[Asterisk-Users] How to Confiure Voicetronix V4PCI16 in asterisk
Hi All I have one voicetronix V4PCI16 card .I want to know how to configure this in linux with asterisk. I need to integrate this one with traditional PBX.pls do the needful. regards ramakrishnan.n __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] How to Confiure Voicetronix V4PCI16 in asterisk
Hi AllI have one voicetronix V4PCI16 card .I want to know how to configure this in linux with asterisk. I need to integrate this one with traditional PBX.pls do the needful.regards ramakrishnan.n Yahoo! Photos Ring in the New Year with Photo Calendars. Add photos, events, holidays, whatever.___ --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] Call forwarding for particular extension when line 1 is busy
Hi All Thanks for ur reply.My phone having 2 line with same extension and also I configure voicemail if the user not pickup the phone within 25 seconds for tht extension but i want if my line 1 is busy then forward the call to some other extension .my config is like following.my phone having the adhoc conference facility so tht I need 2 lines I am using SIPURA IP phones.pls do the needful... exten = 2007,1,Dial(SIP/sipura3,25,r) exten = 2007,2,VoiceMail([EMAIL PROTECTED]) regards ramakrishnan.n __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.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
[Asterisk-Users] Call forwarding for particular extension
Hi all I need to configure call forwarding for particular extension is busy.how to configure this my extension configuration is like following. exten = 2006,1,Dial(SIP/sipura2) regards ramakrishnan.n __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding for particular extension
Hi All
Thanks for ur reply.My phone having 2 line with same
extension and also I configure voicemail if the user
not pickup the phone within 25 seconds for tht
extension but i want if my line 1 is busy then forward
the call to some other extension .my config is like
following.my phone having the adhoc conference
facility so tht I need 2 lines I am using SIPURA IP
phones.pls do the needful...
exten = 2007,1,Dial(SIP/sipura3,25,r)
exten = 2007,2,VoiceMail([EMAIL PROTECTED])
regards
ramakrishnan.n
--- Giovanni Miano [EMAIL PROTECTED] wrote:
exten = 2006,2,goto(s-${DIALSTATUS},1)
exten = s-BUSY,1,DIAL(SIP/sipura3)
exten = s-NOANSWER,1,
exten = s-
www.*voip-info*.org/wiki-Asterisk+variable+DIALSTATUS
Cheers,
Giovanni Miano
2006/1/6, nr k [EMAIL PROTECTED]:
Hi all
I need to configure call forwarding for particular
extension is busy.how to configure this my
extension
configuration is like following.
exten = 2006,1,Dial(SIP/sipura2)
regards
ramakrishnan.n
__
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[Asterisk-Users] Asterisk Voice mail-reg
HI allHow to configure voice mail in asterisk . pls do the needful. regards ramakrishnan.n __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
[Asterisk-Users] Maximum Number of SIP Phones Supported By Asterisk
Hi All Can anybody tell me the maximum number of SIP Phones supported by Asterisk. regards ramakrishnan.n __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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] Maximum Number of SIP Phones Supported By Asterisk
Hi I am having Pentium 4 Machine with 256MB Memory and i am using codec g723. regards ramakrishnan.n --- trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Mon, 2005-11-14 at 06:53 -0800, nr k wrote: Hi All Can anybody tell me the maximum number of SIP Phones supported by Asterisk. If I run asterisk on my ipaq not very many. If I run it on a real server many many more. Your question cant really be answered with the information you have provided. It can only be answered in context. What hardware? What codecs? Any translations (from one medium/codec to another)? What applications are used (AGI, conferences, voicemail, etc)? Is the asterisk server actually pushing the bits for a call or just doing call setup and connecting the two endpoints directly? These are the very minimum questions you have to answer before your question can be answered. There are a few other things that can go into it, but those will help you better define for a rough idea ... And based on the answers to those questions there may be more questions. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
[Asterisk-Users] maximum concurrent conference peers in asterisk
Hi all I have a 2 mbps bandwidth in my central location and 64 kbps in all my branch office. what is the maximum concurrent conference peers possible through asterisk.p placed the asterisk in my central location and i have ip phones in all my branch i am using g729 codec(for testing). regards ramakrishnan.n __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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] maximum concurrent conference peers in asterisk
