[asterisk-users] AGI Dial channel status
Hi guys, I'm writing an AGI program and use EXEC DIAL to do the dialing. The result reply from Asterisk doesn't come back until the dialing times out, or the channel is hung up after the call is connected with remote party and finally hung up. Is it possible to tell from the return code, or whaterver other ways, that the EXEC DIAL went through and the call was connected? Or that the dialing timed out? Thanks, S.P ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to limit the duration of the MeetMe conversation?
In article [EMAIL PROTECTED], Dima Pursanov [EMAIL PROTECTED] wrote: How to limit the duration of the MeetMe conversation? The easiest way is to set an absolute timeout on each participant before they enter, by using AbsoluteTimeout(xxx) or the newer Set(TIMEOUT(absolute)=xxx) You will obviously have to calculate the value of the timeout xxx for each participant according to their time of entry. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] php agi trixbox help
I have this code which was taken from the phpagi project page along with the
following in extensions_conf and the output from the asterisk CLI. When I
call the 311 extension, I does nothing then hangs up. What am I doing
wrong??
php code
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}. Let's enter some text.);
$text = $agi-text_input('UPPERCASE');
$agi-text2wav(You entered $text);
$agi-text2wav('Goodbye');
$agi-hangup();
?
--extensions_custom.php
exten = 311,1,Answer
exten = 311,2 Wait(1)
exten = 311,3,DigitTimeout(7)
exten = 311,4,ResponseTimeout(10)
exten = 311,5,AGI(input.php)
--CLI output --
-- Executing Answer(SIP/200-09b20488, ) in new stack
== Spawn extension (from-internal, 311, 2) exited non-zero on
'SIP/200-09b20488'
-- Executing Macro(SIP/200-09b20488, hangupcall) in new stack
-- Executing ResetCDR(SIP/200-09b20488, w) in new stack
-- Executing NoCDR(SIP/200-09b20488, ) in new stack
-- Executing Wait(SIP/200-09b20488, 5) in new stack
-- Executing Hangup(SIP/200-09b20488, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488'
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RE: [asterisk-users] I cant install zaptel drivers in suse 10.1
Marco, Did you install the kernel sources? the messages that you wrote are telling that you don't have the sources. Regards, _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Torrez Sent: Tuesday, December 26, 2006 6:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] I cant install zaptel drivers in suse 10.1 Hi, All How do I install Zaptel drivers on a system running Suse? Make results: grep: /lib/modules/2.6.16.13-4-smp/build/include/linux/autoconf.h: No existe el fichero o el directorio make[1]: Entering directory /usr/src/asterisk/zaptel/zaptel- 1.4.0-beta2 make -C /lib/modules/2.6.16.13-4-smp/build SUBDIRS=/usr/src/asterisk/zaptel/zaptel-1.4.0-beta2 modules make[2]: Entering directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[2]: *** No hay ninguna regla para construir el objetivo modules. Altc. Make[2]: Leaving directory /usr/src/linux-2.6.16.13-4-obj/i386/smp make[1]: *** [linux26] Error 2 make[1]: Leaving directory /usr/src/asterisk//zaptel/zaptel-1.4.0-beta2 make: *** [all] Error 2 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with extentions
i have problem with dial-plan in php. I have 3 extention in dial plan, 555,551 and 551. the problem is that READ PIN||3 works in 555 and hangs in other extentions. (after timeout asterisk writes thar user entered nothing). i can't get what's wrong... here is my dial-plan [incoming] exten = 555,1,Answer(); exten = 555,2,DigitTimeout(2); exten = 555,3,Wait(1) exten = 555,4,SetLanguage(ge) exten = 555,5,Background(mosalmebadsl); exten = 555,6,Hangup(); exten = 1,1,Goto(register,s,1) exten = 2,1,Goto(viocemail,s,1) exten = 3,1,Goto(sendmsg,s,1) exten = 4,1,Goto(blacklist,s,1) exten = 5,1,Goto(search,s,1) exten = 6,1,Goto(rooms,s,1) exten = 7,1,Goto(adv,s,1) exten = i,1,Hangup() exten = 551,1,Answer() exten = 551,2,DigitTimeout(2) exten = 551,3,Wait(1) exten = 551,4,SetLanguage(ge) exten = 551,5,agi,admin_room.php exten = 551,6,Hangup() exten = 552,1,Answer() exten = 552,2,DigitTimeout(2) exten = 552,3,Wait(1) exten = 552,4,SetLanguage(ge) exten = 552,5,agi,guest_room.php exten = 552,6,Hangup() so when i call admin_room.php from 555 it works, but it crashes from 551. (guest_room.php also crashes in 552 and works in 555) --- thank You ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.4.0 (release) and G.729
There was a new format_g729.so but there was no codec_g729a.so, so I put the right one for the x86_64 kernel. I choose that on the make menuselect already. After your question, I saw in the messages the report that codec_g729a.so was not able to load, so I moved out. No messages about g729 after stop And reinitiate it. I don't see the module on messages, and there is no translation between g729 and lin or slin. Thanks for you overnight counseling. Regards, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 601 Contacts List
Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Contacts List
I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Agent presence
You could put together a web page that talks to the Asterisk Manager. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent presence Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: php agi trixbox help
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown on line 0
Thanks.
On 12/27/06, blackwater dev [EMAIL PROTECTED] wrote:
I have this code which was taken from the phpagi project page along with
the following in extensions_conf and the output from the asterisk CLI. When
I call the 311 extension, I does nothing then hangs up. What am I doing
wrong??
php code
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}. Let's enter some text.);
$text = $agi-text_input('UPPERCASE');
$agi-text2wav(You entered $text);
$agi-text2wav('Goodbye');
$agi-hangup();
?
