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
Re: [asterisk-users] can't seem to register, status unmonitored
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;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 -- _ -- 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
Re: [asterisk-users] can't seem to register, status unmonitored
On Tue, 15 Jun 2010, nikhil singhania wrote: 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. 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?? You cannot execute an AGI that executes dial() from the command line. You can debug an AGI from the command line by feeding the proper responses into STDIN, but it cannot interact with the running instance of Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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
