Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper
Hi Paul, Are you using the h323 or the oh323 channel. Please, what is the status of the bug that you are talking about? Thanks - Original Message - From: Paul Davidson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 22, 2004 3:11 PM Subject: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Asterisk and H.323 Gatekeeper
I'm using the NuFone H323 channel, not the Oh323 channel. I've had problems with both in different situations- in this case, I haven't tried the Oh323 channel in some time. (but I might, given these problems). To date, JerJer (Jeremy McNamara, the NuFone developer who's been assigned the bug) has not responded to my latest log and description of the problem. You can track and monitor the problem yourself through bugs.digium.com if you like- you'll get the updates as I do in that situation. Thanks! -pbd On Tue, 23 Nov 2004 19:37:36 -, kido noagbodji [EMAIL PROTECTED] wrote: Hi Paul, Are you using the h323 or the oh323 channel. Please, what is the status of the bug that you are talking about? Thanks - Original Message - From: Paul Davidson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 22, 2004 3:11 PM Subject: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Asterisk and H.323 Gatekeeper
I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 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] Asterisk and H.323 Gatekeeper
Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way. K. - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 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] Asterisk and H.323 Gatekeeper
Thank you, I will need a SIP client with G723 and/or G.729 then. Do you know any sip clients that do both ? Regards, Jorge A. -Mensaje original- De: kido noagbodji [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 8:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way. K. - Original Message - From: Jorge Alayon [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets. Is that correct ? Regards, Jorge A. -Mensaje original- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 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 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] Asterisk and H.323 Gatekeeper
Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Asterisk and H.323 Gatekeeper
Thank you, I will see into it. Regards, Jorge A. -Mensaje original- De: Paul Davidson [mailto:[EMAIL PROTECTED] Enviado el: Monday, November 22, 2004 12:12 PM Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper Message: 4 Date: Sun, 21 Nov 2004 17:56:10 -0800 From: Paul Mahler [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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 I have been working with this precise same issue, under bug number 0002659. I've seen this problem all the way up to CVS-HEAD-11/21. In my case, I'm using the gnuGK gatekeeper, and connecting to cisco callmanager 3.3.3. While callmanager can call in to Asterisk via the gateway, calls do not proceed in the other direction- the only difference between this setup and my own (aside from a different gatekeeper) is that mine is 100% H.323 with IAX softphones used to attempt the call. I've been bouncing stuff back and forth with JerJer on this isse- one thing that might help you (it didn't help me) is to use CVS-HEAD, which will require an update to OpenH323 and PWLIB (that was a long evening). Not much help- but at least know you're not alone. -pbd ___ 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] Asterisk and H.323 Gatekeeper
Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jorge Alayon Sent: Friday, November 19, 2004 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to the H.323 network but call does not progress from the SIP to the H.323 network. This is the ASterisk console output. -- Registered SIP '1154538511' at 192.168.11.46 port 5060 expires 1800 -- Executing Wait(SIP/1154538511-ed8a, 2) in new stack -- Executing Dial(SIP/1154538511-ed8a, h323/01145568423) in new stack -- Called 01145568423 == No one is available to answer at this time -- Timeout on SIP/1154538511-ed8a == CDR updated on SIP/1154538511-ed8a -- Executing Goto(SIP/1154538511-ed8a, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/1154538511-ed8a, demo-thanks) in new stack -- Playing 'demo-thanks' (language 'en') -- Executing Hangup(SIP/1154538511-ed8a, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/1154538511-ed8a' If I dial from an ATA, An AS5300, or an Audiocodes GW the number 01145568423 through the Gatekeeper, it works. Any ideas ? Regards, Jorge A. ___ 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
