Noone is in your session: 8600051

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Noone is in your session: 8600051

Postby asterisk_at_my_risk » Wed Sep 13, 2006 10:34 am

Noone is in your session: 8600051
Go Back

Call Agent Again

whenever we login to http://x.x.x.x/agc/vicidial.php we gert the above message please help mw what is the reason for the above message


i am also attaching the output of my screen -r output lemme k\me know if you need any other info from my end



006-09-13 20:44:23|| lagged call vac agent DELETED 0|2006-09-13 20:42:23||

loop counter: |5759992|
TIME DEBUG: -5|-5|0| GMT: 01:44
2006-09-13 20:44:26|LIVE AGENTS LOGGED IN: 0|
2006-09-13 20:44:26| : agents: 1 dial_level: 0|
2006-09-13 20:44:26| : Calls to place: 0 (0 - 0) |
2006-09-13 20:44:26| : CALLING|
TIME DEBUG: -5|-5|0| GMT: 01:44
2006-09-13 20:44:26|| lagged call vla agent PAUSED 0|20060913204356|20060913204416|20060913204426||
2006-09-13 20:44:26|| lagged call vac agent DELETED 0|2006-09-13 20:42:26||

loop counter: |5759991|
TIME DEBUG: -5|-5|0| GMT: 01:44
2006-09-13 20:44:28|LIVE AGENTS LOGGED IN: 0|
2006-09-13 20:44:28| : agents: 1 dial_level: 0|
2006-09-13 20:44:28| : Calls to place: 0 (0 - 0) |
2006-09-13 20:44:28| : CALLING|
TIME DEBUG: -5|-5|0| GMT: 01:44
2006-09-13 20:44:28|| lagged call vla agent PAUSED 0|20060913204358|20060913204418|20060913204428||
2006-09-13 20:44:28|| lagged call vac agent DELETED 0|2006-09-13 20:42:28||

loop counter: |5759990|
2006-09-13 20:44:28|| lagged call vac agent DELETED 0|2006-09-13 20:34:28|LIVE||
2006-09-13 20:44:28|| updating server parameters 96|8365|-5|default||
TIME DEBUG: -5|-5|0| GMT: 01:44

PROCESS KILLED NO LISTENER RUNNING... EXITING

checking to see if listener is dead |0|
DONE... Exiting... Goodbye... See you later... Not really, initiating next loop...
2006-09-13 20:44:31|HANGING UP||
2006-09-13 20:44:31|CLOSING DB CONNECTION||
DONE... Exiting... Goodbye... See you later... Really I mean it this time

[screen is terminating]
[/url]
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Wed Sep 13, 2006 11:47 am

What distro of Linux are you running?

Post the results of these MySQL queries while you are logged into vicidial.php:
SELECT * from live_channels;
SELECT * from live_sip_channels;

Post the output of this command:
/bin/ps -o "%p %a" --no-headers -A
mflorell
Site Admin
 