--extensions_custom.php
exten = 311,1,Answer
exten = 311,2 Wait(1)
exten = 311,3,DigitTimeout(7)
exten = 311,4,ResponseTimeout(10)
exten = 311,5,AGI( input.php)
--CLI output --
-- Executing Answer(SIP/200-09b20488, ) in new stack
== Spawn extension (from-internal, 311, 2) exited non-zero on
'SIP/200-09b20488'
-- Executing Macro(SIP/200-09b20488, hangupcall) in new stack
-- Executing ResetCDR(SIP/200-09b20488, w) in new stack
-- Executing NoCDR(SIP/200-09b20488, ) in new stack
-- Executing Wait(SIP/200-09b20488, 5) in new stack
-- Executing Hangup(SIP/200-09b20488, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488'
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RE: [asterisk-users] Asterisk 1.4.0 (release) and G.729
Ok, I realize that by mistake I put the old codec_g729a.so instead of the one belonging to the 1.4 version. (However it worked on the beta). As soon as I changed it, now I got this on the /var/log/asterisk/messages [Dec 27 10:24:25] WARNING[7512] translate.c: plc_samples 160 format 6 [Dec 27 10:24:25] WARNING[7512] translate.c: plc_samples 160 format 6 [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: G.729 transcoding module Copyright (C) 1999-2006 Digium, Inc. [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: This module is supplied under a commercial license granted by Digium, Inc. [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: Please see the full license text supplied by the accompanying [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: register utility, or ask for a copy from Digium. [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: This product includes software developed by the OpenSSL Project [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: for use in the OpenSSL Toolkit. (http://www.openssl.org/) [Dec 27 10:24:25] NOTICE[7512] codec_g729.c: Copyright (C) 1998-2006 The OpenSSL Project [Dec 27 10:24:25] WARNING[7512] translate.c: plc_samples 160 format 6 [Dec 27 10:24:25] WARNING[7512] translate.c: plc_samples 160 format 6 However in core show translation I get fedora5*CLI core show translation Translation times between formats (in milliseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- --- - gsm- -222 21 4- - 142 - ulaw- 2-12 21 4- - 142 - alaw- 21-2 21 4- - 142 - g726aal2- 222- 21 4- - 141 - adpcm- 2222 -1 4- - 142 - slin- 1111 1- 3- - 131 - lpc10- 3333 32 -- - 153 - g729- ---- -- -- --- - speex- ---- -- -- --- - ilbc- 3333 32 5- --3 - g726- 2221 21 4- - 14- - g722- ---- -- -- --- - And still nothing in the CLI fedora5*CLI ! abort add ael agent agi answer autoanswer cdr clear console convert coredatabasedebug dialplandial dnsmgr dontdumpdundi extensions feature filegroup hangup helphttpiax2 include indication initkeys loadlocal logger manager meetme mgcp mixmonitor module moh no originate pri queue realtimereload remove restart rtcp rtp savesay send set showsip skinny sla softstop stunudptl unload voicemail Zap Or in show fedora5*CLI show agentsagi application applications audio channel channels channeltype channeltypes codec codecsconfigdialplan features file frame function functions globals hints image indications keys manager modules parkedcalls profile queuesqueue switches translation version video voicemail There is no more references to g729 as it was on the past. And it seems that there is no translation codec also. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent presence
Not quite the solution I was looking for - I was wanting the agent's status to be reflected in it's presence hint. I'm somewhat inclined to believe that 1.2 isn't going to do the job at this stage since I don't think it supports SIP presence to the degree required. Douglas Garstang wrote: You could put together a web page that talks to the Asterisk Manager. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent presence Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Verbose and sip invitestate logging question (1.4 release)
All, I just installed latest Asterisk 1.4 release today and notice two problems that I couldn’t figure it out how to resolve it. SIP invitestate keep showing up in console. I am unable to stopping it and I even tried to do core set verbose off and core set debug off. What else do I have to do to stop this? chan_sip1 sip_hangup flags invitestate 5 0xa0c001c data INVITE chan_sip1 sip_hangup flags now 0xa0c001c Verbose logging doesn’t since to work for me after loggers reload/rotate, but full, error, and console logging is working fine. Did this happen to anyone else? Logger.conf [general] dateformat=%F %T appendhostname = no queue_log = no event_log = no [logfiles] ;debug = debug ;console = notice,warning,error,debug messages = error ;full = notice,warning,debug,verbose logger show channels Channel Type StatusConfiguration --- --- /var/log/asterisk/full File Enabled- Debug Verbose Warning Notice /var/log/asterisk/messages File Enabled- Error Sincerely, KC -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.28/604 - Release Date: 12/26/2006 12:23 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: php agi trixbox help
Don't know if this will do it (see your full logs for details), but
the timeout lines in your 311 are at least depricated if removed --
use the set statement and functions like this
exten = s,n,Set(TIMEOUT(digit)=5)
Hope this helps.
on Wednesday 12/27/2006 blackwater dev([EMAIL PROTECTED]) wrote
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file:
No such file or directory in Unknown on line 0
Thanks.
On 12/27/06, blackwater dev [EMAIL PROTECTED] wrote:
I have this code which was taken from the phpagi project page along with
the following in extensions_conf and the output from the asterisk CLI.
When
I call the 311 extension, I does nothing then hangs up. What am I doing
wrong??
php code
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}. Let's enter some text.);
$text = $agi-text_input('UPPERCASE');
$agi-text2wav(You entered $text);
$agi-text2wav('Goodbye');
$agi-hangup();
?
--extensions_custom.php
exten = 311,1,Answer
exten = 311,2 Wait(1)
exten = 311,3,DigitTimeout(7)
exten = 311,4,ResponseTimeout(10)
exten = 311,5,AGI( input.php)
--CLI output --
-- Executing Answer(SIP/200-09b20488, ) in new stack
== Spawn extension (from-internal, 311, 2) exited non-zero on
'SIP/200-09b20488'
-- Executing Macro(SIP/200-09b20488, hangupcall) in new stack
-- Executing ResetCDR(SIP/200-09b20488, w) in new stack
-- Executing NoCDR(SIP/200-09b20488, ) in new stack
-- Executing Wait(SIP/200-09b20488, 5) in new stack
-- Executing Hangup(SIP/200-09b20488, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488'
Not sure if this has anything to do with it but running the input.php script
directly from the command line gives this warning:brbrPHP Warning:nbsp;
Unknown(): Unable to load dynamic library #39;/usr/lib/php4/imap.so#39; -
libc-client.so.0: cannot open shared object file: No such file or directory
in Unknown on line 0brbrThanks.brbrdivspan class=gmail_quoteOn
12/27/06, b class=gmail_sendernameblackwater dev/b lt;a
href=mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED]/agt; wrote:/spanblockquote class=gmail_quote
style=border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex;
padding-left: 1ex;I have this code which was taken from the phpagi project
page along with the following in extensions_conf and the output from the
asterisk CLI.nbsp; When I call the 311 extension, I does nothing then hangs
up.nbsp; What am I doing wrong??
