I have a feeling the dchannel is bad. I'll investigate further and post my findings.
-Jon ----- Original Message ----- From: "Atis" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Thursday, September 13, 2007 4:12 AM Subject: Re: [asterisk-users] No Sound on Zap Channels > On 9/13/07, Hoai-Anh Ngo-Vi <[EMAIL PROTECTED]> wrote: >> Have you answered the channel? > > Voicemail doesn't require Answer(). It does that itself, as you > usually get to voicemail after Dial(). It would be silly to require to > do Answer after each Dial and then send to voicemail. > > Regards, > Atis > >> >> Von: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] Im Auftrag >> von Jon Weisman > >> I've got a strange issue here. When I make a SIP call to say my voicemail >> app, I hear audio just fine. However when I dial from PSTN into my >> Asterisk >> box, I see that its playing the voice files, but I hear nothing, then the >> call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI output >> below. T-1 is PRI, showing normal, dchannel is up as well. Any help is >> greatly appreciated. >> >> >> >> >> >> >> >> >> Thanks, >> >> >> Jon >> >> >> >> >> >> >> >> >> -- Accepting call from '2125551212' to '6465551212' on channel 0/23, >> span 4 >> -- Executing VoiceMail("Zap/95-1", "u100") in new stack >> -- Playing 'vm-theperson' (language 'en') >> -- Playing 'digits/1' (language 'en') >> -- Playing 'digits/0' (language 'en') >> -- Playing 'digits/0' (language 'en') >> -- Playing 'vm-isunavail' (language 'en') >> -- Playing 'vm-intro' (language 'en') >> -- Channel 0/23, span 4 got hangup request, cause 34 >> == Spawn extension (default, 6465551212, 1) exited non-zero on >> 'Zap/95-1' >> -- Hungup 'Zap/95-1' >> _______________________________________________ >> >> Sign up now for AstriCon 2007! September 25-28th. >> http://www.astricon.net/ >> >> --Bandwidth and Colocation Provided by http://www.api-digital.com-- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > Atis Lezdins, > IT Responsible of BEST Riga, > [EMAIL PROTECTED] > ICQ: 142239285 > Skype: atis.lezdins > Cell Phone: +371 28806004 [Tele2, Latvia] > Work phone: +1 800 7502835 [Toll free, USA] > ?BEST? -> www.BEST.eu.org > > _______________________________________________ > > Sign up now for AstriCon 2007! September 25-28th. > http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