Posts: 18384
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

here are the output you wanted THNX

Postby asterisk_at_my_risk » Wed Sep 13, 2006 12:38 pm

output for /bin/ps -o "%p %a" --no-headers -A

[root@sip /]# /bin/ps -o "%p %a" --no-headers -A
1 init [3]
2 [ksoftirqd/0]
3 [watchdog/0]
4 [events/0]
5 [khelper]
6 [kthread]
8 [kacpid]
96 [kblockd/0]
148 [pdflush]
149 [pdflush]
151 [aio/0]
150 [kswapd0]
99 [khubd]
238 [kseriod]
389 [kjournald]
882 udevd
1677 syslogd -m 0
1679 klogd -x
1695 portmap
1712 rpc.statd
1724 auditd
1728 [kauditd]
1751 rpc.idmapd
1763 hcid: processing events
1767 sdpd
1786 [krfcommd]
1906 /usr/sbin/automount --timeout=60 /misc file /etc/auto.misc
1953 /usr/sbin/automount --timeout=60 /net program /etc/auto.net
1964 nifd -n
1993 mDNSResponder
2001 /usr/sbin/acpid
2009 cupsd
2086 /usr/sbin/sshd
2102 sendmail: accepting connections
2108 sendmail: Queue runner@01:00:00 for /var/spool/clientmqueue
2116 gpm -m /dev/input/mice -t imps2
2123 crond
2142 xfs -droppriv -daemon
2155 /usr/sbin/atd
2164 dbus-daemon --system
2173 cups-config-daemon
2190 hald --retain-privileges
2195 hald-addon-acpi
2221 hald-addon-storage
2244 /usr/sbin/ntpd
2245 /bin/sh /usr/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache
2294 /usr/libexec/mysqld --basedir=/usr --datadir=/var/lib/mysql --user=mysql --pid-file=/var/run/mysqld/mysqld.pid --skip-l
2320 /usr/sbin/ntpd
2344 /usr/sbin/httpd -k start
2414 /usr/sbin/httpd -k start
2415 /usr/sbin/httpd -k start
2416 /usr/sbin/httpd -k start
2417 /usr/sbin/httpd -k start
2418 /usr/sbin/httpd -k start
2419 /usr/sbin/httpd -k start
2424 /usr/sbin/httpd -k start
2425 /usr/sbin/httpd -k start
2507 /sbin/mingetty tty2
2508 /sbin/mingetty tty3
2509 /sbin/mingetty tty4
2510 /sbin/mingetty tty5
2511 /sbin/mingetty tty6
3975 /sbin/mingetty tty1
5278 SCREEN
5279 /bin/bash
5352 /bin/bash
5656 SCREEN
5657 /bin/bash
8496 asterisk -cccccgv
12056 rasterisk r
10774 sshd: root@pts/5
10864 -bash
13412 /usr/sbin/httpd -k start
13521 screen -r
22345 sshd: root@pts/7
22478 -bash
24588 crond
24596 crond
24605 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
24616 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
24697 /usr/bin/SCREEN -d -m -S ASTVDauto /home/cron/AST_VDauto_dial.pl
24698 /usr/bin/perl /home/cron/AST_VDauto_dial.pl
24705 /usr/bin/SCREEN -d -m -S ASTVDremote /home/cron/AST_VDremote_agents.pl
24706 /usr/bin/perl /home/cron/AST_VDremote_agents.pl
24721 /usr/sbin/atd
24725 /usr/sbin/atd
24728 sh
24729 /bin/sh
24730 /usr/bin/perl /home/cron/ADMIN_keepalive_AST_send_listen.pl
24739 sh
24740 /bin/sh
24741 /usr/bin/perl /home/cron/ADMIN_keepalive_AST_send_listen.pl
24745 /bin/ps -o %p %a --no-headers -A
[root@sip /]#
[root@sip /]#
[root@sip /]# /bin/ps -o "%p %a" --no-headers -A
1 init [3]
2 [ksoftirqd/0]
3 [watchdog/0]
4 [events/0]
5 [khelper]
6 [kthread]
8 [kacpid]
96 [kblockd/0]
148 [pdflush]
149 [pdflush]
151 [aio/0]
150 [kswapd0]
99 [khubd]
238 [kseriod]
389 [kjournald]
882 udevd
1677 syslogd -m 0
1679 klogd -x
1695 portmap
1712 rpc.statd
1724 auditd
1728 [kauditd]
1751 rpc.idmapd
1763 hcid: processing events
1767 sdpd
1786 [krfcommd]
1906 /usr/sbin/automount --timeout=60 /misc file /etc/auto.misc
1953 /usr/sbin/automount --timeout=60 /net program /etc/auto.net
1964 nifd -n
1993 mDNSResponder
2001 /usr/sbin/acpid
2009 cupsd
2086 /usr/sbin/sshd
2102 sendmail: accepting connections
2108 sendmail: Queue runner@01:00:00 for /var/spool/clientmqueue
2116 gpm -m /dev/input/mice -t imps2
2123 crond
2142 xfs -droppriv -daemon
2155 /usr/sbin/atd
2164 dbus-daemon --system
2173 cups-config-daemon
2190 hald --retain-privileges
2195 hald-addon-acpi
2221 hald-addon-storage
2244 /usr/sbin/ntpd
2245 /bin/sh /usr/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache
2294 /usr/libexec/mysqld --basedir=/usr --datadir=/var/lib/mysql --user=mysql --pid-file=/var/run/mysqld/mysqld.pid --skip-l
2320 /usr/sbin/ntpd
2344 /usr/sbin/httpd -k start
2414 /usr/sbin/httpd -k start
2415 /usr/sbin/httpd -k start
2416 /usr/sbin/httpd -k start
2417 /usr/sbin/httpd -k start
2418 /usr/sbin/httpd -k start
2419 /usr/sbin/httpd -k start
2424 /usr/sbin/httpd -k start
2425 /usr/sbin/httpd -k start
2507 /sbin/mingetty tty2
2508 /sbin/mingetty tty3
2509 /sbin/mingetty tty4
2510 /sbin/mingetty tty5
2511 /sbin/mingetty tty6
3975 /sbin/mingetty tty1
5278 SCREEN
5279 /bin/bash
5352 /bin/bash
5656 SCREEN
5657 /bin/bash
8496 asterisk -cccccgv
12056 rasterisk r
10774 sshd: root@pts/5
10864 -bash
13412 /usr/sbin/httpd -k start
13521 screen -r
22345 sshd: root@pts/7
22478 -bash
25179 crond
25188 crond
25203 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
25217 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
25298 /usr/bin/SCREEN -d -m -S ASTVDauto /home/cron/AST_VDauto_dial.pl
25299 /usr/bin/perl /home/cron/AST_VDauto_dial.pl
25485 /usr/sbin/atd
25489 /usr/sbin/atd
25491 sh
25492 /bin/sh
25493 /usr/bin/perl /home/cron/ADMIN_keepalive_AST_send_listen.pl
25495 sh
25496 /bin/sh
25497 /usr/bin/perl /home/cron/ADMIN_keepalive_AST_send_listen.pl
25504 /bin/ps -o %p %a --no-headers -A
You have new mail in /var/spool/mail/root
[root@sip /]# /bin/ps -o "%p %a" --no-headers -A

***********************************************************************************************************************'
output for
SELECT * from live_channels;
SELECT * from live_sip_channels;

mysql> select * from live_sip_channels;
Empty set (0.00 sec)

mysql> select * from live_channels;
Empty set (0.00 sec)

****************************************************************************************************************************
DISTRO of linux used by us
2.6.11-1.1369_FC4
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Wed Sep 13, 2006 12:42 pm

looks like your AST_update.pl script isn't running.

run this and post the output:
/home/cron/AST_update.pl --debug
mflorell
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Location: Florida

output for /home/cron/AST_update.pl --debug THNX

Postby asterisk_at_my_risk » Wed Sep 13, 2006 12:56 pm

[root@sip ~]# /home/cron/AST_update.pl --debug
LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='203.122.26.2 32'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='203.122.26. 232'|
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='203.122.26.2 32'|
3003
3005
3004
Zap Clients: |
IAX2 Clients: |
SIP Clients: |3003|3003@3003|3005|3005@3005|3004|3004@3004|
problem connecting to "localhost", port 5038: Connection refused at /home/cron/A ST_update.pl line 275
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Wed Sep 13, 2006 1:12 pm

