[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 directly to the asterisk server. When the call landed up on the softphone, i pressed keys the softphone detects pressing of the keys but how the server will know which key is pressed and CLI shows no such message of key pressing. Is it supposed to show the message?? There may be other ways too, what ever would be implemented easily. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 functions which we can use , to play some file or actually control this call. I have to play some file and get the user response sitting on the server itself. Please can anyone help!!! -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Warning: Failed to open sound card: Device or resource busy this message is displayed and on CLI -- SIP/2001-081fa758 is ringing -- Got SIP response 480 User not responding back from 172.26.48.113 -- SIP/2001-081fa758 is circuit-busy is displayed, even though i pick up, i can hear the ringing tone in the phone which called and not able to talk . What may be the problem. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 server. One thing, it must be possible to run the php script file on command line, since it was the script which got executed when a particular no. was called. So now instead of calling the no. i am executing the script on command line. I wrote my problem, i get the ip of other machine on my asterisk server saying it has registered. But when i write the command to call the machine in the php script, the command just executed and moves to next command but call is not recieved. Here is the command line output at the server: debian-te410:/home/simmortel/www/wizoz# ./inbound1.php Hello, world!Hello, world!/var/www/wizoz/Hello, world! createdEXEC Dial SIP/wlg-gateway a STREAM FILE /var/www/wizoz/prompts/welcome 123 0 a STREAM FILE /var/www/wizoz/prompts/welcome 123 0 a Hello, world!5STREAM FILE /var/www/wizoz/prompts/record 123 0 a RECORD FILE /var/www/wizoz/wav/1276758826 wav 0 6 BEEP s=5 a STREAM FILE /var/www/wizoz/prompts/messagesent 123 0 a STREAM FILE /var/www/wizoz/prompts/thankyou 123 0 a I have input a, because it shows ouput of any command when i enter any character exept carriage return..though this seems a foolishness. I don't know how to do this. You can also see hello world echoed on output. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't seem to register, status unmonitored
Thanx Zeeshan,
I forgot to thank you , doing qualify=yes shows the status and its active.
1
Name/username HostDyn Nat ACL Port Status
wlg-gateway202.7.4.40 5060 Unmonitored
2002/2002 (Unspecified)D N 0Unmonitored
2001/2001 172.26.48.113D N 5061 OK (1 ms)
3 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 1
offline]
2And yes i didn't know that about 'sip show registry'.
3And I am still stuck with the 3rd problem.
Can you just tell me in the above output on the asterisk server, if i have
to call the user 2...@172.26.48.113, through a php script and not softphone.
Because my sofphone can call it.
This is very silly problem . Please rescue me. status is Ok and online.
i posted the last files to the list also.
On 16 June 2010 18:58, 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 values for changes to this file
writeprotect=no ; by the Asterisk CLI
[globals]
; variables go here
[default]
; default context
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
; Let them know what's going on
exten = 500,3,Echo
; Do the echo test
exten = 500,4,Playback(demo-echodone)
; Let them know it's over
exten = 500,5,Hangup
exten = _.,1,Dial(SIP/${ext...@wlg-gateway); match anything and
send to wlg-gateway
exten = _.,2,Hangup
[from-wlg-gateway]
; context for calls coming from wlg-gateway
exten = 4980007,1,Dial(SIP/2001SIP/2002)
exten = _.,1,Congestion()
; everyone else gets congestion
..
sip.conf
[general]
context=default ; Default context for incoming calls
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
[2001]
type=friend ; both send and receive calls from this peer
host=dynamic ; this peer will register with us
username=2001
secret=j0nny
canreinvite=no ; don't send SIP re-invites (ie. terminate rtp stream)
nat=yes ; always assume peer is behind a NAT
context=phones ; send calls to 'phones' context
dtmfmode=rfc2833 ; set dtmf relay mode
allow=all; allow all codecs
[2002]
type=friend
host=dynamic
username=2002
secret=whyfry
canreinvite=no
nat=yes
context=phones
dtmfmode=rfc2833
allow=all
[wlg-gateway]
type=friend
disallow=all
allow=ulaw
context=from-wlg-gateway
host=202.7.4.40
canreinvite=no
dtmfmode=rfc2833
allow=all
.
inbound.php
..
