[asterisk-users] Low cost routing
Hello, I need some advice: I use 2 different suppliers of trunk SIP in my infrastructure, both send me regularly prices in a .csv format. So I have two SQL tables that contain the prefix and the tariff. For now, I generate a dialplan with a Perl script that allows me to select the prefix trunk to use but the problem is that I change it manually in some cases. For example Trunk A: +3550.1698€ +35521150 0.12815€ +35521151 0.12815€ Trunk B: +3550.1144€ Currently my script sees that +35521150 exists in Trunk A and will therefore use it while on the Trunk B it is less expensive but the prefix +35521150 is not directly indicated since it is covered by the +355 I would like to change that, generate a unique table of prefixes. Someone would have done that already? Thank you, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question Asterisk Manager
Hi A small question on Asterisk Manager. I use Perl Script for start a call: my $response = $astman-sendcommand( Action = 'Originate', Channel = 'SIP/ASTERISK/$Extension', Exten = '200', Context = 'MyContext', Priority = '1', Async = '1' ); That's start the call, but only the position of the corresponding sounds departing. As soon as he clinched, that the second ringing phone. Is there a way for two phone ring at the same time? Thanks Olivier -- _ -- 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] MOH don't work after update
i don't think's that it's the same problems, because me format_mp3.so is loaded: [root@ipbx Conf-Extensions]# asterisk -rx 'core show file formats' Format Name Extensions -- -- slin mp3mp3 gsmgsmgsm slin192sln192 sln192 slin96 sln96 sln96 slin48 sln48 sln48 slin44 sln44 sln44 slin32 sln32 sln32 slin24 sln24 sln24 slin16 sln16 sln16 slin12 sln12 sln12 slin slnsln|raw ilbc iLBC ilbc g723 g723sf g723|g723sf slin16 wav16 wav16 slin wavwav siren14siren14siren14 g719 g719 g719 h264 h264 h264 g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 g729 g729 g729 siren7 siren7 siren7 gsmwav49 WAV|wav49 g722 g722 g722 ulaw au au alaw alaw alaw|al|alw ulaw pcmpcm|ulaw|ul|mu|ulw adpcm voxvox h263 h263 h263 31 file formats registered. i see the mp3 file format 2013/6/17 Thorsten Göllner t...@ovm-group.com Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- 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] MOH don't work after update
Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 thanks for your help Olivier -- _ -- 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] MOH don't work after update
i use the package centos, i can't use menuselect no ? but i think's that is loaded: ipbx*CLI module load format_mp3.so Unable to load module format_mp3.so Command 'module load format_mp3.so' failed. [Jun 17 04:56:42] WARNING[8910]: loader.c:892 load_resource: Module 'format_mp3.so' already exists. ipbx*CLI 2013/6/16 Matthew Jordan mjor...@digium.com On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO o.calv...@gmail.comwrote: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 Do you have the format_mp3 module loaded? Add-on modules are in the addons subdirectory. Typically, these modules are not built and installed by default, and have to be enabled in menuselect. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- 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
[asterisk-users] Extenxions Optimization
Hi
We want optimize my extensions file conf on asterisk 11.4.0 :
We have a big quantity of extensions, all are same design:
; Destination: Gambia Type: Fixe
exten = _00220X.,1,Set(CDR(CodeCom)=BUS-GMB)
exten = _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
exten = _00220X.,3,Set(CALLERID(all)=${NUMID})
exten = _00220X.,4,Set(CALLERPRES()=${CALLPRES})
exten = _00220X.,5,Dial(SIP/Trunk-Telco/${EXTEN:2},180,rt)
exten = _00220X.,6,Hangup
; Destination: Libya Type: Fixe
exten = _00218X.,1,Set(CDR(CodeCom)=BUS-LBY)
exten = _00218X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
exten = _00218X.,3,Set(CALLERID(all)=${NUMID})
exten = _00218X.,4,Set(CALLERPRES()=${CALLPRES})
exten = _00218X.,5,Dial(SIP/Trunk-Telco/${EXTEN:2},180,rt)
exten = _00218X.,6,Hangup
; Destination: Tunisia Type: Fixe
exten = _00216X.,1,Set(CDR(CodeCom)=BUS-TUN)
exten = _00216X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
exten = _00216X.,3,Set(CALLERID(all)=${NUMID})
exten = _00216X.,4,Set(CALLERPRES()=${CALLPRES})
exten = _00216X.,5,Dial(SIP/Trunk-Telco/${EXTEN:2},180,rt)
exten = _00216X.,6,Hangup
; Destination: Algeria Type: Fixe
exten = _00213X.,1,Set(CDR(CodeCom)=BUS-DZA)
exten = _00213X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
exten = _00213X.,3,Set(CALLERID(all)=${NUMID})
exten = _00213X.,4,Set(CALLERPRES()=${CALLPRES})
exten = _00213X.,5,Dial(SIP/Trunk-Telco/${EXTEN:2},180,rt)
exten = _00213X.,6,Hangup
My .conf file is ~8 line
He have a solution for reduc this ? because a lot of line if same
I think's use a AGI style:
; Destination: Libya Type: Fixe
exten = _00218X.,1,AGI(Extensions.agi,${IAXVAR(ACCOUNTID)})
; Destination: Tunisia Type: Fixe
exten = _00216X.,1,AGI(Extensions.agi,${IAXVAR(ACCOUNTID)})
; Destination: Algeria Type: Fixe
exten = _00213X.,1,AGI(Extensions.agi,${IAXVAR(ACCOUNTID)})
and into my Extensions.agi, i sent the other line. Do you think's that it's
a good idea ?
