[asterisk-users] Transfer call in SIP
Hello, I am running TrixBox. if already in a call session from ZAPTEL to SIP, the user want to transfer the call to a different extension. The user have to dial *extension ? Any configuration is needed to be done in trixbox? Thanks Victor ___ --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] How to retrieve voicemail
Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor ___ --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: [Asterisk-Users] Bug in Voicemail ??
Hi, I'm still a newbie, but try to help you, my voicemail works ok, I can also record messages ok. My extension part is: exten = s,1,Background(welcome-cisl) exten = 1,1,Dial(Sip/vmoreno,10) exten = 1,2,Voicemail(victor) exten = 2,1,Dial(Sip/juliansip,10) exten = 2,2,Voicemail(aajulian) exten = 3,1,Playback(demo-echotest) exten = 3,2,Echo exten = 4,1,Congestion exten = 5,1,Dial(Sip/ludmila,10) exten = 5,2,Dial(Sip/vmoreno) exten = 6,1,Goto(testmenu,s,1) And voicemail.conf part is: [general] format=wav49 maxmessage=180 minmessage=2 maxsilence=2 silencethreshold=150 maxlogins=3 [EMAIL PROTECTED] skipms=3000 [victor] victor = 1234, Victor Moreno, [EMAIL PROTECTED] Hope it helps. One question to you, you say you call the malbox, how do you do that? which extension do i have to call to a ccess mailboxes? thank u Victor -- Victor Moreno CISL SPAIN, S.L. Parque Tecnológico de Andalucía Edif. Bic Euronova Avda. Juan López Peñalver, 21 29590 Campanillas (Málaga) Fax +34 95 10 10 561 Tlfn.: +34 95 10 10 581 Web: http://www.cisl.es Email: [EMAIL PROTECTED] Skype: victor.moreno Stefan-Michael. Guenther (in-put GbR) wrote: Hello, I have setup an Asterisk 1.2.7.1 system, with a working voicemail box: /etc/asterisk/extensions.conf exten = 83086921,1,Answer exten = 83086921,2,Dial(SIP/stefan,5,r) exten = 83086921,3,VoiceMail,u111 exten = 83086921,4,Hangup exten = 83086921,103,VoiceMail,b111 exten = 83086921,104,Hangup /etc/asterisk/voicemail.conf [default] language=de 111 = 111,Mailbox 111,[EMAIL PROTECTED] The mailbox starts, I hear the intro and speak my message. In the CLI I can see that the message has been recorded and I get the recorded message via mail. But when I listen to the recorded messages or call the mailbox, I either hear nothing or just a short cracking sound. At least the length of the message is correct. If have tried to record the message with gsm, wav or wav49, the result is always the same. When I use the record() application to record a gsm file, everything is okay. I obviously made something wrong when configuring the voicemail system. Can someone give me a hint what's going wrong? Thanks for your help, stefan ___ --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] timeout 't'
Hello, I have found that the timeout 't' takes to much time to be executed, around 10 seconds. Is there a place to configure this timeout ? thanks -- Victor Moreno CISL SPAIN, S.L. Parque Tecnológico de Andalucía Edif. Bic Euronova Avda. Juan López Peñalver, 21 29590 Campanillas (Málaga) Fax +34 95 10 10 561 Tlfn.: +34 95 10 10 581 Web: http://www.cisl.es Email: [EMAIL PROTECTED] Skype: victor.moreno ___ --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] voicemail issue
Hello, I have created a voicemail with succes for user victor. But a second vocemail for user julian, asterisk claims that the voicemail for user 'lian' is not configured, why asterisk is getting rid of the first 2 chars of 'julian'. For the moment I have created user aajulian in both extensions.cfg and voicemail.cfg and with that tricks it works well, but i would like to know why i cannot use the username julian. Thanks Victor ___ --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] gsm file
Hi, I'm newby here, reading the handbook and starting playing with *. What are the audio .gsm files in /var/lib/asterisk/sounds ? Playback command can only play .gsm ? how do i convert from .wav to .gsm ? Thanks a lot Victor ___ --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