Make sure you have configured your manager.conf as the SCRATCH_INSTALL specifies it and reload Asterisk
mflorell
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Posts: 18384
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

we have made some headway

Postby asterisk_at_my_risk » Wed Sep 13, 2006 2:11 pm

well !!!
there was a issue in manager.conf we corrected that and reloaded the asterisk

now are soft is also reciving incoming calls but that message is still coming

Noone is in your session: 8600051
Go Back

Call Agent Again

we are attaching all output for please let us know if you need any other info from our end

output for
SELECT * from live_channels;
SELECT * from live_sip_channels;




******************************************************************************************************************************
mysql> select * from live_sip_channels;
+-------------------+----------------+---------------+-----------+-------------- +
| channel | server_ip | channel_group | extension | channel_data |
+-------------------+----------------+---------------+-----------+-------------- +
| SIP/3004-09a81b70 | 203.122.26.232 | NULL | ring | SIP/ring |
+-------------------+----------------+---------------+-----------+-------------- +
1 row in set (0.00 sec)

mysql> select * from live_channels;
Empty set (0.00 sec)

mysql> select * from live_sip_channels;
+-------------------+----------------+---------------+-----------+--------------+
| channel | server_ip | channel_group | extension | channel_data |
+-------------------+----------------+---------------+-----------+--------------+
| SIP/3004-09acbc88 | 203.122.26.232 | NULL | ring | SIP/ring |
+-------------------+----------------+---------------+-----------+--------------+
1 row in set (0.00 sec)

mysql> select * from live_channels;
Empty set (0.00 sec)

mysql>

********************************************************************************************************************************

/home/cron/AST_update.pl --debug

|UPDATE server_updater set last_update='2006-09-14 0:33:30' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:30' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:31' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:31' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:32' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:32' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:32' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:33' where server_ip='203.122.26.232'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '203.122.26.232'|

|UPDATE server_updater set last_update='2006-09-14 0:33:33' where server_ip='203.122.26.232'|
concise: -1
************************************************************************************************************]
update for [root@sip /]# /bin/ps -o "%p %a" --no-headers -A
1 init [3]
2 [ksoftirqd/0]
3 [watchdog/0]
4 [events/0]
5 [khelper]
6 [kthread]
8 [kacpid]
96 [kblockd/0]
148 [pdflush]
149 [pdflush]
151 [aio/0]
150 [kswapd0]
99 [khubd]
238 [kseriod]
389 [kjournald]
882 udevd
1677 syslogd -m 0
1679 klogd -x
1695 portmap
1712 rpc.statd
1724 auditd
1728 [kauditd]
1751 rpc.idmapd
1763 hcid: processing events
1767 sdpd
1786 [krfcommd]
1906 /usr/sbin/automount --timeout=60 /misc file /etc/auto.misc
1953 /usr/sbin/automount --timeout=60 /net program /etc/auto.net
1964 nifd -n
1993 mDNSResponder
2001 /usr/sbin/acpid
2009 cupsd
2086 /usr/sbin/sshd
2102 sendmail: accepting connections
2108 sendmail: Queue runner@01:00:00 for /var/spool/clientmqueue
2116 gpm -m /dev/input/mice -t imps2
2123 crond
2142 xfs -droppriv -daemon
2155 /usr/sbin/atd
2164 dbus-daemon --system
2173 cups-config-daemon
2190 hald --retain-privileges
2195 hald-addon-acpi
2221 hald-addon-storage
2244 /usr/sbin/ntpd
2245 /bin/sh /usr/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache
2294 /usr/libexec/mysqld --basedir=/usr --datadir=/var/lib/mysql --user=mysql --pid-file=/var/run/mysqld/mysqld.pid --skip-l
2320 /usr/sbin/ntpd
2344 /usr/sbin/httpd -k start
2507 /sbin/mingetty tty2
2508 /sbin/mingetty tty3
2509 /sbin/mingetty tty4
2510 /sbin/mingetty tty5
2511 /sbin/mingetty tty6
3975 /sbin/mingetty tty1
5278 SCREEN
5279 /bin/bash
5352 /bin/bash
5656 SCREEN
5657 /bin/bash
8496 asterisk -cccccgv
12056 rasterisk r
10774 sshd: root@pts/5
10864 -bash
13412 /usr/sbin/httpd -k start
13521 screen -r
28038 /usr/sbin/httpd -k start
28051 /usr/sbin/httpd -k start
28052 /usr/sbin/httpd -k start
28066 /usr/sbin/httpd -k start
28067 /usr/sbin/httpd -k start
28068 /usr/sbin/httpd -k start
28069 /usr/sbin/httpd -k start
24475 sshd: root@pts/7
24729 -bash
27790 /usr/bin/SCREEN -d -m -S ASTVDauto /home/cron/AST_VDauto_dial.pl
27791 /usr/bin/perl /home/cron/AST_VDauto_dial.pl
27937 /usr/bin/SCREEN -d -m -S ASTlisten /home/cron/AST_manager_listen.pl
27938 /usr/bin/perl /home/cron/AST_manager_listen.pl
27944 /usr/bin/SCREEN -d -m -S ASTlisten /home/cron/AST_manager_listen.pl
27945 /usr/bin/perl /home/cron/AST_manager_listen.pl
27952 /usr/bin/SCREEN -d -m -S ASTsend /home/cron/AST_manager_send.pl
27953 /usr/bin/perl /home/cron/AST_manager_send.pl
28195 /usr/bin/SCREEN -d -m -S ASTVDremote /home/cron/AST_VDremote_agents.pl
28196 /usr/bin/perl /home/cron/AST_VDremote_agents.pl
28210 /usr/bin/SCREEN -d -m -S ASTupdate /home/cron/AST_update.pl
28211 /usr/bin/perl /home/cron/AST_update.pl
28300 /usr/bin/perl /home/cron/AST_update.pl --debug
30660 sshd: root@pts/12
30706 -bash
30817 rasterisk r
30865 sshd: cron [priv]
30897 sshd: cron@pts/13
30899 -bash
31003 top
1998 sshd: root@pts/14
2077 -bash
2207 sshd: root@pts/15
2285 -bash
2455 vi messages
2871 sshd: root@pts/16
2925 -bash
2967 mysql
3712 crond
3721 crond
3731 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
3746 /usr/bin/perl /home/cron/AST_manager_kill_hung_congested.pl
3866 /bin/ps -o %p %a --no-headers -A
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Wed Sep 13, 2006 2:42 pm