Best regards
Olivier
--
_
-- 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] Difference MySQL between 1.6.x and 11.4.x
Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- 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] Difference MySQL between 1.6.x and 11.4.x
The database schema (table) is different in Asterisk 11.4 ? because i have configured: cdr_mysql.conf extconfig.conf res_config_mysql.conf and on the mysql server, it's the old database of 1.6.x i see: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx (err 2003). Check debug for more info. can i put debug ? i don't know where thanks olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com It looks like your database configuration is missing in Asterisk. It is making up information about the connection using defaault values as if it did not find any database configuration. Ron On 03/06/2013 10:49 AM, Olivier CALVANO wrote: Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
Strange too, in the logs: [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). [Jun 3 17:09:49] NOTICE[3464] config.c: Registered Config Engine mysql [Jun 3 17:09:49] ERROR[3464] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on (err 2003). Check debug for more info. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: Table Comptes_IAX not found in database. This table should exist if you're using realtime. Hi said No database host found but in the log i have Failed to connect database server SSI on with SSI and correct into my config file 2013/6/3 Olivier CALVANO o.calv...@gmail.com The database schema (table) is different in Asterisk 11.4 ? because i have configured: cdr_mysql.conf extconfig.conf res_config_mysql.conf and on the mysql server, it's the old database of 1.6.x i see: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx (err 2003). Check debug for more info. can i put debug ? i don't know where thanks olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com It looks like your database configuration is missing in Asterisk. It is making up information about the connection using defaault values as if it did not find any database configuration. Ron On 03/06/2013 10:49 AM, Olivier CALVANO wrote: Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail; +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ | uniqueid | customer_id | context | mailbox| password | fullname | email | pager | tz | attach | saycid | dialout | callback | review | operator | envelope | sayduration | saydurationm | sendvoicemail | delete | nextaftercmd | forcename | forcegreetings | hidefromdir | stamp | +--+-+--++--+--+---+---+-+++-+--++--+--+-+--+---++--+---++-+-+ .. anyone know the problems ? thanks olivier -- _ -- 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] Difference MySQL between 1.6.x and 11.4.x
Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights reserved. Oracle is a registered trademark of Oracle Corporation and/or its affiliates. Other names may be trademarks of their respective owners. Type 'help;' or '\h' for help. Type '\c' to clear the current input statement. mysql select * from VoiceMail
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn after in extconfig.conf: sipusers = mysql,general,Comptes_SIP sippeers = mysql,general,Comptes_SIP iaxusers = mysql,general,Comptes_IAX iaxpeers = mysql,general,Comptes_IAX extensions = mysql,general,Extensions meetme = mysql,general,MeetMe musiconhold = mysql,general,Musiconhold voicemail = mysql,general,VoiceMail and in cdr_mysql.conf [global] hostname=myhost.mydomain.net dbname=MyDB table=Cdr password=MyPassword user=MyUser port=3306 sock=/tmp/mysql.sock [aliases] start=calldate end=callend callerid=clid src=src dst=dst dcontext=dcontext channel=channel dstchannel=dstchannel lastapp=lastapp lastdata=lastdata duration=duration billsec=billsec disposition=disposition amaflags=amaflags accountcode=accountcode userfield=userfield uniqueid=uniqueid CodeTier=CodeTier you know what file I forgot to configure? Olivier 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Fix this. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 17:09:49] WARNING[3464] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). Asterisk is telling you that you have not configured ANY database. It is not worrying about what tables are in it because you have not even defined the database itself. There is NO database at all so worrying about versions is not Asterisk's big problem.. The rest of the messages after that are a bit screwy because the routines producing the error are not aware that there is no database at all so they just complain about the piece that they know about. Ron On 03/06/2013 12:19 PM, Olivier CALVANO wrote: No other idea ? 2013/6/3 Olivier CALVANO o.calv...@gmail.com Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info. [Jun 3 16:27:59] WARNING[3140] res_config_mysql.c: Table VoiceMail not found in database. This table should exist if you're using realtime. [Jun 3 16:27:59] ERROR[3140] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:14] ERROR[3220] cdr_mysql.c: Failed to connect to mysql database SSI on myhost.myserver.com. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database user found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database password found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database host found, using localhost via socket. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database name found, using 'asterisk' as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database port found, using 3306 as default. [Jun 3 16:30:34] WARNING[3220] res_config_mysql.c: MySQL RealTime: No database socket found (and unable to detect a suitable path). The exacly same config work on 1.6.x and from the new server, the database access is Ok: [root@voip-2 log]# !mys mysql -h myhost.myserver.com -u Asterisk -p SSI Enter password: Reading table information for completion of table and column names You can turn off this feature to get a quicker startup with -A Welcome to the MySQL monitor. Commands end with ; or \g. Your MySQL connection id is 5185 Server version: 5.1.42-log Mandriva Linux - MySQL Standard Edition (GPL) Copyright (c) 2000, 2013, Oracle and/or its affiliates. All rights
Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x
oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Difference MySQL between 1.6.x and 11.4.x
grrr no in asterisk -d i have no error, but when i start normaly asterisk i have : [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. what is the command in asterisk for i see the SQL query ? 2013/6/4 Olivier CALVANO o.calv...@gmail.com oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure MySQL socket is in /tmp directory? dbsock = /tmp/mysql.sock Regards El 03/06/2013 12:16, Olivier CALVANO escribió: Thanks for your help Ron, Do you know where is the confirguration ? Because i have put into res_config_mysql.conf: [general] dbhost = myhost.mydomain.net dbname = MyDB dbuser = MyUser dbpass = MyPassword dbport = 3306 dbsock = /tmp/mysql.sock dbcharset = latin1 requirements = warn -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Difference MySQL between 1.6.x and 11.4.x
