[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
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 proxy
What about the SIP leg? - Mensaje Original - De: Michelle Dupuis [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com 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 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 asterisk-users@lists.digium.com 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 - 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: asterisk-users@lists.digium.com 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
[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
[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
[Asterisk-Users] H323 to SIP Gateway
I'm trying to setup an Asterisk box as an H323 to SIP gateway. Basically, I'd like to receive traffic in H323 and forward to another Asterisk box (on the same network) using either IAX2 or SIP so that the second Asterisk box communicates with other gateways using SIP. Therefore, if I receive a request from a remote H323 gateway to dial a particular number, the H323-to-SIP gateway should forward the request to the Asterisk SIP gateway, who would simply terminate the call according to whatever rules are defined in the context. Can anyone tell me how can this be done? I setup chan_oh323 on an * box and played with the configurations but have not been able to make it all work. I can place connect the two * boxes using SIP-to-SIP as well as IAX2-to-IAX2 just fine, but have not gotten the H323 to work. Thanks, Daniel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 to SIP connection problem
Everyone,
I have been trying to connect a PBX with H323 IP trunks
with g711 codec to my Asterisk server running ooh323 service. I can place calls
to and from either the Asterisk, or PBX with no problem, but when I try to
pickup the call on either end, the phone hangs up immediately. Debug shows
normal to me but at the last few lines of data there is an error shown that I
have not been able to find any information to help with troubleshooting. Could
someone assist with the fix?
**Last few line from logfile**
12:48:19:257 Queued H245 messages 1. (outgoing,
ooh323c_o_2)
12:48:19:257 msgCtxt Reset? Done (outgoing,
ooh323c_o_2)
12:48:19:257 MasterSlaveDetermination done -
Slave(outgoing, ooh323c_o_2)
12:48:19:257 Not opening logical channels as Cap
exchange remaining
12:48:19:257 Finished handling H245 message.
(outgoing, ooh323c_o_2)
12:48:19:257 Receiving H.2250 message (outgoing,
ooh323c_o_2)
12:48:19:257 Received Q.931 message: (outgoing,
ooh323c_o_2)
12:48:19:257 Received H.2250 Message = {
12:48:19:258 protocolDiscriminator
= 8
12:48:19:258 callReference = 43
12:48:19:258 from = destination
12:48:19:258 messageType = 5a
12:48:19:258 Cause IE = {
12:48:19:258
Unsupported Cause Type
12:48:19:258 }
12:48:19:258 h323_uu_pdu = {
12:48:19:258
h323_message_body = {
12:48:19:258
releaseComplete = {
12:48:19:259
protocolIdentifier = {
12:48:19:259
{
12:48:19:260 0 0 8 2250 0 2 }
12:48:19:261
}
12:48:19:261
callIdentifier = {
12:48:19:262
guid = {
12:48:19:262
'6f6f68333233632d818e86c6'H
12:48:19:263
}
12:48:19:264
}
12:48:19:264
}
12:48:19:265 }
12:48:19:265 }
12:48:19:265 UUIE decode successful
12:48:19:265 Decoded Q931 message (outgoing,
ooh323c_o_2)
12:48:19:265 }
12:48:19:265 H.225 Release Complete message received
(outgoing, ooh323c_o_2)
12:48:19:265 Cause of Release Complete is 0.