brbrphp codebrbr#!/usr/local/bin/php
-qbrlt;?phpbrnbsp; set_time_limit(30);brnbsp;
require(#39;phpagi.php#39;);brbrnbsp; $agi = new AGI();brnbsp;
$agi-gt;answer();brnbsp; $cid = $agi-gt;parse_callerid();
brnbsp; $agi-gt;text2wav(quot;Hello, {$cid[#39;name#39;]}.nbsp;
Let#39;s enter some text.quot;);brnbsp; $text =
$agi-gt;text_input(#39;UPPERCASE#39;);brnbsp;
$agi-gt;text2wav(quot;You entered $textquot;);brnbsp;
$agi-gt;text2wav(#39;Goodbye#39;);
brnbsp;
$agi-gt;hangup();br?gt;brbr--extensions_custom.phpbrexten
=gt; 311,1,Answerbrexten =gt; 311,2 Wait(1)brexten =gt;
311,3,DigitTimeout(7)brexten =gt; 311,4,ResponseTimeout(10)brexten
=gt; 311,5,AGI(
input.php)brbr--CLI output --brnbsp;--
Executing Answer(quot;SIP/200-09b20488quot;, quot;quot;) in new
stackbrnbsp; == Spawn extension (from-internal, 311, 2) exited non-zero
on #39;SIP/200-09b20488#39;
brnbsp;nbsp;nbsp; -- Executing Macro(quot;SIP/200-09b20488quot;,
quot;hangupcallquot;) in new stackbrnbsp;nbsp;nbsp; -- Executing
ResetCDR(quot;SIP/200-09b20488quot;, quot;wquot;) in new
stackbrnbsp;nbsp;nbsp; -- Executing NoCDR(quot;SIP/200-09b20488quot;,
quot;quot;) in new stack
brnbsp;nbsp;nbsp; -- Executing Wait(quot;SIP/200-09b20488quot;,
quot;5quot;) in new stackbrnbsp;nbsp;nbsp; -- Executing
Hangup(quot;SIP/200-09b20488quot;, quot;quot;) in new stackbrnbsp; ==
Spawn extension (macro-hangupcall, s, 4) exited non-zero on
#39;SIP/200-09b20488#39; in macro #39;hangupcall#39;
brnbsp; == Spawn extension (macro-hangupcall, s, 4) exited non-zero on
#39;SIP/200-09b20488#39;
/blockquote/divbr
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Re: [asterisk-users] Re: php agi trixbox help
If you want to install the php imap support it is usually doen according
to the linux distro you use. On debian the package would be php4-imap.
If you don't want to install it you need to make sure your php and
phpagi config don't require it.
blackwater dev wrote:
Not sure if this has anything to do with it but running the input.php
script directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object
file: No such file or directory in Unknown on line 0
Thanks.
On 12/27/06, *blackwater dev* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have this code which was taken from the phpagi project page
along with the following in extensions_conf and the output from
the asterisk CLI. When I call the 311 extension, I does nothing
then hangs up. What am I doing wrong??
php code
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}. Let's enter some text.);
$text = $agi-text_input('UPPERCASE');
$agi-text2wav(You entered $text);
$agi-text2wav('Goodbye');
$agi-hangup();
?
--extensions_custom.php
exten = 311,1,Answer
exten = 311,2 Wait(1)
exten = 311,3,DigitTimeout(7)
exten = 311,4,ResponseTimeout(10)
exten = 311,5,AGI( input.php)
--CLI output --
-- Executing Answer(SIP/200-09b20488, ) in new stack
== Spawn extension (from-internal, 311, 2) exited non-zero on
'SIP/200-09b20488'
-- Executing Macro(SIP/200-09b20488, hangupcall) in new stack
-- Executing ResetCDR(SIP/200-09b20488, w) in new stack
-- Executing NoCDR(SIP/200-09b20488, ) in new stack
-- Executing Wait(SIP/200-09b20488, 5) in new stack
-- Executing Hangup(SIP/200-09b20488, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488'
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RE: [asterisk-users] Agent presence
Wasn't Olle Johansen working on something that would allow (polycom phones at least) to show the status of agents on the phone... -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Agent presence Not quite the solution I was looking for - I was wanting the agent's status to be reflected in it's presence hint. I'm somewhat inclined to believe that 1.2 isn't going to do the job at this stage since I don't think it supports SIP presence to the degree required. Douglas Garstang wrote: You could put together a web page that talks to the Asterisk Manager. -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 26, 2006 11:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Agent presence Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on pause. I'm using chan_agent for the agents, so agents are logged in and out using AgentCallbackLogin (I know it's deprecated in 1.4, but it's working well for us at the moment) and the agents are put on pause using PauseQueueMember and UnpauseQueueMember. I've figured out I can show whether an agent is logged in or out by creating a dummy extension with a hint as follows:- exten = 151,1,Dial(Agent/151) exten = 151,hint,Agent/151 X-Lite quite happily shows the agent as Ready when they're logged in, unavailable when logged out and On the Phone when (funnily enough) they're taking a call. However, when the agent is on pause, they are still shown as Ready. Is this a limitation of chan_agent, Pause/UnpauseQueueMember, Asterisk 1.2's presence support, or is there something else I can do in order to get the agent shown indicated as something other than Ready when they're on pause? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verbose and sip invitestate logging question (1.4 release)
KC wrote: I just installed latest Asterisk 1.4 release today and notice two problems that I couldn’t figure it out how to resolve it. SIP invitestate keep showing up in console. I am unable to stopping it and I even tried to do core set verbose off and core set debug off. What else do I have to do to stop this? chan_sip1 sip_hangup flags invitestate 5 0xa0c001c data INVITE chan_sip1 sip_hangup flags now 0xa0c001c These messages do not appear in Asterisk 1.4, they appear in Asterisk SVN trunk (the development branch). You installed the wrong version of Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Verbose and sip invitestate logging question(1.4 release)
Lol my mistake.. Thanks Kevin.. Sincerely, KC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Wednesday, December 27, 2006 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Verbose and sip invitestate logging question(1.4 release) KC wrote: I just installed latest Asterisk 1.4 release today and notice two problems that I couldn’t figure it out how to resolve it. SIP invitestate keep showing up in console. I am unable to stopping it and I even tried to do core set verbose off and core set debug off. What else do I have to do to stop this? chan_sip1 sip_hangup flags invitestate 5 0xa0c001c data INVITE chan_sip1 sip_hangup flags now 0xa0c001c These messages do not appear in Asterisk 1.4, they appear in Asterisk SVN trunk (the development branch). You installed the wrong version of Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.28/604 - Release Date: 12/26/2006 12:23 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.15.28/604 - Release Date: 12/26/2006 12:23 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: php agi trixbox help
Not sure if this has anything to do with it but running the input.php script directly from the command line gives this warning: PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 yum install php-imap hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent presence