What is the best commande for put the debug ? because with core set debug, i don't have a return. voip-2*CLI realtime mysql status Vop configured for m...@myhost.mydomain.net, port 3306 with username Asterisk. [Jun 4 06:48:25] ERROR[27879]: res_config_mysql.c:1577 mysql_reconnect: MySQL RealTime: Failed to connect database server Vop on vop.phibee-telecom.net (err 2003). Check debug for more info. He read correctly the config because it's the good DB, Server and username in /var/log/asterisk/message i have: [Jun 4 06:46:21] Asterisk 11.4.0 built by mockbuild @ buildvm-12.phx2.fedoraproject.org on a x86_64 running Linux on 2013-05-20 15:47:05 UTC [Jun 4 06:46:21] NOTICE[27825] loader.c: 1 modules will be loaded. [Jun 4 06:46:21] NOTICE[27825] config.c: Registered Config Engine mysql [Jun 4 06:46:21] NOTICE[27825] cdr.c: CDR simple logging enabled. [Jun 4 06:46:21] NOTICE[27825] loader.c: 192 modules will be loaded. [Jun 4 06:46:21] NOTICE[27825] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener. [Jun 4 06:46:21] WARNING[27825] res_musiconhold.c: No music on hold classes configured, disabling music on hold. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] WARNING[27825] res_config_mysql.c: Table Comptes_IAX not found in database. This table should exist if you're using realtime. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] ERROR[27825] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:46:21] WARNING[27825] res_config_mysql.c: Table Comptes_SIP not found in database. This table should exist if you're using realtime. [Jun 4 06:46:21] NOTICE[27825] confbridge/conf_config_parser.c: Adding default_user profile to app_confbridge [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: Starting AEL load process. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:21] NOTICE[27825] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 4 06:46:22] ERROR[27825] cdr_mysql.c: Failed to connect to mysql database MyDB on MyHost.MyDomain.net http://myhost.mydomain.net/. [Jun 4 06:47:51] ERROR[27879] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. [Jun 4 06:48:25] ERROR[27879] res_config_mysql.c: MySQL RealTime: Failed to connect database server MyDB on MyHost.MyDomain.nethttp://myhost.mydomain.net/(err 2003). Check debug for more info. Asterisk have the good information, but i don't understand why he can't connect to the DB, if i use: mysql -h MyHost.MyDomain.net http://myhost.mydomain.net/ -u Aserisk -p MyDB i have a full access to my MySQL server. may be missing in a Fedora package ? 2013/6/4 Ron Wheeler rwhee...@artifact-software.com Well, at least you are making progress. What is the error in the debug log? Ron On 03/06/2013 8:03 PM, Olivier CALVANO wrote: grrr no in asterisk -d i have no error, but when i start normaly asterisk i have : [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. [Jun 4 02:01:45] ERROR[6563] res_config_mysql.c: MySQL RealTime: Failed to connect database server xxx on xxx.xxx.net (err 2003). Check debug for more info. what is the command in asterisk for i see the SQL query ? 2013/6/4 Olivier CALVANO o.calv...@gmail.com oh ron thanks for your help : We have deleted all commented line, only put the configuration and now that's work ! 2013/6/3 Ron Wheeler rwhee...@artifact-software.com Do you have this problem in your conf file? http://forums.digium.com/viewtopic.php?p=63736 The parser won't accept an ; (semicolon) for remarks! So he found at the first the old remarks and tried to access my database with the false data. Ron On 03/06/2013 3:18 PM, Olivier CALVANO wrote: on this server we don't have mysql.socket because he don't have mysql server we want access to a mysql based on a other server 2013/6/3 Bakko asannu...@gmail.com Hello, are you sure
Re: [asterisk-users] Strange problem on ougoing call
Perfect that's work ;=)
very thanks
Le 25 avril 2012 10:19, Olivier CALVANO o.calv...@gmail.com a écrit :
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server -
only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier
which
is not registered on server-2 but is declared. Server-2 thinks that this
is
my valid extension but it is not registered with me and so lets
authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has
different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
Re: [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??
Anyknow know this problems ? I read on the net that it's a possible network problems, but i don't think because it's a VMWare server and in the same server i have other asterisk without this problems. best regards Olivier Le 25 avril 2012 09:35, Olivier CALVANO o.calv...@gmail.com a écrit : Hi i have a lot of error in the CLI of one of my Asterisk: [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:47] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:49] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:50] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:51] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x862b178 (len 886) to 172.16.251.46:5060 returned -1: Operation not permitted [Apr 25 09:30:53] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:54] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted anyone know what is this error ? thanks olivier -- _ -- 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] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??
Hi No firewall on the server Other idea ?? Hihi Olivier Le jeudi 26 avril 2012, Duncan Turnbull a écrit : Usually its a firewall issue, or at least it has been for me Its saying it can't form sip packets, and that will be because something isn't letting it, Cheers Duncan On 26/04/2012, at 8:15 PM, Olivier CALVANO wrote: Anyknow know this problems ? I read on the net that it's a possible network problems, but i don't think because it's a VMWare server and in the same server i have other asterisk without this problems. best regards Olivier Le 25 avril 2012 09:35, Olivier CALVANO o.calv...@gmail.comjavascript:; a écrit : Hi i have a lot of error in the CLI of one of my Asterisk: [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:47] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:49] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:50] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:51] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x862b178 (len 886) to 172.16.251.46:5060 returned -1: Operation not permitted [Apr 25 09:30:53] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:54] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted anyone know what is this error ? thanks olivier -- _ -- 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 -- _ -- 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] Strange problem on ougoing call
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server - only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier which
is not registered on server-2 but is declared. Server-2 thinks that this is
my valid extension but it is not registered with me and so lets authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr
[asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??
