[asterisk-users] PSTN-IVR call

hi all, I am calling a PSTN and trying to transfer it to another asterisk server through exec_dial function. $agi-exec_dial(SIP,2001:j0...@172.26.48.62:5060,NULL,NULL,NULL); Though this is the function written by me in a file inbound.php which is called when an extension is dialled. When ever

[asterisk-users] asterisk registration

Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten = 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of

[asterisk-users] Fwd: asterisk registration

-- Forwarded message -- From: nikhil singhania niksingha...@gmail.com Date: 10 June 2010 14:08 Subject: asterisk registration To: asterisk-users@lists.digium.com Cc: Ma Hu Ma anshumishra6...@gmail.com Hi all, I think i understand the problem, actually I have two asterisk server

[asterisk-users] calling peer from server

Hi everybody, This is the console output of the asterisk server. debian-te410*CLI sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have

[asterisk-users] calling peer from server

Thanks for the reply. Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i

[asterisk-users] can't seem to register, status unmonitored

Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2...@172.26.48.208 sip%3a2...@172.26.48.208;expires=3013 208 is ip

Re: [asterisk-users] can't seem to register, status unmonitored

-15 4:11 AM, nikhil singhania niksingha...@gmail.com wrote: Hi everybody, I am trying to register my softphone(twinkle) on an asterisk server. Everything seems to be fine. Here is the output on show registrations in twinkle: Tue 18:57:51 nikhil: you have the following registrations sip:2

[asterisk-users] Fwd: can't seem to register, status unmonitored

-- Forwarded message -- From: nikhil singhania niksingha...@gmail.com Date: 16 June 2010 12:15 Subject: Re: [asterisk-users] can't seem to register, status unmonitored To: Zeeshan Zakaria zisha...@gmail.com Here is my extensions.conf: [general] static=yes ; default

[asterisk-users] writing echo in inbound file

Thanx for the reply. The reason i wrote echo is, i was running the script on the command line, and i wanted to see if the particular function is running. Just like i do debugging in c++. I didn't know that it sends messages to asterisk. But again i was not able to see any message on asterisk

Re: [asterisk-users] can't seem to register, status unmonitored

, Zeeshan Zakaria zisha...@gmail.com wrote: you should post this to the list, not to my personal email. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-16 2:45 AM, nikhil singhania niksingha...@gmail.com wrote: Here is my extensions.conf: [general] static=yes ; default

[asterisk-users] calling machine over sip

Actually my problem is not related to sip.conf and extensions.conf. I have used only standard files from martin pdf which are given as example. I am able to call some system connected over LAN, when each has a softphone and are registered on a asterisk server. But now what i want is instead of

[asterisk-users] device or sound card busy

Hi guys, Thanx a lot to all of you. My call is now forwarded to sip form PSTN, but again a new problem is coming. When i pick up the call from my softphone it says the can not access speaker or microphone. But i have my headphone plugged in and in working stage. on softphone: Fri 18:08:17

[asterisk-users] playing file when using call file in /var/spool/asterisk/outgoing in asterisk

HI everybody, Thanks for your support till now. I am using call files to initiate call from asterisk to twinkle(softphone) over LAN, the call file is generated and moved using a php script to the location /var/spool/asterisk/outgoing. I want to know while we are calling are there some

[asterisk-users] using call file

HI list-users, Greetings!! I have been using call file, i playback my file using * application:playback* and once the playback is over the call is disconnected. Is there any way it can wait and also record the dtmf inputs once the playback is over. Thanks in advace Nikhil Kumar summer

[asterisk-users] storing DTMF inputs

Thanks a lot Danny. I have done the part of playing a file by creating a context in my dialplan. Now I am puzzled as i wish to store the DTMF inputs done by the users who is listening to the playback. I found there are ways, but some specific way by which it is not stored in file but conveyed