'ip$192.168.6.176:2229/10680' detected. Channel > OH323/R10680 created and attached for inbound H.323 call > 'ip$192.168.6.176:2229/10680'. May 17 21:43:50 WARNING[29317]: > pbx.c:1890 ast_pbx_run: Channel 'OH323/R10680' sent into invalid > extension 's' in context 'default', but no invalid handler > Call 'ip$192.168.6.176:2229/10680' cleared. > Call 'ip$192.168.6.176:2229/10680' without owner has already been cleared > (1). > Cancelled scheduled release of call 'ip$192.168.6.176:2229/10680'. > > TDM > H323 (public IP) > Asterisk > SIP Endpoint (w/ private IP) > > fails with this message: > > May 17 05:29:20 WARNING[27823]: pbx.c:1890 ast_pbx_run: Channel > 'OH323/R156' sent into invalid extension 's' in context 'default', but > no invalid handler Call 'ip$200.94.273.2:10172/156' cleared. > > TDM > H323 (public IP) > Asterisk > H323 Endpoint (w/ private IP) > > fails with this message: > > Inbound H.323 call 'ip$200.93.237.82:10237/222' detected. Channel > OH323/R222 created and attached for inbound H.323 call > 'ip$200.93.237.82:10237/222'. May 17 21:49:43 WARNING[29317]: > pbx.c:1890 ast_pbx_run: Channel 'OH323/R222' sent into invalid > extension 's' in context 'default', but no invalid handler Call > 'ip$200.93.237.82:10237/222' cleared. Call > 'ip$200.93.237.82:10237/222' without owner has already been cleared > (1). Cancelled scheduled release of call 'ip$200.93.237.82:10237/222'. > > anyone with any ideas i would greatly appreciate it... > > Thanks, > Jeromy > > > Global reach, local touch... > Jeromy Grimmett > CEO Comuniquémonos, Inc. / SA > 1212 South Hampton Drive > Alexandria, LA 71301 > [EMAIL PROTECTED] > IM: MSN: [EMAIL PROTECTED] http://www.comuniquemonos.com > tel: > fax: > mobile: +593 (4) 287 3854 > (501) 646-0680 > +593 (9) 366 6521 > Add me to your address book...Want a signature like this? > > ___ > 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 > > ___ 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] H323 to SIP
You need to post your extensions.conf and oh323.conf for further assistance. It sounds like though that the h.323 endpoints are sending a call to you and since you didn't define a default extension/context for them to go to, they are trying to go to extension 's' in the default context, but this isn't defined either. On 5/17/05, Jeromy Grimmett <[EMAIL PROTECTED]> wrote: > > Hi all, > > Of course I am a newbie, so please bear with me... > > I'm having a lot of trouble getting things to work properly and I am sure it > is a configuration issue somewhere, I'm just not sure where...I have been > all through my extensions.conf and cannot seem to see a problem. > > SIP Endpoint (w/ private IP) > Asterisk > H323 (public IP) > TDM > works > perfect > > SIP Endpoint (w/private IP) > Asterisk > H323 Endpoint (w/ private IP) > > works perfect > > H323 Endpoint (w/ private IP) > Asterisk > H323 (public IP) > TDM fails with > this message: > > Inbound H.323 call 'ip$192.168.6.176:10270/2258' detected. > Channel OH323/R2258 created and attached for inbound H.323 call > 'ip$192.168.6.176:10270/2258'. > May 17 21:46:33 WARNING[29317]: pbx.c:1890 ast_pbx_run: Channel > 'OH323/R2258' sent into invalid extension 's' in context 'default', but no > invalid handler > Call 'ip$192.168.6.176:10270/2258' cleared. > Call 'ip$192.168.6.176:10270/2258' without owner has already been cleared > (1). > Cancelled scheduled release of call 'ip$192.168.6.176:10270/2258'. > > H323 ATA 186 (w/ private IP) > Asterisk > SIP Softphone (w/ private IP) > > fails with this message: > > Inbound H.323 call 'ip$192.168.6.176:2229/10680' detected. > Channel OH323/R10680 created and attached for inbound H.323 call > 'ip$192.168.6.176:2229/10680'. > May 17 21:43:50 WARNING[29317]: pbx.c:1890 ast_pbx_run: Channel > 'OH323/R10680' sent into invalid extension 