Did you apply the cli_chan_concise_delimiter.patch to Asterisk before you compiled it?
mflorell
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Location: Florida

yeah we applied the patch

Postby asterisk_at_my_risk » Thu Sep 14, 2006 4:54 am

i tried to confirm it by doing this........


[root@sip asterisk-1.2.10]# patch -p1 < ./cli_chan_concise_delimiter.patch
missing header for unified diff at line 3 of patch
can't find file to patch at input line 3
Perhaps you used the wrong -p or --strip option?
The text leading up to this was:
--------------------------
|--- cli-orig.c 2005-12-29 13:25:32.000000000 -0500
|+++ cli.c 2005-12-29 13:03:48.000000000 -0500
--------------------------
File to patch: cli.c
patching file cli.c
Reversed (or previously applied) patch detected! Assume -R? [n] n
Apply anyway? [n] n
Skipping patch.
1 out of 1 hunk ignored -- saving rejects to file cli.c.rej
asterisk_at_my_risk
 
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Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Thu Sep 14, 2006 6:19 am

what was your location when you ran that? (pwd)
mflorell
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Posts: 18384
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Location: Florida

location of the patch

Postby asterisk_at_my_risk » Thu Sep 14, 2006 6:48 am

[root@sip asterisk-1.2.10]# pwd
/usr/src/asterisk-1.2.10
[root@sip asterisk-1.2.10]#
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Thu Sep 14, 2006 6:50 am

OK, then the patch is applied, and after you make clean; make; make install does it change what shows up in the live_channels and live_sip_channels table at all when you connect?

If not, can you post the output of "show channels concise" in the Asterisk CLI?
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Posts: 18384
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

i made one Successful call !!!!

Postby asterisk_at_my_risk » Thu Sep 14, 2006 1:34 pm

well i was able to make one successful call but i am not sure of the exact way of dialing in one-call-at-a-time way of dialing

when i log in to http://x.x.x.x/agc/vicidial.php i immediadetly get a incoming call to my softphone now what should i do shall i answer that incoming call and then click on DIAL NEXT NUMBER or shall i disconnect
that incoming call and then click on DIAL NEXT NUMBER

and is it nessary to logged into http://x.x.x.x/agc/astguiclient.php
for it
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby mflorell » Thu Sep 14, 2006 1:37 pm

Have you read the agent manual at eflo.net?

you do not need to login to astguiclient.php.
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HELP OUT

Postby asterisk_at_my_risk » Fri Sep 15, 2006 8:25 am

Hello

we read the manual carefuly now but there still seems to be a issue
it still giving us us that message "noone is in your session 8600051"

just to to inform that right now we are not having any ZAP card we are using ZTDUMMY for timing thing
we are attaching some output at the time we were trying to make the call
please have a look and let us knoqw where are we going wrong
and let us know if you want any other info from our end


****************************************************************************************************************************

sip*CLI>
<-- SIP read from 203.122.26.234:47318:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:61854:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Sep 15 18:43:56 NOTICE[24359]: chan_local.c:498 local_alloc: No such extension/c ontext 8600051@default creating local channel
Sep 15 18:43:56 NOTICE[24359]: channel.c:2445 __ast_request_and_dial: Unable to request channel Local/8600051@default/n
sip*CLI>
<-- SIP read from 203.122.26.234:47318:


--- (0 headers 0 lines) Nat keepalive ---
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:61854:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Reliably Transmitting (NAT) to 203.122.26.234:47318:
CANCEL sip:3003@203.122.26.234:47318 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport
From: "S0609151843338600051" <sip:asterisk@203.122.26.232>;tag=as55a7634a
To: <sip:3003@203.122.26.234:47318>
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '7b540611714508db45dffd9668ce7902@203.122.26.232' in 32000 ms
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:47318:
SIP/2.0 200 OK
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
From: "S0609151843338600051"<sip:asterisk@203.122.26.232>;tag=as55a7634a
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport=5060;received=203.122.26.232
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 CANCEL
Contact: <sip:3003@203.122.26.234:47318>
Content-Length: 0


--- (8 headers 0 lines)---
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:47318:
SIP/2.0 487 Request Terminated
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
From: "S0609151843338600051"<sip:asterisk@203.122.26.232>;tag=as55a7634a
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport=5060;received=203.122.26.232
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines)---
Transmitting (NAT) to 203.122.26.234:47318:
ACK sip:3003@203.122.26.234:47318 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport
From: "S0609151843338600051" <sip:asterisk@203.122.26.232>;tag=as55a7634a
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