Hints for agents work just fine insofar as showing when the agent is not online (logged off), Ready (logged on) or on a call, but nothing when the line is ringing or when the agent has been paused... at least not in Asterisk 1.2.13. I'm not quite ready to take the somewhat significant plunge into 1.4 yet, though I may expend more effort in this direction if chan_agent hasn't been completely deprecated and hints include this extra information. Douglas Garstang wrote: Wasn't Olle Johansen working on something that would allow (polycom phones at least) to show the status of agents on the phone... -Original Message- From: Rob Hillis [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Agent presence Not quite the solution I was looking for - I was wanting the agent's status to be reflected in it's presence hint. I'm somewhat inclined to believe that 1.2 isn't going to do the job at this stage since I don't think it supports SIP presence to the degree required. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Contacts List
There is an index in the configuration file which I believe it will obey. I'll try and find it later if you haven't found it by the time I get to the office. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, December 27, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Searching the list
Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 601 Contacts List
The directory file has an sd Speed dial index tag. The phone honours this index when displaying entries on the LCD screen and when the up arrow is pressed. However, it does not honor this order, and instead displays entries in alphabetical order, when you press the 'Directories' button. -Original Message- From: Jonathan k. Creasy [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List There is an index in the configuration file which I believe it will obey. I'll try and find it later if you haven't found it by the time I get to the office. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, December 27, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searching the list
Mark Greene wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) http://lists.digium.com/mailman/listinfo/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searching the list
Hi Mark, I don't think there's a built in search (someone please correct me if I'm mistaken here), but Google can filter results for you: site:http://lists.digium.com/pipermail/asterisk-users/ searchterm Alex On 12/27/06, Mark Greene [EMAIL PROTECTED] wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Searching the list
I'm not sure if there is a more official method but Google has always been my friend when searching the lists. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene Sent: Wednesday, December 27, 2006 12:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Searching the list Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searching the list
On Wed, Dec 27, 2006 at 11:05:29AM -0600, Mark Greene wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) http://gmane.org/find.php?list=asterisk This one is http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk.user There's a search box there. And here are some people who asked the same question in the past: http://search.gmane.org/?query=search+this+listgroup=gmane.comp.telephony.pbx.asterisk.user -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Searching the list
You can only search a month at a time... :( -Original Message- From: Doug Lytle [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Searching the list Mark Greene wrote: Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) http://lists.digium.com/mailman/listinfo/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and Queues
Ed,
Thanks for the help. One more question, however. Everything is working
fine with the exception of sox joining the calls. I have sox installed
and monitor-join set to yes in both queues.conf and agents.conf
I installed sox after I installed Asterisk. Do I need to recompile
Asterisk for it to work with sox?
This is the last hurdle I need to overcome (I hope) before I can use my
Asterisk box in a live situation. Any help would be much appreciated.
Regards,
Jay
Ed Nuñez wrote:
In queues.conf you must have the following under the queues you want to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
thiswill join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be
recorded.
That's all
If you want to change the file name place this in your extensions.conf on a
line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once the
call comes out of queue (no point in recording hold music, unless I want
to hear people mumble about how lousy a company we are for placing them
on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I ask
is that I'd like to put the receiving extension in the name of the file
that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to record
the call when it's pulled out of either queue using MixMonitor. Bonus
points if I can determine the answering extension before MixMonitor
starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
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Re: [asterisk-users] Re: php agi trixbox help
Thanks, I'll try. I'm using the trixbox and my 311 info was in
extensions_custom.conf if that means anything.
On 12/27/06, John covici [EMAIL PROTECTED] wrote:
Don't know if this will do it (see your full logs for details), but
the timeout lines in your 311 are at least depricated if removed --
use the set statement and functions like this
exten = s,n,Set(TIMEOUT(digit)=5)
Hope this helps.
on Wednesday 12/27/2006 blackwater dev([EMAIL PROTECTED]) wrote
Not sure if this has anything to do with it but running the input.phpscript
directly from the command line gives this warning:
PHP Warning: Unknown(): Unable to load dynamic library
'/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object
file:
No such file or directory in Unknown on line 0
Thanks.
On 12/27/06, blackwater dev [EMAIL PROTECTED] wrote:
I have this code which was taken from the phpagi project page along
with
the following in extensions_conf and the output from the asterisk
CLI. When
I call the 311 extension, I does nothing then hangs up. What am I
doing
wrong??
php code
#!/usr/local/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-text2wav(Hello, {$cid['name']}. Let's enter some text.);
$text = $agi-text_input('UPPERCASE');
$agi-text2wav(You entered $text);
$agi-text2wav('Goodbye');
$agi-hangup();
?
--extensions_custom.php
exten = 311,1,Answer
exten = 311,2 Wait(1)
exten = 311,3,DigitTimeout(7)
exten = 311,4,ResponseTimeout(10)
exten = 311,5,AGI( input.php)
--CLI output --
-- Executing Answer(SIP/200-09b20488, ) in new stack
== Spawn extension (from-internal, 311, 2) exited non-zero on
'SIP/200-09b20488'
-- Executing Macro(SIP/200-09b20488, hangupcall) in new stack
-- Executing ResetCDR(SIP/200-09b20488, w) in new stack
-- Executing NoCDR(SIP/200-09b20488, ) in new stack
-- Executing Wait(SIP/200-09b20488, 5) in new stack
-- Executing Hangup(SIP/200-09b20488, ) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/200-09b20488'
Not sure if this has anything to do with it but running the input.phpscript directly from
the command line gives this warning:brbrPHP
Warning:nbsp; Unknown(): Unable to load dynamic library
#39;/usr/lib/php4/imap.so#39; -
libc-client.so.0: cannot open shared object file: No such file or
directory in Unknown on line 0brbrThanks.brbrdivspan
class=gmail_quoteOn 12/27/06, b class=gmail_sendernameblackwater
dev/b lt;a href=mailto:[EMAIL PROTECTED]
[EMAIL PROTECTED]/agt; wrote:/spanblockquote
class=gmail_quote style=border-left: 1px solid rgb(204, 204, 204);
margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;I have this code which was
taken from the phpagi project page along with the following in
extensions_conf and the output from the asterisk CLI.nbsp; When I call the
311 extension, I does nothing then hangs up.nbsp; What am I doing wrong??