's' in context 'default', but no > invalid handler > Call 'ip$192.168.6.176:2229/10680' cleared. > Call 'ip$192.168.6.176:2229/10680' without owner has already been cleared > (1). > Cancelled scheduled release of call 'ip$192.168.6.176:2229/10680'. > > TDM > H323 (public IP) > Asterisk > SIP Endpoint (w/ private IP) > fails > with this message: > > May 17 05:29:20 WARNING[27823]: pbx.c:1890 ast_pbx_run: Channel 'OH323/R156' > sent into invalid extension 's' in context 'default', but no invalid handler > Call 'ip$200.94.273.2:10172/156' cleared. > > TDM > H323 (public IP) > Asterisk > H323 Endpoint (w/ private IP) > fails > with this message: > > Inbound H.323 call 'ip$200.93.237.82:10237/222' detected. > Channel OH323/R222 created and attached for inbound H.323 call > 'ip$200.93.237.82:10237/222'. > May 17 21:49:43 WARNING[29317]: pbx.c:1890 ast_pbx_run: Channel 'OH323/R222' > sent into invalid extension 's' in context 'default', but no invalid handler > Call 'ip$200.93.237.82:10237/222' cleared. > Call 'ip$200.93.237.82:10237/222' without owner has already been cleared > (1). > Cancelled scheduled release of call 'ip$200.93.237.82:10237/222'. > > anyone with any ideas i would greatly appreciate it... > > Thanks, > Jeromy > > > Global reach, local touch... > Jeromy Grimmett > CEO Comuniquémonos, Inc. / SA > 1212 South Hampton Drive > Alexandria, LA 71301 > [EMAIL PROTECTED] > IM: MSN: [EMAIL PROTECTED] > http://www.comuniquemonos.com > tel: > fax: > mobile: +593 (4) 287 3854 > (501) 646-0680 > +593 (9) 366 6521 > Add me to your address book...Want a signature like this? > > ___ > 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 > > ___ 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] h323 to SIP Server Load
Steve Totaro wrote: Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers. Not true for asterisk-oh323. Micheal. ___ 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] h323 to SIP Server Load
What solution provides a higher number of simultaneous calls? I found this http://www.mera-voip.com/voip/sip-hit.php. They claim 150 with a dedicated server and relatively modest hardware. Thanks, Steve Totaro - Original Message - From: "Jeremy McNamara" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, July 24, 2004 11:09 AM Subject: Re: [Asterisk-Users] h323 to SIP Server Load > Steve Totaro wrote: > > > Does anyone do any large scale SIP to H323 conversion? How many > > simultaneous calls can your server handle and on what hardware? I think > > I read on the wiki that twenty five would max out most servers. > > > The wiki is very wrong then. > > > 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] h323 to SIP Server Load
> Steve Totaro wrote: > > > Does anyone do any large scale SIP to H323 conversion? How many > > simultaneous calls can your server handle and on what hardware? I think > > I read on the wiki that twenty five would max out most servers. > > > The wiki is very wrong then. > > > Jeremy McNamara > That is what I figured. Care to share some actual numbers? ___ 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] h323 to SIP Server Load
Jeremy McNamara wrote: The wiki is very wrong then. At least regarding chan_h323. 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] h323 to SIP Server Load
Steve Totaro wrote: Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers. The wiki is very wrong then. 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] h323 to SIP Server Load
- Original Message - From: Steve Totaro To: [EMAIL PROTECTED] Sent: Saturday, July 24, 2004 6:31 AM Subject: [Asterisk-Users] h323 to SIP Server Load Does anyone do any large scale SIP to H323 conversion? How many simultaneous calls can your server handle and on what hardware? I think I read on the wiki that twenty five would max out most servers.
