SV: [asterisk-users] RESEND: Blind transfer # not working forforwarded or picked calls
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Eric ManxPower Wieling Skickat: den 5 december 2006 15:06 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [asterisk-users] RESEND: Blind transfer # not working forforwarded or picked calls Roger Lewau wrote: Resending this since I got no response Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 The t and or T option to Dial must be set. --- T and t is set for the dialed extension. If you answer the dialed extension by picking up the handset you can use the # key to transfer the call. But if you pickup the call from any other extension with *8 or if you had unconditional forwarding on the dialed extension, the extension who pickup or receive the call from forwarding is not able to use the # key to further transfer the call. Kind regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RESEND: Blind transfer # not working for forwarded or picked calls
Resending this since I got no response Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 Kind regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blind transfer # not working for forwarded or picked calls
Hello list We have a situation where calls need to be transfered to another extension. We are using # to accomplish this but we found this is only working for calls answered at the original called extension. If the call has been forwarded to another extension or if the call has been picked up by any other phone in the same pickup group the # key does not work. How can we solve this issue? Any parameters that need to be set? We are using Asterisk 1.2.13 Kind regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Slow playback of sound prompts
I have this very weird situation where some callers hear the playback of sound prompts on half speed. It only lasts a few second but it can happen at any time during playback. My server is a 3.4 Ghz Xeon with 1 GB RAM and 80 GB SATA disk. I run Asterisk 1.2.13 on FreeBSD 6.1 Anyone who has a clue to what can be the cause of this? Med vänliga hälsningar/Kind regards Roger Lewau Serverhallen i Norden ABBox 20087, 200 74 Malmö, SwedenTel: +46-40-6905000Fax: +46-40-6905001 Web: www.serverhallen.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] Problems with voicemail
I checked the permitions and updated the ones with the wrong permissions. No it is reading the number of messages correct, but as soon as I press 1 to listen it stops again. So again, I checked the permissions on the messagefolder but it seemed ok. I see now that another person on this lista has the exact same problem. Kind regards Roger -Original Message- From: Dinesh Nair [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Thu, 23 Feb 2006 20:00:30 +0800 Subject: Re: SV: [Asterisk-Users] Problems with voicemail On 02/22/06 23:11 Roger Lewau said the following: Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' it's borking when attempting to read numbers. is sounds/digits populated with adequate perms ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo=== ===+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +== ===+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with voicemail
Hello list Recently I run into some serious voicemail problems. When I call voicemailmain the prompts are presented ok if the voicemailbox is empty. If the voicemailbox contains messages the voicemail application exits with a "non-zero status" either when reading the number of messages or when selecting 1 for listening to new messages. I use Asterisk 1.2.4 on FreeBSD 5.4 without zaptel and G729a codecs installed. Any one with any pointers or idea Regards Roger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Problems with voicemail
Hello At the moment we are running asterisk as root. But I just checked and did chmod 777 to be certain, but the problem still remains. This is the output from asterisk. Connected to Asterisk 1.2.4 currently running on ns2 (pid = 47562) Verbosity is at least 9 -- Remote UNIX connection -- Executing VoiceMailMain(SIP/asterisk-0946, @sip) in new stack -- Playing 'vm-login' (language 'se') -- Playing 'vm-password' (language 'se') -- Playing 'vm-youhave' (language 'se') == Spawn extension (sip, 990, 1) exited non-zero on 'SIP/asterisk-0946' Regards Roger -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Johnathan Corgan Skickat: den 22 februari 2006 15:52 Till: Asterisk Users Mailing List - Non-Commercial Discussion Ämne: Re: [Asterisk-Users] Problems with voicemail Roger Lewau wrote: If the voicemailbox contains messages the voicemail application exits with a non-zero status either when reading the number of messages or when selecting 1 for listening to new messages. Is it possible the permissions for the sounds directory or individual files within have changed such that the user asterisk runs under no longer has read access? -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
