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.
Are you using oh323 ? Paul Mahler pmahler@signate.com Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] 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 > 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
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:pmahler@signate.com] 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 pmahler@signate.com Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] 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 > 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
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:kido@cafe.tg] 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" <j.alayon@ses.com.ar> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> 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 believeis> the Nufone Channel. > Previously I did compile the PWLIB and OH323 packets. > > Is that correct ? > > Regards, > > Jorge A. > > -----Mensaje original----- > De: Paul Mahler [mailto:pmahler@signate.com] > 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 > pmahler@signate.com > Signate, LLC > 665 Third Street > Suite 100 > San Francisco, CA > 94107-1901 > > Asterisk Services and Training > > > > > > > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] 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 > > 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 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
> Message: 4 > Date: Sun, 21 Nov 2004 17:56:10 -0800 > From: "Paul Mahler" <pmahler@signate.com> > Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <20041121205619.GA56563@mail26f.sbc-webhosting.com> > Content-Type: text/plain; charset="us-ascii" > > Are you using oh323 ? > > Paul Mahler > pmahler@signate.com > Signate, LLC > 665 Third Street > Suite 100 > San Francisco, CA > 94107-1901 > > Asterisk Services and Training > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] 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 > > 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 >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
Thank you, I will see into it. Regards, Jorge A. -----Mensaje original----- De: Paul Davidson [mailto:planac@gmail.com] Enviado el: Monday, November 22, 2004 12:12 PM Para: asterisk-users@lists.digium.com Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper> Message: 4 > Date: Sun, 21 Nov 2004 17:56:10 -0800 > From: "Paul Mahler" <pmahler@signate.com> > Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <20041121205619.GA56563@mail26f.sbc-webhosting.com> > Content-Type: text/plain; charset="us-ascii" > > Are you using oh323 ? > > Paul Mahler > pmahler@signate.com > Signate, LLC > 665 Third Street > Suite 100 > San Francisco, CA > 94107-1901 > > Asterisk Services and Training > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] 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 > > 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 >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 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