RE: [Asterisk-Users] Sip provider problem or?
Sorry. Forgot to say that if I connect an ip phone directly to the provider it works without problwm Anders From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anders Svensson Sent: den 8 november 2005 11:09 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Sip provider problem or? Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sip provider problem or?
Anders Svensson wrote: Hi! We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Try something like: disallow=all allow=ulaw If that works then you do some trial and error to see which codecs are really supported. I remember doing this with a broadvoice account on incoming. The primary DID worked but it seemed that the tollfree virtual number did not allow the same codecs. Don't make any assumptions. Test everything. I have seen cases where termination and origination codecs allowed were different. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Sip provider problem or?
We are running an * with 3 sip providers. Provider 1 works perfect, provider 2 also. But the 3:rd one is a problem. All seems normal until we try to make a call. The phone rings by the called party and picks is up and hear only silence. The caller (local extension on the *) still gets ring tone as of no one answer the call. The providers ssw treats the call as answered and get no errors Any hints where to start looking? Turn on sip debug from the CLI and place a test call. There should be some pretty good clues that would tell you what's happening (or not happening). Best guess... probably a codec incompatibility issue. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