Hi i have a lot of error in the CLI of one of my Asterisk: [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:47] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:49] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:50] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted [Apr 25 09:30:51] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x862b178 (len 886) to 172.16.251.46:5060 returned -1: Operation not permitted [Apr 25 09:30:53] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation not permitted [Apr 25 09:30:54] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8639de8 (len 778) to 172.16.0.70:5060 returned -1: Operation not permitted anyone know what is this error ? thanks olivier -- _ -- 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] Strange problem on ougoing call
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
2012/4/25 Olivier CALVANO o.calv...@gmail.com
Sure, sorry for the Confusion ;=)
Server A Trader:
Asterisk Server 1.6.x for call routing only.
IP Adress: 172.16.0.11
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Server B Ipbx
Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone.
IP Adress: 172.16.0.70
Second IP: 172.16.1.70 (used for phone lan)
Use Realtim on MySQL Database
This server route all call to a lot of VoIP Carrier.
Linksys SPA942 A:
IP Adress: 172.16.1.200
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User01
Linksys SPA942 B:
IP Adress: 172.16.1.220
Connected in SIP at Server B IPBX
use sip.conf (no realtime)
context: I-User02
On Server A Trader, we have two sip account:
accountname: USER01 for user in group 1
accountname: USER02 for user in group 2
On Server B Ipbx, i use registry:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
for two connection to the Trader Server. Registry is good:
on server A Trader:
trader*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
On server B Ipbx, i have into my sip.conf after the registry:
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
and in extensions.conf:
[I-User01]
exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
[I-User02]
exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
When i call with Linksys SPA942 A, i use the context I-User01 and
the call are sent
to SIP account USER01 and No problems.
When i call with Linksys SPA942 B, i use the context I-User02 and
the call are sent
to SIP account USER02 but Server A Trader reject the call
immediatly with this error:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
Olivier and 906280 is the information that i have on the Linksys
SPA942 B, 906280 is the username used between
best ? hihi
Olivier
Le 25 avril 2012 06:38, SamyGo govoi...@gmail.com a écrit :
Hi,
Lots of mixing and confusing stuff - Can you re-explain the topology you
are
trying to achieve with proper IP addresses and declared sip ext. names.
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
Somehow it reminds of the same situation I always face when a peer is
declared with the same name as of the dialing one on second server -
only
Its just not registered there instead registered on server-1.
So when the call comes in from server-1 to server-2 fromuser=olivier
which
is not registered on server-2 but is declared. Server-2 thinks that this
is
my valid extension but it is not registered with me and so lets
authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has
different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat
[asterisk-users] Asterisk don't use context=
Hi I have a iax configuration: [IaxServer] type=friend host=172.16.1.14 port=4569 defaultuser=IaxServer auth=md5 secret=mypassword context=Internal peercontext=Internal qualify=yes trunk=no disallow=all allow=alaw i see the peer: ipbx*CLI iax2 show peers Name/UsernameHost Mask Port Status IaxServer 172.16.1.14 (S) 255.255.255.255 4569 (E) OK (6 ms) ipbx*CLI iax2 show peer IaxServer * Name : IaxServer Secret : Set Context : Internal Parking lot : Mailbox : Dynamic : No Callnum limit: 0 Calltoken req: No Trunk: No Callerid : Expire : -1 ACL : No Addr-IP : 172.16.1.14 Port 4569 Defaddr-IP : 0.0.0.0 Port 0 Username : Codecs : 0x8 (alaw) Codec Order : (alaw) Status : OK (4 ms) Qualify : every 6ms when OK, every 1ms when UNREACHABLE (sample smoothing Off) But when i receive the call, my server said: [Apr 24 09:25:30] NOTICE[22148]: chan_iax2.c:10241 socket_process: Rejected connect attempt from 172.16.1.14, request '280@default' does not exist He search the extention 280 in default but not in Internal Anyone know why ? for information, the 172.16.1.14 is a old asterisk server and i have put it into calltokenoptional thanks for your help Olivier -- _ -- 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] No extension found ?
Hi Sammy, Yes my telco have a lot of IP, i receive a call from ~20 ip .. I can't put a subnet ? best regards Le 23 avril 2012 07:57, SamyGo govoi...@gmail.com a écrit : Hi, No matching peer for '+331MYCLID' from '84.xx.xx.72:5060' This line is telling you everything. The peer you've declared isn't being matched for the incoming call and hence it tries to look in default context (I assume allowguest=yes in your sip.conf) Make sure that your peer is matched, since you've qualify=yes defined execute the command sip show peer Trunk-Telco in asterisl CLI and see the status of the peer. What I'm guessing is that the telco has multiple IPs to send you calls and the incoming call isn't coming from the IP you've declared in your sip telco-trunk section. I don't think we can set a subnet in host=87.XX.XX.XX/28 parameter.!! Regards, Sammy. -- _ -- 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] Strange problem on ougoing call
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
i have one phone connected to the context I-User01 and another
connected to I-User02
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
On the first server:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
The exten:
On I-User01: exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
On I-User02: exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
i i change on the I-User02:
Dial(SIP/USER02/${EXTEN:1},90,r)
in
Dial(SIP/USER01/${EXTEN:1},90,r)
all call work's.
anyone have a idea ? i think's that i have a error but don't see where
best regards
Olivier
--
_
-- 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] Strange problem on ougoing call
Hi
No idea ?
thanks
Olivier
Le 24 avril 2012 16:06, Olivier CALVANO o.calv...@gmail.com a écrit :
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this server):
I have in sip.conf:
register = USER01:1234@172.16.0.11/USER01
register = USER02:5678@172.16.0.11/USER02
[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite
i see the registration:
ipbx*CLI sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59
i have one phone connected to the context I-User01 and another
connected to I-User02
When i call with the phone connected to I-User01, no problems, that's
work but when i call
with the second phone (use I-User02) i have a error:
On the first server:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have USER01, digest has USER02
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device Olivier
sip:906280@172.16.0.70;tag=as0cd775ab
The exten:
On I-User01: exten = _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
On I-User02: exten = _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)
i i change on the I-User02:
Dial(SIP/USER02/${EXTEN:1},90,r)
in
Dial(SIP/USER01/${EXTEN:1},90,r)
all call work's.
anyone have a idea ? i think's that i have a error but don't see where
best regards
Olivier
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No extension found ?