(outgoing, ooh323c_o_2)
12:48:19:265 Closing H.245 connection (outgoing,
ooh323c_o_2)
12:48:19:266 Closed H245 connection. (outgoing,
ooh323c_o_2)
12:48:19:266 In ooEndCall call state is -
OO_CALL_CLEARED (outgoing, ooh323c_o_2)
12:48:19:266 Cleaning Call (outgoing, ooh323c_o_2)-
reason:OO_REASON_UNKNOWN
12:48:19:266 Removed call (outgoing, ooh323c_o_2)
from list
DJ
___
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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
[Asterisk-Users] h323 to sip ringing indication
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 ___ --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
[Asterisk-Users] H323 to SIP
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
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
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 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
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
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
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
[Asterisk-Users] h323 to sip
I have a problem when dialing from outside line to sip server. I get this output on debug. Could someone give me a hint what could be wrong? == Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to exten 's' == Starting OH323/R30149 at from-pstn,s,1 still failed so falling back to context 'default' How do you make connection between incoming h323 connection and user who is connected to sip server? ___ 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] h323 to sip
I have a problem when dialing from outside line to sip server. I get this output on debug. Could someone give me a hint what could be wrong? == Starting OH323/R30149 at from-pstn,6000622,1 failed so falling back to exten 's' == Starting OH323/R30149 at from-pstn,s,1 still failed so falling back to context 'default' How do you make connection between incoming h323 connection and user who is connected to sip server? ___ 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] H323 to SIP
Title: Message 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. SIPEndpoint (w/ private IP) Asterisk H323 (public IP) TDM works perfect SIPEndpoint (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 handlerCall '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 handlerCall '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 SIPEndpoint (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 handlerCall 'ip$200.94.273.2:10172/156' cleared. TDM H323 (public IP) Asterisk H323Endpoint (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 handlerCall '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 GrimmettCEO Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, 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? small comuniquemonos.jpg___ 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] H323 to SIP
Title: Message 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. SIPEndpoint (w/ private IP) Asterisk H323 (public IP) TDM works perfect SIPEndpoint (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 handlerCall '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 handlerCall '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 SIPEndpoint (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 handlerCall 'ip$200.94.273.2:10172/156' cleared. TDM H323 (public IP) Asterisk H323Endpoint (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 handlerCall '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 GrimmettCEO Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, 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? small comuniquemonos.jpg___ 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
[Asterisk-Users] H323 to SIP
Title: Message Hi all, All outbound calls work perfect from my SIP ATA 186... SIP ATA 186 (w/ private IP) Asterisk H323 (public IP) TDM works perfect TDM H323 (public IP) Asterisk SIP ATA 186 (w/ private IP) fast busy with this error message in the CLI of Asterisk: 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 handlerCall 'ip$200.94.273.2:10172/156' cleared. anyone with any ideas i would greatly appreciate it... Thanks, Jeromy Global reach, local touch... Jeromy GrimmettCEO Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, 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? small comuniquemonos.jpg___ 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] H323-Asterisk-SIP-TNT consultant needed
We are in urgent need of some help getting Asterisk to gateway between an incoming H323 connection and SIP to a Lucent TNT. We have the incoming H323 already set up and the SIP going to the TNT but the media stream is getting lost somewhere as no audio is heard. We are willing to pay $$$ for an extra set of eyes to get this resolved fast. It's probably something quick and easy and we are just missing it. Email me ASAP if you are willing to help. -- Tracy Reedhttp://copilotcom.com This message is cryptographically signed for your protection. Info: http://copilotconsulting.com/sig pgp4oM5Pj0XGE.pgp Description: PGP signature ___ 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. 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
[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.
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.
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
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 and SIP
have you looked at digiums site? there are few simple sample there. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Adelino BaenaSent: 09 August 2003 21:47To: [EMAIL PROTECTED]Subject: [Asterisk-Users] H323 and SIP Dear Colleagues I am a newbie on Asterisk and am having difficulties to find documentation about how to configure the H323 and Sip services. Could somebody of you have the kindness to send me functional samples of conf files to my personal e-mail ? Im testing two VoIP clients: H.323 (OpenH323 Client) and SIP (X-Ten). Tks in advance and best regards Adelino Baena [EMAIL PROTECTED] Brazil ---Outgoing mail is certified Virus Free.Checked by AVG anti-virus system (http://www.grisoft.com).Version: 6.0.507 / Virus Database: 304 - Release Date: 4/8/2003
Re: [Asterisk-Users] H323 and SIP
try this: http://www.loligo.com/asterisk/current/ Lubo Adelino Baena wrote: Dear Colleagues I am a newbie on Asterisk and am having difficulties to find documentation about how to configure the H323 and Sip services. Could somebody of you have the kindness to send me functional samples of conf files to my personal e-mail ? Im testing two VoIP clients: H.323 (OpenH323 Client) and SIP (X-Ten). Tks in advance and best regards Adelino Baena [EMAIL PROTECTED] Brazil --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.507 / Virus Database: 304 - Release Date: 4/8/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 and SIP
Dear Colleagues I am a newbie on Asterisk and am having difficulties to find documentation about how to configure the H323 and Sip services. Could somebody of you have the kindness to send me functional samples of conf files to my personal e-mail ? Im testing two VoIP clients: H.323 (OpenH323 Client) and SIP (X-Ten). Tks in advance and best regards Adelino Baena [EMAIL PROTECTED] Brazil --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.507 / Virus Database: 304 - Release Date: 4/8/2003