brbrphp codebrbr#!/usr/local/bin/php
-qbrlt;?phpbrnbsp; set_time_limit(30);brnbsp;
require(#39;phpagi.php#39;);brbrnbsp; $agi = new AGI();brnbsp;
$agi-gt;answer();brnbsp; $cid = $agi-gt;parse_callerid();
brnbsp; $agi-gt;text2wav(quot;Hello, {$cid[#39;name#39;]}.nbsp;
Let#39;s enter some text.quot;);brnbsp; $text =
$agi-gt;text_input(#39;UPPERCASE#39;);brnbsp;
$agi-gt;text2wav(quot;You entered $textquot;);brnbsp;
$agi-gt;text2wav(#39;Goodbye#39;);
brnbsp;
$agi-gt;hangup();br?gt;brbr--extensions_custom.phpbrexten
=gt; 311,1,Answerbrexten =gt; 311,2 Wait(1)brexten =gt;
311,3,DigitTimeout(7)brexten =gt; 311,4,ResponseTimeout(10)brexten
=gt; 311,5,AGI(
input.php)brbr--CLI output --brnbsp;--
Executing Answer(quot;SIP/200-09b20488quot;, quot;quot;) in new
stackbrnbsp; == Spawn extension (from-internal, 311, 2) exited non-zero
on #39;SIP/200-09b20488#39;
brnbsp;nbsp;nbsp; -- Executing Macro(quot;SIP/200-09b20488quot;,
quot;hangupcallquot;) in new stackbrnbsp;nbsp;nbsp; -- Executing
ResetCDR(quot;SIP/200-09b20488quot;, quot;wquot;) in new
stackbrnbsp;nbsp;nbsp; -- Executing NoCDR(quot;SIP/200-09b20488quot;,
quot;quot;) in new stack
brnbsp;nbsp;nbsp; -- Executing Wait(quot;SIP/200-09b20488quot;,
quot;5quot;) in new stackbrnbsp;nbsp;nbsp; -- Executing
Hangup(quot;SIP/200-09b20488quot;, quot;quot;) in new stackbrnbsp; ==
Spawn extension (macro-hangupcall, s, 4) exited non-zero on
#39;SIP/200-09b20488#39; in macro #39;hangupcall#39;
brnbsp; == Spawn extension (macro-hangupcall, s, 4) exited non-zero
on #39;SIP/200-09b20488#39;
/blockquote/divbr
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Re: [asterisk-users] Re: php agi trixbox help
Ok, I did that and I can now do a locate and do now see: /usr/lib/php4/imap.so /usr/lib/httpd/modules/mod_imap.so So I restarted apache, and tried to run my file from the command line and get the same error. I have another one that doesn't use the phpagi class and it works ok but throws the same warning so I don't think that is why the script doesn't seem to work...not sure. Thanks! On 12/27/06, Time Bandit [EMAIL PROTECTED] wrote: Not sure if this has anything to do with it but running the input.phpscript directly from the command line gives this warning: PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 yum install php-imap hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searching the list
On google, do your search terms site:lists.digium.com - Original Message - From: Mark Greene [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, December 27, 2006 11:05:29 AM GMT-0600 US/Central Subject: [asterisk-users] Searching the list Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] comand stun (for what) asterisk 1.4
I'm looking for some doc's what is new in 1.4 in deep for the comand stun debug neptun:/usr/src/asterik_pgk/asterisk-1.4.0/main # asterisk -r Asterisk 1.4.0, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.0 currently running on neptun (pid = 4390) Verbosity is at least 4 neptun*CLI stun debug no neptun*CLI stun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Warnings
I get the following warning when starting Asterisk 1.4. Does anyone know what these mean, and/or how I can get rid of them? [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show cache' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show channels' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show firmware' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show netstats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show peers' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show registry' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show stats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show threads' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show users' already registered (or something close enough) Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
Yes, timing (ie; ztdummy, if no hardware is installed) is still required for things like meetme or iax2 trunking. - David Thomas [EMAIL PROTECTED] wrote: Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Searching the list
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RE: [asterisk-users] Asterisk 1.4 Warnings
I got the same messages but as far as I can see my problems are still with translation codec problems between g726aal2, g726 and g729. The problem with the G729 is refered to registration, since I cannot see anything on the CLI referred to my license, however the message says that is there. Also, I have video working h264 on SIP but not on IAX. Happy new year, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Wednesday, December 27, 2006 1:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk 1.4 Warnings I get the following warning when starting Asterisk 1.4. Does anyone know what these mean, and/or how I can get rid of them? [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: translate.c:675 __ast_register_translator: plc_samples 160 format 6 [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show cache' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show channels' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show firmware' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show netstats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show peers' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show registry' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show stats' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show threads' already registered (or something close enough) [Dec 28 02:12:28] WARNING[3419]: cli.c:1434 __ast_cli_register: Command 'iax2 show users' already registered (or something close enough) Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
Do you have a zap section on the CLI? Just do ? And check if you have that. I have zaptel working on two machines with wtc1xxp and ztdummy. The one with the card doesn't show zap section, the other one with ztdummy does. I thought that both should show the section on the CLI. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Thomas Sent: Wednesday, December 27, 2006 1:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is ZTDUMMY still required with Asterisk 1.4? Is ztdummy still required with Asterisk 1.4 when no zaptel cards are available to use for timing? In all the beta releases I used to get a warning when Asterisk started up, saying that no timing device was found. The warning seems to have gone away with the full release of 1.4, which prompts the question... Is it still required? Does 1.4 do something different for timing? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.4.0, IMAP and Dovecot
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
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Re: [asterisk-users] Follow-me challenge
Re: [asterisk-users] Follow-me challengeWhat you can do is have the called
person press a digit to accept the call. If the user dosent then you can set
the h extension to update the call logs
- Original Message -
From: Eric Jacksch
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, December 20, 2006 11:47 PM
Subject: Re: [asterisk-users] Follow-me challenge
Sorry, didn't realize you were sending the call out on a Zap channel.