Hi
I have a small problems with incoming call.
I have a peer actually configured for outcall :
sip.conf:
[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming
This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a extension not found.
In extensions.conf for incoming:
[incoming]
exten = _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)
in dialplan show incoming, no problems i see the dialplan.
when i call, i have:
--- SIP read from UDP://84.xx.xx.72:5060 ---
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route: sip:84.xx.xx.72;r2=on;lr;f=4
Record-Route: sip:172.16.21.172;r2=on;lr;f=4
Record-Route: sip:172.16.21.67;lr;f=8
Record-Route: sip:172.16.20.119;lr;did=247.29f60367
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: +331MYCLID
sip:+331MYCLID;tgrp=RT43@172.16.21.11;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To: sip:+331NUMNOFOUND@172.16.20.119
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact: sip:+331MYCLID@172.16.21.11:5060
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity: sip:+331myc...@domaineofmysupplier.net
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value=4f924d2c1e20abe1d@172.16.20.119
X-PSN-Trunk: ME
v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes
-
--- (25 headers 17 lines) ---
== Using SIP RTP CoS mark 5
Sending to 84.xx.xx.72 : 5060 (no NAT)
Using INVITE request as basis request -
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Peer audio RTP is at port 84.xx.xx.34:64296
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found unknown media description format X-CCD for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 84.xx.xx.34:64296
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)
--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0;received=84.xx.xx.72
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: +331MYCLID
sip:+331MYCLID;tgrp=RT43@172.16.21.11;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To: sip:+331NUMNOFOUND@172.16.20.119;tag=as53fc96aa
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Server: Asterisk PBX 1.6.1.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.
a idea of the problems ?
My supplier use a lot of server, i thinkss that my asterisk don't link
IP of the incoming server to the extensions
thanks for your help
olivier
--
_
-- 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] Delete Session timer ?
Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Best regards Olivier -- _ -- 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] Delete Session timer ?
yes i have put this option, but asterisk sent in the Header that he support the Session Timers, the sip server of the operator sent a session timer too and asterisk ignor it. my objectifs is asterisk don't sent the session timer Le 18 avril 2012 17:56, Barry Miller asterisk-us...@notanet.net a écrit : On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote: Hi can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6 Have you tried 'session-timers=refuse' ? -- Barry -- _ -- 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] Set variables from one asterisk ta a second.
Hi
greats thanks that work very good
Olivier
Le 16 avril 2012 12:47, Stuart Elvish - IP Exchange Systems
asterisk.li...@ipesys.com a écrit :
Hi,
If you are using IAX and a later version (I know it works in 1.8.x) you
can use IAXVAR.
The following URL has a post which has a good example.
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022313.html
Kind Regards
Stuart Elvish
On 04/16/2012 08:16 AM, Steve Edwards wrote:
On Sun, 15 Apr 2012, Olivier CALVANO wrote:
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
Yes.
(I'm just a 1.2 Luddite, so the exact capabilities and syntax available
to your version may be different.)
The first question is 'Do you want to use SIP or IAX?' You've used IAX
in your dialplan snippet, but you may want to consider SIP. The initial
configuration is a bit more involved, but you will have a more flexible
and maintainable solution.
Using IAX is simpler but you are limited to 'overloading' the caller ID
'name' and 'num' fields*. If you have more than a couple of fields of
data to pass you may find it easier to pass a 'key' (like the server
name and the channel unique ID) and use that to retrieve data from a
database instead of having to parse a bunch of fields from the caller ID
name or number.
Using SIP you can also pass data by adding custom SIP headers.
Personally, I've always used IAX because it was easy and it worked in my
environment. If I were to start over, I would seriously consider SIP.
A simple IAX example snippet...
On server1:
exten = *,n, set(CALLERID(name)=olivier-calvano)
exten = *,n, dial(iax2/server2/${EXTEN})
exten = *,n, hangup()
On server2:
exten = *,n, set(FIRST=${CUT(CALLERID(name),,1)})
exten = *,n, set(LAST=${CUT(CALLERID(name),,2)})
exten = *,n, agi(lookup-client,${FIRST},${LAST})
exten = *,n, hangup()
*) The extension and context are also under your control and can be set
in the IAX 'dial string' but manipulating these fields to pass multiple
data fields can get convoluted.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set variables from one asterisk ta a second.
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the USRSIP001
for get it into a AGI script.
It's possible ?
thanks for your help
Olivier
--
_
-- 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] Set variables from one asterisk ta a second.
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
exten = _x,3,Dial(IAX2/Srv2/${EXTEN}/USRSIP001,180,rt)
you should get the value in $CALLID[1-4] on the second server.
--
Daniel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set variables from one asterisk ta a second.
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set variables from one asterisk ta a second.
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
exten = _x,3,Dial(IAX2/Srv2/${EXTEN}/USRSIP001,180,rt)
you should get the value in $CALLID[1-4] on the second server.
--
Daniel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set variables from one asterisk ta a second.
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO o.calv...@gmail.com a écrit :
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set variables from one asterisk ta a second.
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
exten = _x,3,Dial(IAX2/Srv2/${EXTEN}/USRSIP001,180,rt)
you should get the value in $CALLID[1-4] on the second server.
--
Daniel
--
_
-- 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
--
_
-- 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] Set variables from one asterisk ta a second.