#!/usr/bin/php
?php
ob_implicit_flush(true);
set_time_limit(0);
echo(Hello, world!);
require_once phpagi.php;
error_reporting(E_ALL);
echo(Hello, world!);
$dir_base = /var/www/wizoz/;
echo $dir_base;
$dir_prompt = $dir_base.prompts;
$dir_wav = $dir_base.wav;
$rel_dir_mp3 = mp3;
$dir_mp3 = $dir_base.$rel_dir_mp3;
$agi = new AGI();
echo(created);
$agi-answer();
$agi-exec_dial(SIP,2002);
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
echo(Hello, world!);
?
..
Though I am new, but i am somewhat familiar, and am devoting a great deal
of time. Now you have all the files. I highlited the exec_dial function.
This inbound.php is the file i am executing on the command line on the
server. But I am not gettting the call at my end. May be the way i am doing
it is wrong. Please suggest me. Rest of the code works fine.
On 15 June 2010 18:15, Zeeshan Zakaria zisha...@gmail.com wrote:
The r...
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiit...
--
Nikhil Kumar
summer intern:simmortel voice technologies
rit2007033
b.tech IT 6th sem
IIIT Allahabad
cont...@9793905858
email: rit2007...@iiita.ac.in
niksingha...@gmail.com
http://profile.iiita.ac.in/RIT2007033/
--
_
-- Bandwidth and Colocation Provided
[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 using the softphone I write a function in my file which will be executed when the call is placed. In that file i wrote $agi=new AGI(); $agi-exec_dial(SIP,2002,NULL, NULL,NULL); I have used the exec_dial function found in phpagi.php . It is built above basic dial function. Here 'agi' is an instance of class AGI, which has a method exec_dial. When i execute the php file, over command line on my unix machine, I am expecting a call on the softphone which I have registered on the asterisk server. For ip clarification: all have static ip: server ip:172.26.48.208:5060 i have configured twinkle as a softphone client on 172.26.48.113:5061, since on the asterisk server cli when i use 'sip debug set peer 2002' it shows the registered ip, as i said earlier, but again on 'sip show registry' no value is displayed. I don't know what's going on. In the softphone I give domin information as the ip of the asterisk server i.e. 172.26.48.208. In the softphone it shows registration successful. Thanks in advance -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 values for changes to this file
writeprotect=no ; by the Asterisk CLI
[globals]
; variables go here
[default]
; default context
[phones]
; context for our phones
exten = 2001,1,Dial(SIP/2001)
exten = 2002,1,Dial(SIP/2002)
exten = 500,1,Answer()
exten = 500,2,Playback(demo-echotest)
; Let them know what's going on
exten = 500,3,Echo
; Do the echo test
exten = 500,4,Playback(demo-echodone)
; Let them know it's over
exten = 500,5,Hangup
exten = _.,1,Dial(SIP/${ext...@wlg-gateway); match anything and
send to wlg-gateway
exten = _.,2,Hangup
[from-wlg-gateway]
; context for calls coming from wlg-gateway
exten = 4980007,1,Dial(SIP/2001SIP/2002)
exten = _.,1,Congestion()
; everyone else gets congestion
..
sip.conf
[general]
context=default ; Default context for incoming calls
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes; Enable DNS SRV lookups on outbound calls
[2001]
type=friend ; both send and receive calls from this peer
host=dynamic ; this peer will register with us
username=2001
secret=j0nny
canreinvite=no ; don't send SIP re-invites (ie. terminate rtp stream)
nat=yes ; always assume peer is behind a NAT
context=phones ; send calls to 'phones' context
dtmfmode=rfc2833 ; set dtmf relay mode
allow=all; allow all codecs
[2002]
type=friend
host=dynamic
username=2002
secret=whyfry
canreinvite=no
nat=yes
context=phones
dtmfmode=rfc2833
allow=all
[wlg-gateway]
type=friend
disallow=all
allow=ulaw
context=from-wlg-gateway
host=202.7.4.40
canreinvite=no
dtmfmode=rfc2833
allow=all
.
inbound.php
..