Yes, as soon as the call goes out a Zap channel it is answered as far as
Asterisk is concerned. I send out all my findme traffic via SIP.
On 2006-12-19 21:19, Chris Johnson [EMAIL PROTECTED] wrote:
On 12/18/06, Eric Jacksch [EMAIL PROTECTED] wrote:
Is the problem just when you don't answer the cell phone? Many cell
phones go to a voice announcement when they're turned off or not answered, and
Asterisk thinks the call has been answered. The other issue could be that your
gateway (asterisk1) is answering the call before the outbound leg is answered.
One workaround would be to use a macro that requires you to press a key to
accept the call on your cell. (See the M option to the dial command and
http://www.voip-info.org/wiki/view/Asterisk+tips+findme)
http://www.voip-info.org/wiki/view/Asterisk+tips+findme%29
Also, I see that you're using the r option - you might want to drop that.
I'm also not convinced that it will ever find 102,107 in your dialplan.
You might want to look at using ${DIALSTATUS} and making it a bit more explicit.
Cheers,
Eric
Dropped the r option and line 107.
The M option had the same result.
Based on another comment :
Is your other server patching through to a Zap channel (analog)?
If so, as soon as the dial goes out, an analog Zap channel is considered
answered, which could be your issue.
Doesn't sound like follow-me will work properly with an analog trunk
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Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
A PI that does asterisk on the side ?? WTF ?? - Original Message - From: C F [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, December 24, 2006 2:22 PM Subject: Fwd: [asterisk-users] The Good, Bad and Scam VoIP Providers I Find It Funny, So I Decided To Let Others Laugh As Well -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Sun, 24 Dec 2006 14:01:06 -0500 Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: [EMAIL PROTECTED] This is off the list C F, You are an ass Bret is a scammer you can take that to the bank from a PI. Sorry I never stated what I do for a living. Did I? I will be dealing with Bret. And 2007 is not going to be a good year for that scammer. So why are you hiding use a real email address. And a real name. Looks like you have an in with Bret Master of Cybercrimes May have to my homework on you to. What is you think? I really don't care if you if you trust me. Your reply is only a pop out trying to save your ass. Please stay on the POINT! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: Al, Nobody Cares About Your Problems With Bret. Most People Here Know And Trust Bret More Than They Do You. All You Have Done So Far Is Made A Fool Out Of Yourself. At This Point All I Can Think Of Is That If Bret Does Hold Some Of Your Money That It Is A Significant Amount And He Wont Ever Give It To You. Move On And Dont Make A Bigger Fool Out Of Yourself. Swallow Your Pride Its Not Fattening. For You I Can Say: Temper Is What Gets You Into Trouble Pride Is What Keeps You There. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 1:41:46 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Embarased like when you were caught wuth your pants down ? - Original Message - From: Al Bochter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 24, 2006 3:37 PM Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers Peter // I'm done with this. I thought we were discussing VoIP provider scams? You are the one posting massages that are off the subject I took your replys off the list. Please keep your posts on the subject ( Thank You ) :-) Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: Oh no, the game's up - Al's found my IP address. Wait - no he hasn't - he's found an IP address that belongs to McAfee Security in Spain - with whom I have no connection at all. (Hint: whois ip address) Those PI classes really paid off, Al. Supposing you had managed to find out one of my IP addresses (which isn't really too hard, I have NIC handles at ARIN and RIPE, and hold addresses on behalf of more than one major organisation), what were you going to do with it? I'm done with this. I thought we were discussing VoIP provider scams? On 24/12/06, Al Bochter [EMAIL PROTECTED] wrote: Peter, This is off the list? it looks like ip: 62.189.112.129 Country GB: Britain AM I close? Anyways This is off my point! And should not be posted to the list. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email Peter Bowyer wrote: This is getting funnier by the minute. Way to go, Al. On 24/12/06, C F [EMAIL PROTECTED] wrote: I Find It Funny, So I Decided To Let Others Laugh As Well -- Forwarded message -- From: Al Bochter [EMAIL PROTECTED] Date: Sun, 24 Dec 2006 14:01:06 -0500 Subject: Re: [asterisk-users] The Good, Bad and Scam VoIP Providers To: [EMAIL PROTECTED] This is off the list C F, You are an ass Bret is a scammer you can take that to the bank from a PI. Sorry I never stated what I do for a living. Did I? I will be dealing with Bret. And 2007 is not going to be a good year for that scammer. So why are you hiding use a real email address. And a real name. Looks like you have an in with Bret Master of Cybercrimes May have to my homework on you to. What is you think? I really don't care if you if you trust me. Your reply is only a pop out trying to save your ass. Please stay on the POINT! Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email C F wrote: Al, Nobody Cares About Your Problems With Bret. Most People Here Know And Trust Bret More Than They Do You. All You Have Done So Far Is Made A Fool Out Of Yourself. At This Point All I Can Think Of Is That If Bret Does Hold Some Of Your Money That It Is A Significant Amount And He Wont Ever Give It To You. Move On And Dont Make A Bigger Fool Out Of Yourself. Swallow Your Pride Its Not Fattening. For You I Can Say: Temper Is What Gets You Into Trouble Pride Is What Keeps You There. On 12/24/06, Al Bochter [EMAIL PROTECTED] wrote: So you would deal with a criminal ? Bret McDanel was *Convicted Of Cybercrimes * Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VoIP PBX) 1-563-773-6610 EXT: 250 Peter Bowyer wrote: On 23/12/06, Al Bochter [EMAIL PROTECTED] wrote: We have to put the SCAMMERS like trxtel.com out of business (That don't pay there users) You know, I'd deal with a professional like Bret a thousand times before I considered dealing with a mom-and-pop lemonade stall like you. And this kind of posting will only move you further down the list. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0662-1, 12/24/2006 - 12/24/2006 1:41:46 PM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.0, IMAP and Dovecot
Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
I've been attempting the same with Cyrus and get the same results. The
interesting thing is if I take the same string (like
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX) and plug it
into the 'mtest' command from the c-client package, it works OK. I have
not tried this with the production release with Asterisk. Only beta's 1-4.