SORRY OUTPUT PPOSTED ABOVE IS WRONG this one is THE ACTUAL

Postby asterisk_at_my_risk » Fri Sep 15, 2006 8:28 am

ACTUAL OUTPUT WHICH WE WANT TO GIVE

OUTPUT FOR SIP CHAANNLS CONCISE
********************************************************************************************************************

sip*CLI> show channels concise
SIP/3003-084cf0a8!default!s!5!Up!BackGround!demo-congrats!!!3!10!(None)
SIP/3003-084b7338!default!s!6!Up!BackGround!demo-instruct!!!3!37!(None)



OUTPUT FOR GENERAL ASTERISK DEBUGGUNG OF SIP
****************************************************************************************************************************

sip*CLI>
<-- SIP read from 203.122.26.234:47318:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:61854:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Sep 15 18:43:56 NOTICE[24359]: chan_local.c:498 local_alloc: No such extension/c ontext 8600051@default creating local channel
Sep 15 18:43:56 NOTICE[24359]: channel.c:2445 __ast_request_and_dial: Unable to request channel Local/8600051@default/n
sip*CLI>
<-- SIP read from 203.122.26.234:47318:


--- (0 headers 0 lines) Nat keepalive ---
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:61854:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Reliably Transmitting (NAT) to 203.122.26.234:47318:
CANCEL sip:3003@203.122.26.234:47318 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport
From: "S0609151843338600051" <sip:asterisk@203.122.26.232>;tag=as55a7634a
To: <sip:3003@203.122.26.234:47318>
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of call '7b540611714508db45dffd9668ce7902@203.122.26.232' in 32000 ms
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:47318:
SIP/2.0 200 OK
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
From: "S0609151843338600051"<sip:asterisk@203.122.26.232>;tag=as55a7634a
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport=5060;received=203.122.26.232
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 CANCEL
Contact: <sip:3003@203.122.26.234:47318>
Content-Length: 0


--- (8 headers 0 lines)---
== Manager 'sendcron' logged off from 127.0.0.1
sip*CLI>
<-- SIP read from 203.122.26.234:47318:
SIP/2.0 487 Request Terminated
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
From: "S0609151843338600051"<sip:asterisk@203.122.26.232>;tag=as55a7634a
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport=5060;received=203.122.26.232
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines)---
Transmitting (NAT) to 203.122.26.234:47318:
ACK sip:3003@203.122.26.234:47318 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK6f186eb6;rport
From: "S0609151843338600051" <sip:asterisk@203.122.26.232>;tag=as55a7634a
To: <sip:3003@203.122.26.234:47318>;tag=3d48344c
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 7b540611714508db45dffd9668ce7902@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

OUTPUT FORM MYSQL
************************************************************************************************
mysql> select * from live_sip_channels;
+-------------------+----------------+---------------+-----------+-------------- +
| channel | server_ip | channel_group | extension | channel_data |
+-------------------+----------------+---------------+-----------+-------------- +
| SIP/3003-084b7338 | 203.122.26.232 | NULL | ring | SIP/ring |
+-------------------+----------------+---------------+-----------+-------------- +
1 row in set (0.00 sec)

mysql> select * from live_channels;
Empty set (0.00 sec)
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby gerski » Fri Sep 15, 2006 12:03 pm

are you using softphone, if yes when you login to the vicidial.php does your softphone rings and when you answer it says... you are currently the only person in this conference
gerski
 
Posts: 432
Joined: Fri Jul 14, 2006 6:21 am

THanks for your reply

Postby asterisk_at_my_risk » Fri Sep 15, 2006 12:23 pm

THanks for your reply

i am using a eyebeam soft phone

and yes i am getting a incoming ring on my soft phone but the file which plays after that is DEMO-congrats from asterisk default sounds
i am also placing my my sip logs in which i have highlighted that part
also i am using ztdummy which i belive is working i checked it thru lsmod command i am currently trying to test it on on 2 seats after that i will exapnd it
please lemme know if you need any other info from my end

thanks again for your help



Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.3.15:6182
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex |ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 ( nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event) , combined - 0x1 (telephone-event)
list_route: hop: <sip:3003@203.122.26.234:27076>
set_destination: Parsing <sip:3003@203.122.26.234:27076> for address/port to sen d to
set_destination: set destination to 203.122.26.234, port 27076
Transmitting (NAT) to 203.122.26.234:27076:
ACK sip:3003@203.122.26.234:27076 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK0570e8b9;rport
From: "S0609152242098600051" <sip:asterisk@203.122.26.232>;tag=as65645334
To: <sip:3003@203.122.26.234:27076>;tag=4e04391d
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
> Channel SIP/3003-084b7338 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/3003-084b7338 at default,8600051,1 failed so falling back to e xten 's'
-- Executing Wait("SIP/3003-084b7338", "1") in new stack
-- Executing Answer("SIP/3003-084b7338", "") in new stack
-- Executing Set("SIP/3003-084b7338", "TIMEOUT(digit)=5") in new stack
-- Digit timeout set to 5
-- Executing Set("SIP/3003-084b7338", "TIMEOUT(response)=10") in new stack
-- Response timeout set to 10
--

*********************************************************************************************************************************
Executing BackGround("SIP/3003-084b7338", "demo-congrats") in new stack
*************************************************************************************************************************************



-- Playing 'demo-congrats' (language 'en')
sip*CLI> sip d
<-- SIP read from 203.122.26.234:27076:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI> sip dbeug
No such command 'sip dbeug' (type 'help' for help)
sip*CLI>
<-- SIP read from 203.122.26.234:4148:


--- (0 headers 0 lines) Nat keepalive ---
sip*CLI>
<-- SIP read from 203.122.26.234:27076:
BYE sip:asterisk@203.122.26.232:5060 SIP/2.0
To: "S0609152242098600051"<sip:asterisk@203.122.26.232>;tag=as65645334
From: <sip:3003@203.122.26.234:27076>;tag=4e04391d
Via: SIP/2.0/UDP 203.122.26.234:27076;branch=z9hG4bK-d87543-214828442-1--d87543-;rport
Call-ID: 11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 2 BYE
Contact: <sip:3003@203.122.26.234:27076>
Max-Forwards: 70
User-Agent: eyeBeam release 3007n stamp 17816
Content-Length: 0