#!/usr/bin/php
?php
ob_implicit_flush(true);
set_time_limit(0);
echo(Hello, world!);
require_once phpagi.php;
error_reporting(E_ALL);
echo(Hello, world!);
$dir_base = /var/www/wizoz/;
echo $dir_base;
$dir_prompt = $dir_base.prompts;
$dir_wav = $dir_base.wav;
$rel_dir_mp3 = mp3;
$dir_mp3 = $dir_base.$rel_dir_mp3;
$agi = new AGI();
echo(created);
$agi-answer();
$agi-exec_dial(SIP,2002);
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
# welcome to yumphone.com
$agi-stream_file($dir_prompt.'/welcome','123'); fflush($agi-out);
echo(Hello, world!);
$result = $agi-get_variable(CALLERID(num));
echo $result;
$phonenum = $result['data'];
if (strlen($phonenum) != '10')
{
$phonenum = substr($phonenum,-10);
}
$uid = $phonenum.time();
$agi-stream_file($dir_prompt.'/record','123'); fflush($agi-out);
# please record your message after the beep. press 0 at the end of the
message
$agi-record_file($dir_wav./.$uid,'wav','0','6',NULL,true,5);
# fname, format, escape, timeout, offset, beep, silence
$agi-stream_file($dir_prompt.'/messagesent','123'); fflush($agi-out);
# your message has been sent
$agi-stream_file($dir_prompt.'/thankyou','123'); fflush($agi-out);
# thank you
?
..
Though I am new, but i am somewhat familiar, and am devoting a great deal of
time. Now you have all the files. I highlited the exec_dial function. This
inbound.php is the file i am executing on the command line on the server.
But I am not gettting the call at my end. May be the way i am doing it is
wrong. Please suggest me. Rest of the code works fine.
On 15 June 2010 18:15, Zeeshan Zakaria zisha...@gmail.com wrote:
The reason I said it'll take you one week, is because you seem new to
asterisk. It may take even more.
Pasting a part of the code is not enough for anybody to be able to help
you. You should paste the relevant parts of your sip.conf, extensions.conf
and the agi script. To me it seems you are new to dial plans, and if this is
true, first you need to focus on understanding dial plans, and then jump to
agi.
Did the other two issue get resolved?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-06-15 7:49 AM, nikhil singhania niksingha...@gmail.com wrote:
Hi Zeeshan,
Thanx for ur reply!!
The reason for this question was that i am actually doing the 3rd part,
which you said will take me 1 week to learn.
I have modified
[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 of the asterisk server. on the server on doing 'sip show peers' , it shows the user and the ip but status is unmonitored. debian-te410*CLI sip show peers Name/username HostDyn Nat ACL Port Status wlg-gateway202.7.4.40 5060 Unmonitored 2002/2002 (Unspecified)D N 0Unmonitored 2001/2001 172.26.48.113D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI sip show registry HostUsername Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't seem to register, status unmonitored
Hi Zeeshan, Thanx for ur reply!! The reason for this question was that i am actually doing the 3rd part, which you said will take me 1 week to learn. I have modified a file inbound.php which uses function of phpagi.phpexec_dial. But since i am not able to get the call on softphone. Here is part of code: $agi = new AGI(); $agi-answer(); $agi-exec_dial(SIP,2001); when i execute the php file on the command line of server, nothing happens in my softphone. Since it's registered as i told you then when the file is executed at server, my phone is supposed to ring , but its not ringing. Where I am going wrong?? Message: 19 Date: Tue, 15 Jun 2010 07:01:43 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] can't seem to register, status unmonitored To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlktil6aaf21hcg4jpf7sv9yzpja7w-yo8st6ppf...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 1. Do 'qualify=yes' in sip.conf for the extenstions for which you want to see the status. 2. 'sip show registry' doesn't show anything for the extensions registering on your server, it shows your server registering on another server, i.e. when when setting up a trunk. 3. Using php to make a call, you need to dedicate some time (probably a week) for learning AGI using phpagi. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-15 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...@172.26.48.208 sip%3a2...@172.26.48.208 sip%3a2...@172.26.48.208 sip%253a2...@172.26.48.208;expires=3013 208 is ip of the asterisk server. on the server on doing 'sip show peers' , it shows the user and the ip but status is unmonitored. debian-te410*CLI sip show peers Name/username HostDyn Nat ACL Port Status wlg-gateway202.7.4.40 5060 Unmonitored 2002/2002 (Unspecified)D N 0Unmonitored 2001/2001 172.26.48.113D N 5062 Unmonitored 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 1 offline] 113 is my ip. This may be the reason that when i do 'sip show registry' no value is displayed even though i get message of successful registration on my sofphone. debian-te410*CLI sip show registry HostUsername Refresh State Reg.Time Please help, what may be the problem here, should the status be different? I want to make a call from server to the 2001 user through a php file, how can I do so?? Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100615/79ebe9fb/attachment.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 71, Issue 33 ** -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have tried diff ways but can't seem to get it work. Can please some one suggest me anything in this regard. -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 want is instead of using the softphone I write a function in my file which will be executed when the call is placed. In that file i wrote $agi=new AGI(); $agi-exec_dial(SIP,2002,NULL,NULL,NULL); I have used the exec_dial function found in phpagi.php . It is built above basic dial function. Here 'agi' is an instance of class AGI, which has a method exec_dial. When i execute the php file, over command line on my unix machine, I am expecting a call on the softphone which I have registered on the asterisk server. For ip clarification: all have static ip: server ip:172.26.48.208:5060 i have configured twinkle as a softphone client on 172.26.48.113:5061, since on the asterisk server cli when i use 'sip debug set peer 2002' it shows the registered ip, as i said earlier, but again on 'sip show registry' no value is displayed. I don't know what's going on. In the softphone I give domin information as the ip of the asterisk server i.e. 172.26.48.208. In the softphone it shows registration successful. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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. 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 inbound.php which contains the IVR script to be executed. Now what i want is that through this inbound.php , i should be able to call another asterisk server, where I have also configured twinkle as a softphone. The problems: --I am not able to register this softphone on the previous asterisk server as user 2001, though i modified the server's extension and sip file to include the user 2001 under [phones] context. ---cli chan_sip.c:15839 handle_request_register: Registration from 'user1 sip:2...@172.26.48.208 sip%3a2...@172.26.48.208' failed for '172.26.48.62' - No matching peer found shows this error upon registration.. --at my server it shows 3 unmonitored peers, but the previous server doesn't show any peers on sip show peers..though i have added all three users in sip file, and yes reloaded the dial plan. WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001 [Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) is the error when i do not give ip..assuming 2001 to be registered at the server. when i give the ip of my server.. chan_sip.c:20039 handle_request_invite: Call from '' to extension '2001' rejected because extension not found. is the error..call actually lands up on asterisk server but it shows the above error and ofcourse can not be recieved with softphone. Please help me out in this regard. Though above details may be confusing..I have tried to briefly write in case any more explanation needed, please mail me.I am stuck in this so please help. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 inbound.php which contains the IVR script to be executed. Now what i want is that through this inbound.php , i should be able to call another asterisk server, where I have also configured twinkle as a softphone. The problems: --I am not able to register this softphone on the previous asterisk server as user 2001, though i modified the server's extension and sip file to include the user 2001 under [phones] context. ---cli chan_sip.c:15839 handle_request_register: Registration from 'user1 sip:2...@172.26.48.208 sip%3a2...@172.26.48.208' failed for '172.26.48.62' - No matching peer found shows this error upon registration.. --at my server it shows 3 unmonitored peers, but the previous server doesn't show any peers on sip show peers..though i have added all three users in sip file, and yes reloaded the dial plan. WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001 [Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) is the error when i do not give ip..assuming 2001 to be registered at the server. when i give the ip of my server.. chan_sip.c:20039 handle_request_invite: Call from '' to extension '2001' rejected because extension not found. is the error..call actually lands up on asterisk server but it shows the above error and ofcourse can not be recieved with softphone. Please help me out in this regard. Though above details may be confusing..I have tried to briefly write in case any more explanation needed, please mail me.I am stuck in this so please help. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 i dial through PSTN it gives beeps sound, but without this line program runs smoothly. Can someone help??? -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- Nikhil Kumar rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