Mark
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Re: [asterisk-users] Is ZTDUMMY still required with Asterisk 1.4?
On 12/27/06, Carlos Alperin [EMAIL PROTECTED] wrote: Do you have a zap section on the CLI? Just do ? And check if you have that. I have zaptel working on two machines with wtc1xxp and ztdummy. The one with the card doesn't show zap section, the other one with ztdummy does. I thought that both should show the section on the CLI. No, I do not have zap listed when I type ? at the CLI. Does this indicate a problem? David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.0, IMAP and Dovecot
As an FYI, this has absolutely NOTHING AT ALL to do with Asterisk.
Asterisk is a PBX application. Dovcot is an email application. They have
nothing to do with each other. Asterisk is not a Linux distribution or
operating system.
I suggest that you ask your question on a more appropriate mailing list,
but it seems as though you are so thourougly confused as to what is
what, that you should probably pick up a beginners book on Linux and go
from there. I know this may sound harsh, but trust me: you will be much
better off in the long run if you educate yourself somewhat.
To give you an idea on how far off you are, it would be like going into
a car repair shop and asking about a furnace problem. While someone
there may be able to help you, you went to the wrong place.
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin said:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
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--
Walt Reed
[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
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Re: [asterisk-users] 1.4.0, IMAP and Dovecot
Dan, Please accept my sincerest appology. I had my head thoroughly up my
back orifice. I haven't kept up with the new IMAP feature in 1.4
I'll go back in my corner now :-)
On Wed, Dec 27, 2006 at 02:19:07PM -0500, Walt Reed said:
As an FYI, this has absolutely NOTHING AT ALL to do with Asterisk.
Asterisk is a PBX application. Dovcot is an email application. They have
nothing to do with each other. Asterisk is not a Linux distribution or
operating system.
I suggest that you ask your question on a more appropriate mailing list,
but it seems as though you are so thourougly confused as to what is
what, that you should probably pick up a beginners book on Linux and go
from there. I know this may sound harsh, but trust me: you will be much
better off in the long run if you educate yourself somewhat.
To give you an idea on how far off you are, it would be like going into
a car repair shop and asking about a furnace problem. While someone
there may be able to help you, you went to the wrong place.
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin said:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
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--
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[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
--
Walt Reed
[EMAIL PROTECTED]
Office: 207-753-7333
Cell: 207-577-0699
http://www.vinq.com
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RE: [asterisk-users] 1.4.0, IMAP and Dovecot
Mark wrote:
I've been attempting the same with Cyrus and get the same results.
The
interesting thing is if I take the same string (like
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX) and plug it
into the 'mtest' command from the c-client package, it works OK. I
have
not tried this with the production release with Asterisk. Only beta's
1-4.
Thanks. I had not uncovered the mtest tool, and that was a great hint.
It fails with:
(127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX,
Works up until certificate validation with
(127.0.0.1:143/imap/authuser=root/user=dan_austin}INBOX
And succeeds with:
(127.0.0.1:143/imap/authuser=root/novalidate-cert/user=dan_austin}INBOX
I had tried imapflags=novalidate-cert, but I only reloaded the voicemail
Module. Perhaps a full restart is required, which I will try this
afternoon.
Dan
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Re: [asterisk-users] 1.4.0, IMAP and Dovecot
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
Not that I can be of much help, but:
what is your MAILBOX env. set to in dovecot.conf?
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
What exactly passes on the wire? Can you get a dump of the session?
I have a feeling that the imap client is supposed to take this string
and interpert it into some parameters.
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
One useful tool dovecot borrowd from uw-imapd is pre-authentication:
you can use the binaries in /usr/libexec/dovecot to start an imap (or
pop3) session with the permission of the relevant user.
authuser=root ?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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[asterisk-users] Fax on Asterisk 1.4
Did someone has an ATA Grandstream HT496 working on T.38 on Asterisk 1.4? I'm trying different configurations but so far none has worked. Thanks, Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload
Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search on the server and those files do NOT exist. Otherwise can you tell me how to load the MySQL in Asterisk 1.4 to make it work? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Tuesday, December 26, 2006 11:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Savoy, Kevin - Williston, ND wrote: I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk - I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]: loader.c:607 load_resource: Module 'cdr_addon_mysql.so' could not be loaded. The module that is being loaded is not a 1.4 module. It is using the old way of module loading. You should make sure that you are using 1.4 addons and that they are installed. -- Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: php agi trixbox help
Thanks for all the help. I finally got it all working. There was some problems with the swift functions of phpagi as it couldn't find it so that command was failing. I fixed it and all is well now. Thanks! On 12/27/06, blackwater dev [EMAIL PROTECTED] wrote: Ok, I did that and I can now do a locate and do now see: /usr/lib/php4/imap.so /usr/lib/httpd/modules/mod_imap.so So I restarted apache, and tried to run my file from the command line and get the same error. I have another one that doesn't use the phpagi class and it works ok but throws the same warning so I don't think that is why the script doesn't seem to work...not sure. Thanks! On 12/27/06, Time Bandit [EMAIL PROTECTED] wrote: Not sure if this has anything to do with it but running the input.phpscript directly from the command line gives this warning: PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0 yum install php-imap hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: php agi trixbox help
curious about what swift problem you saw and what you did to get the swift problems resolved... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of blackwater dev Sent: Wednesday, December 27, 2006 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: php agi trixbox help Thanks for all the help. I finally got it all working. There was some problems with the swift functions of phpagi as it couldn't find it so that command was failing. I fixed it and all is well now. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.0, IMAP and Dovecot
Dan,
I have IMAP support working now with Courier IMAP. Since Courier (and
probably Dovecot) do not support a single authuser connection that may
access any mailbox, you have to omit the 'authuser' and 'authpassword'
settings in voicemail.conf and then add the username/password login per
extension... e.g.
[general]
...
imapserver=localhost
imapport=143
imapflags=novalidate-cert
expungeonhangup=yes
...