--- (10 headers 0 lines)---
Sending to 203.122.26.234 : 27076 (NAT)
Transmitting (NAT) to 203.122.26.234:27076:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.122.26.234:27076;branch=z9hG4bK-d87543-214828442-1--d87543-;received=203.122.26.234;rport=27076
From: <sip:3003@203.122.26.234:27076>;tag=4e04391d
To: "S0609152242098600051"<sip:asterisk@203.122.26.232>;tag=as65645334
Call-ID: 11b82fbf708ced7a24ed12310d998b92@203.122.26.232
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:asterisk@203.122.26.232>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
== Spawn extension (default, s, 5) exited non-zero on 'SIP/3003-084b7338'
Destroying call '11b82fbf708ced7a24ed12310d998b92@203.122.26.232'
sip*CLI>
<-- SIP read from 203.122.26.234:27076:


--- (0 headers 0 lines) Nat keepalive ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 203.122.26.232 port 19880
Adding codec 0x40 (slin) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 21 lines
Reliably Transmitting (NAT) to 203.122.26.234:27076:
INVITE sip:3003@203.122.26.234:27076 SIP/2.0
Via: SIP/2.0/UDP 203.122.26.232:5060;branch=z9hG4bK4e94039c;rport
From: "ACagcW11583403496666" <sip:asterisk@203.122.26.232>;tag=as61d01c4f
To: <sip:3003@203.122.26.234:27076>
Contact: <sip:asterisk@203.122.26.232>
Call-ID: 1623fe4057d53b8529cc80556fc30d20@203.122.26.232
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Sep 2006 17:12:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 494

v=0
o=root 2296 2296 IN IP4 203.122.26.232
s=session
c=IN IP4 203.122.26.232
t=0 0
m=audio 19880 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

here is my screen output

Postby asterisk_at_my_risk » Fri Sep 15, 2006 12:32 pm

[root@sip ~]# screen -r
There are several suitable screens on:
11338.ASTlisten (Detached)
2608.ASTVDremote (Detached)
32129.pts-6.sip (Detached)
2598.ASTVDauto (Detached)
2295.asterisk (Detached)
2619.ASTupdate (Detached)
11342.ASTlisten (Detached)
11352.ASTsend (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.
[root@sip ~]#
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby gerski » Fri Sep 15, 2006 12:32 pm

could you please post your sip.conf and extension.conf?
gerski
 
Posts: 432
Joined: Fri Jul 14, 2006 6:21 am

Postby Dishem » Sat Sep 16, 2006 8:39 am

Where you get it???
Dishem
 
Posts: 1
Joined: Sat Sep 16, 2006 8:19 am
Location: USA

the issue is resloved thanks to matt , gerski

Postby asterisk_at_my_risk » Sun Sep 17, 2006 6:52 am

thanks we solved that issue there was issue in my extensions.conf
i left some original part of the config which came with asterisk when we do make samples i removed that and it is working fine

i am thankful to matt and gerskiwho help us to solve this issue
now slowly we are having a good idea of VICIDIAL

its great forum to learn where moderators are veruy prompt and eager to help
thanks again
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby gerski » Sun Sep 17, 2006 7:42 am

thanks... no problemo 8)
gerski
 
Posts: 432
Joined: Fri Jul 14, 2006 6:21 am

Postby sharad » Mon Sep 18, 2006 3:12 pm

Hi asterisk_at_my_risk,

We are facing a similar issue. Can you please let us know what changes did you make in extensions.conf to make this work?

Thanks again for your feedback.
sharad
 
Posts: 36
Joined: Fri Jul 21, 2006 7:47 am

hi sharad

Postby asterisk_at_my_risk » Thu Sep 21, 2006 11:10 am

i will be more than happy to help

with me the issue was in my extension.conf file there were some samples settings involved which were making my calls loopback rest every thing was fine
for i will suggest first check that all your important scripts are running
and after that also check your select * from live_channels and select * from
sip_live_channels
and also try to get a debug from your asterisk by asterisk -vccccccgc
and then post it on forum here moderators are great they will offcourse help you out
i am also in india
asterisk_at_my_risk
 
Posts: 102
Joined: Mon Sep 04, 2006 10:50 am
Location: New Delhi

Postby sharad » Fri Sep 22, 2006 6:43 pm

Hi asterisk_at_my_risk,

Thanks for your feedback. We will try your suggestions.

We are in Delhi too...can you pm me your contact number...we can collaborate and help each other out. Thanks again.
sharad
 
Posts: 36
Joined: Fri Jul 21, 2006 7:47 am

no session

Postby prabhakar_asterisk » Sat Oct 14, 2006 3:07 am

hi guys,
i am facing the same problem as asterisk_at_myrisk user

ouput
mysql> select * from live_sip_channels;
Empty set (0.00 sec)

mysql> select * from live_channels;
Empty set (0.00 sec)

[root@localhost ~]# /home/cron/AST_update.pl --debug
LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='125.22.242.191'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='125.22.242.191'|
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='125.22.242.191'|
prabhakar
Zap Clients: |
IAX2 Clients: |
SIP Clients: |prabhakar|prabhakar@prabhakar|
unknown remote host: localhost at /home/cron/AST_update.pl line 275

i configured manger.conf as given in scratch install
(his problem is because of managers.conf but my managers.conf is exact copy of scratch install)

i also installed in cli.c patch file correctly

i am eagerly awaiting the reply

thanks again

prabhakar
prabhakar_asterisk
 
Posts: 79
Joined: Sun Aug 06, 2006 11:09 pm

Postby mflorell » Sat Oct 14, 2006 9:13 pm

check that in your /etc/hosts file you have a localhost record.
mflorell
Site Admin
 
