[asterisk-users] CDR in case of CallForwarding

Hello users, i am looking for a solution in terms of CDR for the outbound only call. presently i have the following setup. //extensions.conf [from-outside] exten = _X.,1,NoOp(IncomingCall) exten = _X.,n,BackGround(choce.wav) exten = _X.,n,WaitExten(5) exten = _X.,n,Hangup exten =

[asterisk-users] Dialing OutBound SIP trunk using Dial() command

Hello users, i am working on directly calling the numbers from the sip provider of my choice from asterisk using Dial command as follows. extensions.conf [dial-out] exten = _XX,1,NoOp(Dialing out) exten = _XX,n,Dial(SIP/1{EXTEN}:password:md5secret:authname:tarnsp...@host:port ,

[asterisk-users] Free FaxForAsterisk ReceiveFAX not working

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes

[asterisk-users] Asterisk Manager API - DTMF issues

Hello users, i have been testing the DTMF tone detection using originate command both from Asterisk CLI and java API. but my DTMF entry at the originate user is not getting detected by the asterisk in both the cases what i should do to make it work any help will be appreciated. my versions

[asterisk-users] SkypeForAsterisk

Hello users, i am planning to forward my skype calls from skype to the asterisk registerd skype. The scenario is as follows. i)SkypeUserA calls SkypeUserB ii)SkypeUserB forwards his calls to SkypeUserC iii)SkypeUserC sees he got call from SkypeUserA. do i have a way to extract the

[asterisk-users] Gtalk Asterisk integration

Hello users, I am trying to integrate asterisk and gtalk. my configuration is as follows OS:centos asterisk-1.6.0 asterisk-addons-1.6.0 dahdi-linux-2.2 dahdi-tools-2.2 libpri-1.4 share iksemel-1.2 #/etc/asterisk/jabber.conf [general] debug=yes autoprune=no autoregister=no [google]

[asterisk-users] Swift application and DTMF

the following in my asterisk CLI app_swift.c:453 engine:DTMF=7 --when i press 7 app_swift.c:482 engine: No DTMF --when i press #,0,* Please help me out Thanks in advance srinivas antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] X100P FXO PCI card not receiving calls

. can anybody advise?? Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list

[asterisk-users] DUNDi Errors (ENCREJ)

- To resolve this i tried to remove all keys in all servers and once again created and distributed the loaded in each system with keys init command but stilll i am getting the same error can anybody help me out??? Thanks and regards srinivas antarvedi

[asterisk-users] Problems with ReceiveFAX (asterisk 1.6.0.3 and t38)

completely(3rd attempt also fails) i have the tcpdump's .cap files so if anybody want to look at them too i can send.i tried to send along with this mail but the mail was rejected may be because of exceeding the attachment size. Any help is appreciable Thanks and regards Srinivas Antarvedi

[asterisk-users] FaxIn problems

be the problem??? how to debug this issues with the fax in asterisk CLI?? any help is appreciated.. Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

[asterisk-users] How to bind a SIP channel to an IP

defaultip=192.168.0.15 in this case i dont have any problems and it was working fine... can anybody helpme out to bind the phones to a particular ip if not is it possible to do at all just give me a hint so that i will work on Thanks in advances Srinivas Antarvedi

[asterisk-users] MeetMeAdmin() working problem

suggestions thanks in advance regards srinivas antarvedi Srinivas Antarvedi ___ -- 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

[asterisk-users] Listen And Talk mode differentiation of meetme() conference

Hello users, i am trying to setup a conference system and i have following requirement 1)some users are only in listen mode 2)some users are only in talk mode 3)some users are able to do both talk and listen how to diffrentiate them when they enter into a particular mode? meaning do i have to

[asterisk-users] MeetMeAdmin() working problem

,unmute,kick(1,2,3 options) not working and the CLI showing specified user not found can anybody helpme out not using any zaptel drivers using only ztdummy Thanks in advance Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] chan_sip.c:2918 auto congestion

be the problem? give a hint Thanks and regards Srinivas Antarvedi ___ -- 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] Failure of Sending Voicemail As an attachment in E-mail

account.. What should i do ? Actually my company is using a third party email server.. Just give me a hint Thanks in advance for your reply Regards Srinivas Antarvedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

[asterisk-users] service provider connection problem

ideas to this solution thanks and regards srinivas antarvedi ___ --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] Asterisk.NET API --help required

,orginate sucess event , originate failure event can anybody give me a hint so that i can proceed further thanks in advance for the kind suggestions. regards srinivas antarvedi -- Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http

[asterisk-users] Text-To-Speech synthesizer--help required

srinivas Antarvedi ___ --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

[asterisk-users] Dial application response code--help required

is correct? just correct me if i am wrong and what about the case 1 (404 Not Found)? Thanks and regards srinivas antarvedi -- Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] MediaHandling--Help Required

which is equal to Session-Timeout value from radius? Can anybody have any idea of handling network problem of his type? Looking forward for suggestions Thanks in advance srinivas antarvedi ___ --Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Users location --help required

can above fetching mechanism from openser to asterisk using database views be possible? Thanks in advance Srinivas Antarvedi ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update

[asterisk-users] Realtime Agents.conf

Hello, i have a small setup which requries that agents should be added dynamically, means their usernames and passwords using a database (MySql). can anybody have idea please give me a hint thanks in advance -- Srinivas Antarvedi ___ --Bandwidth

[asterisk-users] voicemail advanced options problem with mysql datbase

to send a reply to the person who sent me the message using advanced options no1 Can anybody plaease help me out? Thanks in advace Srinivas Antarvedi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] voicemail advanced options problem with mysql datbase

to send a reply to the person who sent me the message using advanced options no1 Can anybody plaease help me out? Thanks in advace Srinivas Antarvedi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

[asterisk-users] Dialout option problem in voicemail.conf

! plz help me out ! thanks in advance srinivas antarvedi ___ --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] problem with dialout option in voicemail.conf

hello all, i have a small setup in my office which can just send voicemails and retrive them on a LAN now we wanted to go for a nat with the 2 different contexts with entirely different environement the problem i have faced is: when one of the local guy leaves a message i can call him back