hi generally we describe the bandwidth in kilobits per second only. --- Matt Riddell [EMAIL PROTECTED] wrote: nr k wrote: Hi all I have a 2 mbps bandwidth in my central location and 64 kbps in all my branch office. what is the maximum concurrent conference peers possible through asterisk.p placed the asterisk in my central location and i have ip phones in all my branch i am using g729 codec(for testing). is the 64 kilo bits or kilo bytes? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
[Asterisk-Users] meetme conference getting error using codec g729
Hi all when i try to the conference the number i am getting the following error in asterisk console. i am using the g729 codec in asterisk and my sip devices but i can able make the call between the device. error: Nov 7 16:07:49 NOTICE[3190]: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729 Nov 7 16:07:49 WARNING[3190]: file.c:787 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
[Asterisk-Users] ad hoc conferencing-reg
Hi all How to configure adhoc conferencing in asterisk for sip phones.pls give me if any document for that. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
[Asterisk-Users] how to conferencd in Asterisk
Hi all How ro enable conference in asterisk and also how to make call between sccp device and sip device is there any special config in asterisk. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
[Asterisk-Users] Meetme Conference-reg
Hi all I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 extension.conf [default] exten = 1234,1,Meetme(1234) pls do the needful.. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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] Meetme Conference-reg
Hi I configured the meetme number in the area where i specified the other extensions but still i am having pbm. herewith i am sending the error i got in the asterisk console. Nov 6 19:07:35 WARNING[4952]: chan_zap.c:770 zt_open: Unable to open '/dev/zap/pseudo': No such device or address Nov 6 19:07:35 ERROR[4952]: chan_zap.c:6731 chandup: Unable to dup channel: No such device or address Nov 6 19:07:35 WARNING[4952]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Nov 6 19:07:35 WARNING[4952]: app_meetme.c:230 build_conf: Unable to open pseudo device regards ramakrishnan.n --- Rich Adamson [EMAIL PROTECTED] wrote: I am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234 and also i add the extension 1234 in extension.conf.if i call to 1234 asterisk says it's invalid conference number. i am having both sccp and sip devices. [room] ; Usage is conf = confno[,pin] conf = 1234 I assume you put the above in meetme.conf file? extension.conf [default] exten = 1234,1,Meetme(1234) Is the [default] section of extensions.conf where all of your other extensions are defined? If not, move the above entry to whatever section you have your other extensions defined. Then stop and restart asterisk. If the above doesn't address your issue, then copy/paste the CLI stuff so we can see what it is telling you. ___ --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 __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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
[Asterisk-Users] meetme conference pbm using g723.1 codec
Hi all i am having Asterisk 1.0.9. now i configured the meetme conference with conference number 1234.I have both sccp ande sip device.if i use the codec ulaw i can able make call between sip and sccp devices and also put meetme conference.if i use g.723.1 codec i have pbm in conference and call between sip and sccp devices.how to solve this pbm.pls do the needful... regards ramakrishnan.n __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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
[Asterisk-Users] how to configure adhoc conference in Asterisk
Hi all how to configure adhoc conference in asterisk. regards ramakrishnan.n __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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
[Asterisk-Users] How to configure Asterisk through webmin
Hi all I configured asterisk and webmin.i dont know how to integrate webmin with asterisk and how to access asterisk through webmin.pls do the needful. regards ramakrishnan.n __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --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
[Asterisk-Users] Call Disconnect problem
Hi all I have configured Asterisk call manager and i conneted 2 cisco ata 186 (SCCP).I make call between the ata's through Asterisk.the phones are perfectly registered with asterisk i am able to make calls but the call not disconnected after hangup and also i got an error msg RECEIVE MESSAGE TYPE UNKNOWN; 26 regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --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