Posts: 18384
Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby prabhakar_asterisk » Sun Oct 15, 2006 11:13 pm

thanks ..localhost was missing i entered it..even after correcting it i m getting no sessions

OUTPUT
mysql> use asterisk;
Database changed
mysql> select * from live_sip_channels;
Empty set (0.00 sec)
mysql> select * from live_channels;
Empty set (0.00 sec)

[root@dsl-tn-static-191 ~]# /home/cron/AST_update.pl --debug
LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='125.22.242.191'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='125.22.242.191'|
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='125.22.242.191'|
prabhakar
Zap Clients: |
IAX2 Clients: |
SIP Clients: |prabhakar|prabhakar@prabhakar|

|SELECT channel,extension FROM live_channels where server_ip = '125.22.242.191'|

|UPDATE server_updater set last_update='2006-10-16 9:40:18' where server_ip='125.22.242.191'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '125.22.242.191'|

|UPDATE server_updater set last_update='2006-10-16 9:40:19' where server_ip='125.22.242.191'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '125.22.242.191'|

|UPDATE server_updater set last_update='2006-10-16 9:40:19' where server_ip='125.22.242.191'|
concise: -1

|DELETE FROM live_sip_channels where server_ip='125.22.242.191'|

|DELETE FROM live_channels where server_ip='125.22.242.191'|

|SELECT channel,extension FROM live_channels where server_ip = '125.22.242.191'|

IN THE ASTERISK CONSOLE
Parsing '/etc/asterisk/manager.conf': Found
[color=blue] == Manager 'updatecron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[/color]





prabhakar_asterisk
 
Posts: 79
Joined: Sun Aug 06, 2006 11:09 pm

Postby mflorell » Mon Oct 16, 2006 9:08 am

what do you have in your vicidial_conferences table?
mflorell
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Posts: 18384
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Location: Florida

Postby prabhakar_asterisk » Tue Oct 17, 2006 12:21 am

i have this in vicidial_conference table

mysql> select * from vicidial_conferences;
+------------+----------------+---------------+
| conf_exten | server_ip | extension |
+------------+----------------+---------------+
| 8600066 | 125.22.242.191 | SIP/prabhakar |
| 8600065 | 125.22.242.191 | |
| 8600064 | 125.22.242.191 | |
| 8600063 | 125.22.242.191 | |
| 8600062 | 125.22.242.191 | |
| 8600061 | 125.22.242.191 | |
| 8600060 | 125.22.242.191 | |
| 8600059 | 125.22.242.191 | |
| 8600058 | 125.22.242.191 | |
| 8600057 | 125.22.242.191 | |
| 8600056 | 125.22.242.191 | |
| 8600055 | 125.22.242.191 | |
| 8600054 | 125.22.242.191 | |
| 8600053 | 125.22.242.191 | |
| 8600052 | 125.22.242.191 | |
| 8600051 | 125.22.242.191 | |
+------------+----------------+---------------+

eagerly awaiting your reply
cheers
prabhakar
prabhakar_asterisk
 
Posts: 79
Joined: Sun Aug 06, 2006 11:09 pm

Postby prabhakar_asterisk » Thu Oct 26, 2006 1:45 am

i still couldnt able to figure out the problem
awaiting for your reply
thanks
prabhakar_asterisk
 
Posts: 79
Joined: Sun Aug 06, 2006 11:09 pm

Re:

Postby egris » Tue Nov 21, 2006 2:01 am

I've got the same problem with 1.2.13.

My instalation includes SIP phone and I could not place any outbound call from VICIdialer.

And I could not see any messages at my console (for example, executing meetme ...) and no info from live_channels and live_sip_channels
egris
 
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Postby mflorell » Tue Nov 21, 2006 7:57 am

post results of "screen -r"
mflorell
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Joined: Wed Jun 07, 2006 2:45 pm
Location: Florida

Postby egris » Tue Nov 21, 2006 8:03 am

root@debian:/var/www/gnudialer# screen -r
There are several suitable screens on:
5731.ASTVDadapt (Detached)
23091.ASTlisten (Dead ???)
10888.ASTsend (Dead ???)
6572.ASTlisten (Dead ???)
30296.ASTVDauto (Detached)
30426..debian (Detached)
13755.ASTupdate (Detached)


But I'm using OOH323 channel to make outbound calls. As far as I know from the docs it's not possible to use VICIDIAL with OOH323 trunks?
egris
 
Posts: 8
Joined: Tue Nov 21, 2006 2:00 am

Postby mflorell » Tue Nov 21, 2006 11:32 am

That is correct, you cannot use H323 with VICIDIAL.

Does your provider not support SIP or IAX?
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Site Admin
 
Posts: 18384
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Location: Florida

Postby egris » Tue Nov 21, 2006 11:41 am

Of cource not.