Ok,
the CLI of the server one :
-- Executing [06@Unlimited-outgoing:3]
Dial(SIP/USRSIP05-0a7a52e8,
IAX2/Srv2/06/USRSIP05,180,rt) in new stack
-- Called Trader/06/USRSIP05
The CLI of the server two:
srv2*CLI
-- Accepting AUTHENTICATED call from 172.20.8.1:
requested format = alaw,
requested prefs = (alaw|g729),
actual format = alaw,
host prefs = (alaw|g729),
priority = mine
[Apr 15 21:46:26] ERROR[31094]: pbx.c:2860 ast_func_read: Function
$CALLERID not registered
-- Executing [0@Appels-Sortants:1] Verbose(IAX2/VoIP-953,
passed ID ) in new stack
passed ID
[Apr 15 21:46:26] ERROR[31094]: pbx.c:2860 ast_func_read: Function
$CALLERID not registered
-- Executing [0@Appels-Sortants:2] AGI(IAX2/VoIP-953,
MyScript.agi,) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/MyScript.agi
-- IAX2/VoIP-953AGI Script MyScript.agi completed, returning 0
-- Executing [0@Appels-Sortants:3] Dial(IAX2/VoIP-953,
SIP/Telco/33,180,rt) in new stack
== Using SIP RTP CoS mark 5
-- Called Telco/336
== Spawn extension (Appels-Sortants, 06, 3) exited non-zero on
'IAX2/VoIP-953'
-- Hungup 'IAX2/VoIP-953'
srv2*CLI
Le 15 avril 2012 21:39, Danny Nicholas da...@debsinc.com a écrit :
Change this
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
to this
exten = _x,2,Verbose(passed ID ${$CALLERID(num)})
exten = _x,3,AGI(MyScript.agi,${$CALLERID(num){0:4}})
and post your CLI output. We need to see if the OP's suggestion is getting
to Asterisk #2.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set variables from one asterisk ta a second.
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO o.calv...@gmail.com a écrit :
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set variables from one asterisk ta a
second.
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten = _x,2,Set(CALLERID(num)=${CALLERIDNUM})
exten = _x,3,Dial(IAX2/Srv2/${EXTEN},180,rt)
exten = _x,4,Hangup
i have SIP user: USRSIP001
(user sip is in realtime)
he use this name with a password
i want that the first server sent to the second into a variable the
USRSIP001
for get it into a AGI script.
It's possible ?
exten = _x,3,Dial(IAX2/Srv2/${EXTEN}/USRSIP001,180,rt)
you should get the value in $CALLID[1-4] on the second server.
--
Daniel
--
_
-- 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
--
_
-- 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
[asterisk-users] Change extension for international ?
Hi i am search a solution for change the number called. Sample: I have a Linksys SPA942 connected in SIP with my server. When this phone call a number: 043112 automatiquely change in 3343112 because my carrier want a number in international format. It's possible ? thanks Olivier -- _ -- 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] Limit Call ?
Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- 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] Limit Call ?
Thanks but i read: ; The maximum number of concurrent calls you want to allow Not limit the duration of a call ;=) Le 2 avril 2012 16:55, Bakko asannu...@gmail.com a écrit : Hi, look at maxcalls parameter on the asterisk.conf file. regards El 02/04/2012 16:46, Olivier CALVANO escribió: Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier -- _ -- 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 -- _ -- 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] Know the number of concurrent dial ?
Hi
Thanks for your answer and help but your script don't work on my server
best regards
olivier
2011/8/4 Danny Nicholas da...@debsinc.com:
I did a PERL routine to get this information out of Master.csv
(/var/log/asterisk/cdr-csv)
open (my $cdr_in, , /var/log/asterisk/cdr-csv/Master.csv);
my %call_start;
my %call_end;
my %call_con;
my $call_max=0;
while ($cdr_in) {
my (@cdr_data) = split /\,/, $_;
my $day=unpack(x1 a10, $cdr_data[9]);
my $dx=sprintf(%20s, $cdr_data[9]);
my $hour=unpack(x12 a2, $dx);
my $xx= $cdr_data[11];
my $dur= $cdr_data[12];
my $syr=unpack(x1 a4, $dx);
my $smon=unpack(x6 a2, $dx);
my $sday=unpack(x9 a2, $dx);
my $shr=unpack(x12 a2, $dx);
$shr = $shr * 1;
my $smin=unpack(x15 a2, $dx);
my $ssec=unpack(x18 a2, $dx);
if ($smon == 1 $sday 27) {
$sday=27;
}
if ($sday == 31 ($smon == 3 || $smon == 10 || $smon == 5 || $smon ==
8)) {
$sday=30;
}
my $stime = timelocal($ssec,$smin,$shr,$sday,$smon,$syr);
my $etime = $stime + $dur;
$call_start{$call_max}=$stime;
$call_end{$call_max}=$etime;
$call_conhour{$call_max}=$shr;
$call_con{$call_max}=0;
$call_max++;
}
close $cdr_in;
for (my $i=0;$i$call_max;$i++) {
my $test_start=$call_start{$i}-1;
my $test_end=$call_end{$i}+1;
for (my $j=0;$j$call_max;$j++) {
if ($call_start{$j}$test_start $call_end{$j} $test_end) {;
$call_con{$i}++;
}
}
}
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Wednesday, August 03, 2011 4:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Know the number of concurrent dial ?
Hi
I connected Asterisk 1.6 has several SIP provider, Do you know a tool to
make a graph of the number of simultaneous calls incoming and outgoing ? and
know the max outgoing call in same time ?
thanks
Olivier.
--
_
-- 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
--
_
-- 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] Know the number of concurrent dial ?
