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you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa <daniel_eboa@creolink.com> wrote:> > > > Hello all, > > > > I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b > channel driver for H323. All installation and packages compilation was > successful. I have a SIP account to a SIP provider and I use it for outgoing > calls. I'm using Cisco ATA boxes both SIP and H323, and all the boxes > connect to my Asterisk Server. I can call a SIP box from H323 and vis-versa. > But when I want to dial out through my SIP account from my H323 box, the > call goes through but I got this error: chan_oh323.c:3180 > setup_h323_connection: Channel's format changed from 8 to 4??? > > Can some body help me out to find where is the problem ?? > > > > Thanks. > > > > > > > _______________________________________________ > 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 > >
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This is the version with whom i compile asterisk-oh323 channel successfully. I try the latest version of both asterisk and asterisk-oh323, but impossible to compile -----Original Message----- From: Paradise Dove [mailto:pardove@gmail.com] Sent: vendredi 12 novembre 2004 18:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Strange error you're using out of date and buggy versions of * and oh323. try to update them and check if the error is occurring again. On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa <daniel_eboa@creolink.com> wrote:> > > > Hello all, > > > > I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b > channel driver for H323. All installation and packages compilation was > successful. I have a SIP account to a SIP provider and I use it foroutgoing> calls. I'm using Cisco ATA boxes both SIP and H323, and all the boxes > connect to my Asterisk Server. I can call a SIP box from H323 andvis-versa.> But when I want to dial out through my SIP account from my H323 box,the> call goes through but I got this error: chan_oh323.c:3180 > setup_h323_connection: Channel's format changed from 8 to 4??? > > Can some body help me out to find where is the problem ?? > > > > Thanks. > > > > > > > _______________________________________________ > 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
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again. Error No. 2". But, after sometime, its working fine again without doing anything. My intercom is also working fine always. What is this error? Please tell me the solution. Looking forward to your response. Thanks&Regards, Chandra. --------------------------------- Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1?/min. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060726/0f2da929/attachment.htm
This post is from three days ago. Dont know if you found a solution or not. It
sounds look to be your provider. What comes up in the CLI ?
----- Original Message -----
From: Crazy Boy
To: asterisk-users@lists.digium.com
Sent: Wednesday, July 26, 2006 3:39 AM
Subject: [asterisk-users] Strange error
Hi Friends,
We are using "Asterisk" in our office and using "XLite" as
softphone and "Teliax" service for USA dialing. Sometimes It is
working fine. But, sometime, when i am trying to make a call to USA, my
softphone is telling that "I am sorry. That is not a valid extension.
Please try again. Error No. 2". But, after sometime, its working fine again
without doing anything. My intercom is also working fine always. What is this
error? Please tell me the solution.
Looking forward to your response.
Thanks&Regards,
Chandra.
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starting at 1?/min.
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