:wink:

this is Russia.
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Joined: Tue Nov 21, 2006 2:00 am

Postby gerski » Wed Nov 22, 2006 11:36 am

i have once used H323 in vicidial, but you will be needing voip gateway that is connected to your E1 card.

you will configured your Asterisk E1 by zaptel

and in voip gateway you will configure your h323 gateway.

correct me if im wrong.. 8)
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Joined: Fri Jul 14, 2006 6:21 am

Postby ckwall » Wed Nov 22, 2006 3:38 pm

mysql> select * from vicidial_conferences
-> ;
+------------+------------+--------------------------+
| conf_exten | server_ip | extension |
+------------+------------+--------------------------+
| 8600051 | 10.0.1.180 | SIP/IAX2/1718@10.0.1.189 |
| 8600052 | 10.0.1.180 | |
| 8600053 | 10.0.1.180 | |
| 8600054 | 10.0.1.180 | |
| 8600055 | 10.0.1.180 | |
| 8600056 | 10.0.1.180 | |
| 8600057 | 10.0.1.180 | |
| 8600058 | 10.0.1.180 | |
| 8600059 | 10.0.1.180 | |
| 8600060 | 10.0.1.180 | |
| 8600061 | 10.0.1.180 | |
| 8600062 | 10.0.1.180 | |
| 8600063 | 10.0.1.180 | |
| 8600064 | 10.0.1.180 | |
| 8600065 | 10.0.1.180 | |
| 8600066 | 10.0.1.180 | |
| 8600067 | 10.0.1.180 | |
| 8600068 | 10.0.1.180 | |
| 8600069 | 10.0.1.180 | |
| 8600070 | 10.0.1.180 | |
| 8600071 | 10.0.1.180 | |
| 8600072 | 10.0.1.180 | |
| 8600073 | 10.0.1.180 | |
| 8600074 | 10.0.1.180 | |
| 8600075 | 10.0.1.180 | |
| 8600076 | 10.0.1.180 | |
| 8600077 | 10.0.1.180 | |
| 8600078 | 10.0.1.180 | |
| 8600079 | 10.0.1.180 | |
| 8600080 | 10.0.1.180 | |
| 8600081 | 10.0.1.180 | |
| 8600082 | 10.0.1.180 | |
| 8600083 | 10.0.1.180 | |
| 8600084 | 10.0.1.180 | |
| 8600085 | 10.0.1.180 | |
| 8600086 | 10.0.1.180 | |
| 8600087 | 10.0.1.180 | |
| 8600088 | 10.0.1.180 | |
| 8600089 | 10.0.1.180 | |
| 8600090 | 10.0.1.180 | |
| 8600091 | 10.0.1.180 | |
| 8600092 | 10.0.1.180 | |
| 8600093 | 10.0.1.180 | |
| 8600094 | 10.0.1.180 | |
| 8600095 | 10.0.1.180 | |
| 8600096 | 10.0.1.180 | |
| 8600097 | 10.0.1.180 | |
| 8600098 | 10.0.1.180 | |
| 8600099 | 10.0.1.180 | |
| 8600100 | 10.0.1.180 | |
| 8600051 | 10.0.1.180 | SIP/IAX2/1718@10.0.1.189 |
| 8600052 | 10.0.1.180 | |
| 8600053 | 10.0.1.180 | |
| 8600054 | 10.0.1.180 | |
| 8600055 | 10.0.1.180 | |
| 8600056 | 10.0.1.180 | |
| 8600057 | 10.0.1.180 | |
| 8600058 | 10.0.1.180 | |
| 8600059 | 10.0.1.180 | |
| 8600060 | 10.0.1.180 | |
| 8600061 | 10.0.1.180 | |
| 8600062 | 10.0.1.180 | |
| 8600063 | 10.0.1.180 | |
| 8600064 | 10.0.1.180 | |
| 8600065 | 10.0.1.180 | |
| 8600066 | 10.0.1.180 | |
| 8600067 | 10.0.1.180 | |
| 8600068 | 10.0.1.180 | |
| 8600069 | 10.0.1.180 | |
| 8600070 | 10.0.1.180 | |
| 8600071 | 10.0.1.180 | |
| 8600072 | 10.0.1.180 | |
| 8600073 | 10.0.1.180 | |
| 8600074 | 10.0.1.180 | |
| 8600075 | 10.0.1.180 | |
| 8600076 | 10.0.1.180 | |
| 8600077 | 10.0.1.180 | |
| 8600078 | 10.0.1.180 | |
| 8600079 | 10.0.1.180 | |
| 8600080 | 10.0.1.180 | |
| 8600081 | 10.0.1.180 | |
| 8600082 | 10.0.1.180 | |
| 8600083 | 10.0.1.180 | |
| 8600084 | 10.0.1.180 | |
| 8600085 | 10.0.1.180 | |
| 8600086 | 10.0.1.180 | |
| 8600087 | 10.0.1.180 | |
| 8600088 | 10.0.1.180 | |
| 8600089 | 10.0.1.180 | |
| 8600090 | 10.0.1.180 | |
| 8600091 | 10.0.1.180 | |
| 8600092 | 10.0.1.180 | |
| 8600093 | 10.0.1.180 | |
| 8600094 | 10.0.1.180 | |
| 8600095 | 10.0.1.180 | |
| 8600096 | 10.0.1.180 | |
| 8600097 | 10.0.1.180 | |
| 8600098 | 10.0.1.180 | |
| 8600099 | 10.0.1.180 | |
| 8600100 | 10.0.1.180 | |
+------------+------------+--------------------------+
100 rows in set (0.00 sec)

mysql>
ckwall
 
Posts: 105
Joined: Tue Nov 21, 2006 1:03 pm

Postby ckwall » Wed Nov 22, 2006 3:57 pm

ok, more tweaking and testing... now I am getting this when i try to dial a lead:

== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-5e27,2", "8600051") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Playing 'conf-invalid' (language 'en')
== Starting Local/8600051@default-5e27,1 at default,9918018285230,1 failed so falling back to exten 's'
== Starting Local/8600051@default-5e27,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-5e27,1
-- Executing Playback("Local/8600051@default-5e27,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-5e27,2'
ckwall
 
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