Hi I connected Asterisk 1.6 has several SIP provider, Do you know a tool to make a graph of the number of simultaneous calls incoming and outgoing ? and know the max outgoing call in same time ? thanks Olivier. -- _ -- 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 = Request a Code
Hi i want add a numeric password to a call in : User call to a number, Asterisk answer and request: please insert your pin code the user enter a numeric code of 4 number and # when asterisk have the code, he start a api. Anyone have a sample of extension.conf for this ? thanks Olivier -- _ -- 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] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
very thanks, that's work
bye
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
I gave you the syntax in ael format, if you want to use extensions.conf
you'll have to use the syntax that's applicable, which is:
[start-audio]
exten = s,1,Playback(silence/1)
On 04/03/11 14:14, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawskimarkm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes
the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten}
])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten =
_00339,7,Set(CALLERPRES()=prohib_not_screened)
exten =
_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
--
--
_
-- 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
[asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct, asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
--
_
-- 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] Asterisk 1.6 = No sound/voice when i redirect the call
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawski markm-li...@intellasoft.net:
In that situation, I've had to do a pickup macro that kind of primes the
audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s = {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _00339,6,AGI(Ddi-Network.agi,${toexten})
exten = _00339,7,Set(CALLERPRES()=prohib_not_screened)
exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt)
exten = _00339,9,Hangup
and i have in sip.conf:
[MyOperator]
type=peer
host=host-of-my-operator
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=g729
allow=alaw
allow=g723
defaultuser=0033xx
secret=x
When i call directly from [MyOperator], no probleme i have sound/Voice
but when a customer call to the 00339xxx.., the call are correct,
asterisk
call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice
(i receive the call without problems, only sound off)
anyone have a idea of this problems ?
bye
Olivier
--
_
-- 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
--
_
-- 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] Problems Extension with a Call In on Asterisk 1.6
Hi
Very thanks for your helps, that's work very goo
Bye
Olivier
2011/3/25 DHAVAL INDRODIYA dhaval.it01...@gmail.com:
Hi Olivier,
here is solutions for your situation , ideally you need to talk with
Provider and they can set SIP URI
for given DID numbre , but that can be solved by dial-plan like this.
exten = _003318364,1,Set(foo=${SIP_HEADER(To)})
exten = _003318364,n,Set(cut1=${CUT(foo,:,2)})
exten = _003318364,n,Set(CLI=${CUT(cut1,,1)})
exten = _003318364,n,Set(toexten=${CUT(CLI,@,1)})
exten = _003318364,n,Noop(ORIGINAL NUMBER : [ ${toexten} ])
exten = _003318364,n,ExecIf($[${toexten} =
81169]?Dial(SIP/204,180,rt):Noop(${toexten}))
exten = _003318364,n,ExecIf($[${EXTEN} =
003318364]?Dial(SIP/203,180,rt):Noop(${toexten}))
On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
Anyone know a solution at my problems ?
Thanks
Olivier
2011/3/23 Olivier CALVANO o.calv...@gmail.com:
Hi
I request your help because i don't have actually a solution at my
problems.
I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers:
003318364 (official number)
081169 (Nddi Number)
When i receive a call on the 081169, he don't use
the extension. He use the 003318364 extension.
SIP Debug:
--- SIP read from UDP://91.121.xxx.xxx:5060 ---
INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From: 033426aa
sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
To: sip:081169x...@91.121.xxx.xxx;user=phone
User-Agent: Cirpack/v4.42s (gw_sip)
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
Content-Length: 481
v=0
o=cp10 130085910854 130085910854 IN IP4 10.7.1.121
s=SIP Call
c=IN IP4 91.121.bbb.bbb
t=0 0
m=audio 36146 RTP/AVP 18 4 0 8 125 111 101
b=AS:21
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000/1
a=fmtp:4 annexa=no
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:125 CLEARMODE/8000/1
a=rtpmap:111 iLBC/8000/1
a=fmtp:111 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=sqn:0
a=cdsc: 1 image udptl t38
-
--- (13 headers 22 lines) ---
Sending to 91.121.xxx.xxx : 5060 (no NAT)
Using INVITE request as basis request -
04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 125
Found RTP audio format 111
Found RTP audio format 101
Peer audio RTP is at port 91.121.bbb.bbb:36146
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CLEARMODE for ID 125
Found audio description format iLBC for ID 111
Found audio description format telephone-event for ID 101
Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d
(g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),
combined - 0x109 (g723|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 91.121.bbb.bbb:36146
Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx)
--- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx
From: 033426aa
sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
CSeq: 1602837515 INVITE
Server: Asterisk PBX 1.6.1.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
[Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527
handle_request_invite: Call from '0033459aa' to extension
'003318364' rejected because extension not found.
Scheduling destruction of SIP dialog
'04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method:
INVITE)
--- SIP read from UDP://91.121.xxx.xxx:5060 ---
ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
CSeq: 1602837515 ACK
From: 033426aa
sip:033426aaa
[asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 0 I see in the debug: To: sip:081169x...@91.121.xxx.xxx;user=phone but he search the 003318364 extension [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Anyone know the solution for he use the extension based on the To: ? thanks Olivier -- _ -- 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] Problems Extension with a Call In on Asterisk 1.6
Hi Dhaval, Thanks for your answer, but i not my question ;=) My asterisk have a entry into the sip.conf with a context. in extensions.conf, i have this extensions: exten = _003318364,1,Dial(SIP/203,180,rt) exten = _003381169,1,Dial(SIP/204,180,rt) (in my debug, i have deleted the exten = _003318364) When i call to 3318364 that's work When i call to 3381169 that's work but it's the _003318364 is used and phone 203 ring bye olivier 2011/3/23 DHAVAL INDRODIYA dhaval.it01...@gmail.com: Hi Oliver , This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add, context=(desired_context for peer) and then into context write a dial-plan for given number and route a call or whatever you want to do. On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B
Re: [asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi Anyone know a solution at my problems ? Thanks Olivier 2011/3/23 Olivier CALVANO o.calv...@gmail.com: Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 0 I see in the debug: To: sip:081169x...@91.121.xxx.xxx;user=phone but he search the 003318364 extension [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Anyone know the solution for he use the extension based on the To: ? thanks Olivier -- _ -- 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
[asterisk-users] SIP Invite and Asterisk API/Variable
Hi
I have in a SIP invite of a incoming call:
INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From: 033426aa
sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
To: sip:081169x...@91.121.xxx.xxx;user=phone
User-Agent: Cirpack/v4.42s (gw_sip)
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
Content-Length: 481
The To, To: sip:081169x...@91.121.xxx.xxx;user=phone, can i get it into
a variable for sent it at a API ?
Sample:
in extension.conf:
exten =
_003318364,1,AGI(Caller-ID_Phibee.agi,${CALLERID(name)},${VARIABLE})
exten = _003318364,2,Dial(SIP/185,180,rt)
in this sample, ${VARIABLE} = 081169
Thanks for your help
Olivier
--
_
-- 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] Help Asterisk / API / Perl
Hi
i want use the API on my asterisk 1.6, but i have a small problems :
In extension, i start it :
exten = _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems
but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)
but that's don't work:
use Asterisk::AGI;
use lib /var/lib/asterisk/agi-bin;
$AGI = new Asterisk::AGI;
$typ = $AGI-get_variable('agi_type');
$typ don't have SIP or IAX, same test without succes:
$typ = $AGI-get_variable('type');
anyone know this problems ?
thanks
Olivier
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, Sent accountcode between 2 asterisk
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
exten = _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt)
exten = _X.,3,Hangup
anyone know if it's possible to add the CDR Accountcode to this process
for get it on the second server B ?
i want the same accountcode on the 2 servers
thanks
Olivier
--
_
-- 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] Asterisk 1.6 and Username in Dial
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
--
_
-- 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 1.6 and Music on Hold
Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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] Asterisk 1.6 and Music on Hold
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com: On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi i have a small problems on Asterisk 1.6 with the MusiconOld : musiconhold.conf: [Sound_1] mode=quietmp3 directory=/var/lib/asterisk/moh/Sound_1 in extensions.conf: exten = 0532xx,1,Answer exten = 0532xx,2,MusicOnHold(Sound_1) exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t) exten = 0532xx,4,Hangup When i call to the number, i have the Music Sound_1 but the SIP Phone don't ring ... Where is my error ? and second question, can i said at asterisk that when he receive the call, he play the music from first second ? and repeat at the end of the music. Thanks for your help -- _ -- 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 First, if you don't use the Music on hold command prior to the dial, do you hear ringing? It seems to me that what's going on here is that you're overriding the progress notification that results from the device responding to the invite with TRYING or RINGING by running MOH. If the ringing doesn't occur even when you remove the MOH command, your device is probably not signaling properly and you'll need to use the r option in your Dial command. Hi Thanks for your help, yes, if i don't put the music on hold command, the phone ringing. I have change for put the r but no effect bye olivier -- _ -- 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 1.6 and Username in Dial
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
--
_
-- 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] Audiocode Median 2000 Gateway with Asterisk ?
Anyone have a AudioCodes with Asterisk ??? 2010/9/18 Olivier CALVANO o.calv...@gmail.com: Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I don't want specify numbers on the audiocode, a +33* = Asterisk. Thanks for your help Olivier -- _ -- 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] Audiocode Median 2000 Gateway with Asterisk ?
Hi i have buy a Audiocode Median 2000 VoIP Gateway and connect it on : 1 E1 30 channels 1 Lan Port Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). I want that all calls arrives on the AudioCode are sent to the asterisk by SIP (trunk ?) and all outgoing call from Asterisk are sent to the AudioCode. I don't want specify numbers on the audiocode, a +33* = Asterisk. Thanks for your help Olivier -- _ -- 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] Audiocode Median 2000 Gateway with Asterisk ?
Sorry i don't really understand your message ;=) my english are bad. I am search a sample of configuration of the audiocode. 2010/9/18 Paul Belanger paul.belan...@polybeacon.com: On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO o.calv...@gmail.com wrote: Anyone use this equipements with asterisk ? because i am search a config sample for AudioCode and for Asterisk (i am new in VoIP). Why would you want too? Asterisk can do everything, and more, then the Audiocodes. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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
[asterisk-users] Problems with Asterisk and two Linksys SPA941
Hi I have a big problems on my Asterisk systems : I have one Asterisk Server with static IP (no nat) and 6 Linksys SPA941. All SPA are after a router with NAT: * SPA-1 and SPA-2 are on the same network, we have a pat 5060 = SPA-1 and 5061= SPA-2 on the internet router * SPA-3, we have a pat 5062 = SPA-3 * SPA-4, we have a pat 5063 = SPA-4 * SPA-5, we have a pat 5064 = SPA-5 * SPA-6, we have a pat 5065 = SPA-6 On the Asterisk Sip conf, we have nat=yes and dynamic host. The problems are SPA-1 and SPA-2 can call to all other SPA except SPA-3 with SPA-3, i speak, it's good, spa-3 have the sound, but spa-3 speak i don't have the sound. SPA-3 can speak with SPA-4,5 and 6 without problems a idea of the problems ? Thanks Bye -- _ -- 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] Gateway E1 = Asterisk ?
Hi i want change my asterisk server. Actually, Asterisk work's on a IBM Server with a internal digium E1 card. For High availability, i don't want now use internal E1 card. In my new asterisk systems, i have two server and two E1 not in the same site. I am search a hardware gateway, if possible in 1U Rack with 2/4 or 8 E1 capacity with echo cancellation. I want that this gateway connect in trunk sip to my asterisk. Anyone have idea of good products for this ? Redfone ? but no SIP i thnk's, only in MAC/Ethernet Patton ? Not in rack other ? thanks for your help Olivier -- _ -- 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