[default]
1234 = 1234,Fred,[EMAIL PROTECTED],,imapuser=fred|imappassword=fredspasswd
4321 = 4321,Bill,[EMAIL PROTECTED],,imapuser=bill|imappassword=billspasswd
I couldn't get Realtime voicemail support working using the imapuser or
imappassword - only via the flat-file, although I have a patch that I am
testing that will allow the imapuser and imappassword settings to be
retrieved from Realtime. The key thing to remember is that you MUST
remove any authuser/authpassword setting in voicemail.conf, otherwise it
doesn't even look for the imapuser/imappassword settings per user.
Cheers,
Ray
Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid remote
specification
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
Thanks,
Dan
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Re: [asterisk-users] Billing solution
Can anyone recommend a call accounting solution with rating for post paid billing that works well with asterisk using the account code or any other info from the CDR? I don't want the billing software to any phone calls for me, therefore any solution that modifies my extensions.conf is out, nor does it have to allow for customers the ability to log in to check their usage/balances. I have looked at astbill but it looks to be way overcomplicated for what I want, as well as it requires realtime. Thank you CDRTool does call rating Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Toll-Free number in India
Can anybody point me to a vendor that can provide a toll free number that can be used in India to reach the united states? Verizon Business is telling me they can't get one. Thanks - Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: php agi trixbox help
I printed out the exec statement in the swift function within the phpagi script and noticed that it didn't have a swift location. For some reason, it wasn't picking it so I just changed that to the location on my machine. I also noticed that it was adding to many slashes to the temp file name. I don't have access to the box here but if needed, can send my modified swift function if you're haing the same problem. On 12/27/06, Mike D'Ambrogia [EMAIL PROTECTED] wrote: curious about what swift problem you saw and what you did to get the swift problems resolved... -Original Message- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *blackwater dev *Sent:* Wednesday, December 27, 2006 1:14 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Re: php agi trixbox help Thanks for all the help. I finally got it all working. There was some problems with the swift functions of phpagi as it couldn't find it so that command was failing. I fixed it and all is well now. Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Toll-Free number in India
Am Mittwoch, den 27.12.2006, 17:22 -0800 schrieb Tom Lynn: Can anybody point me to a vendor that can provide a toll free number that can be used in India to reach the united states? Verizon Business is telling me they can't get one. Looks like a -biz related question... A quick google search turned out www.tollfreeforwarding.com which I do not have experience with. They seem to have free test setups, so serve yourself. A minimum 99$/month call volume and 50c/min for forwarding to US landline is not what I call cheap, but that is rather not my problem ;) Perhaps a local dealer could offer better deals - if you need a 800 access number there, you probably have some business connections in that country, so ask around there to see wether better deals exist. The rates they offer for Germany are way above cheap (2-3 times of what local companies list). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me challenge
This is not true of PRI (ZAP channels). Asterisk gets status such as
ringing, answered, busy...
Thanks,
Steve
- Original Message -
*From:* Eric Jacksch mailto:[EMAIL PROTECTED]
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Wednesday, December 20, 2006 11:47 PM
*Subject:* Re: [asterisk-users] Follow-me challenge
Sorry, didn’t realize you were sending the call out on a Zap channel.
Yes, as soon as the call goes out a Zap channel it is “answered”
as far as Asterisk is concerned. I send out all my findme traffic
via SIP.
On 2006-12-19 21:19, Chris Johnson [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On 12/18/06, *Eric Jacksch* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Is the problem just when you don't answer the cell phone?
Many cell phones go to a voice announcement when they're
turned off or not answered, and Asterisk thinks the call
has been answered. The other issue could be that your
gateway (asterisk1) is answering the call before the
outbound leg is answered. One workaround would be to use a
macro that requires you to press a key to accept the call
on your cell. (See the M option to the dial command and
http://www.voip-info.org/wiki/view/Asterisk+tips+findme)
http://www.voip-info.org/wiki/view/Asterisk+tips+findme%29
http://www.voip-info.org/wiki/view/Asterisk+tips+findme%29
Also, I see that you're using the r option — you might
want to drop that.
I'm also not convinced that it will ever find 102,107 in
your dialplan. You might want to look at using
${DIALSTATUS} and making it a bit more explicit.
Cheers,
Eric
Dropped the r option and line 107.
The M option had the same result.
Based on another comment :
Is your other server patching through to a Zap channel (analog)?
If so, as soon as the dial goes out, an analog Zap channel is
considered
answered, which could be your issue.
Doesn't sound like follow-me will work properly with an analog
trunk
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RE: [asterisk-users] 1.4.0, IMAP and Dovecot
Walt wrote: Dan, Please accept my sincerest appology. I had my head thoroughly up my back orifice. I haven't kept up with the new IMAP feature in 1.4 Don't sweat it... I saw your first post before going out for the day with the family, and couldn't figure out how it wasn't related to Asterisk. Glad I waited to respond until after I got back... I'll go back in my corner now :-) Thanks, Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 1.4.0, IMAP and Dovecot
Tzafrir wrote:
On Wed, Dec 27, 2006 at 10:44:10AM -0800, Dan Austin wrote:
I thought I would give the new IMAP support a spin on my home
server, but without much luck so far.
Asterisk 1.4.0
Dovecot 0.99.14
Maildir format
C-client 2006d
Not that I can be of much help, but:
what is your MAILBOX env. set to in dovecot.conf?
No customization at all. The MailDir folders are in
my home directory (more below)
The imap server is also the Asterisk server, so connections are
on the localhost.
The error posted to the logs is:
IMAP Error: Can't open mailbox
{127.0.0.1:143/imap/authuser=root//user=dan_austin}INBOX: invalid
remote
specification
What exactly passes on the wire? Can you get a dump of the session?
I am connecting over the localhost loopback, I might be able to get
a dump, I've honestly never tried a capture against 127.0.0.1
I have a feeling that the imap client is supposed to take this string
and interpert it into some parameters.
Yes, and I think I found the culprit.
Digging in the code and the c-client documentation the '//' is where
additional flags would go. I've tried a number of the flags
supported
by the c-client library, but the results are the same.
Has anyone managed to get IMAP working in Asterisk with Dovecotas the
backend?
With the mtest tool in the c-client package, I found that dovecot
does not like the '//' in the connection string. So imapflags is not
really optional against Dovecot, which has an easy work-around. I'd
Guess that the code in app_voicemail should either use a sane default
for the imapflags or conditionally not include the second '/' if the
option is not set.
One useful tool dovecot borrowd from uw-imapd is pre-authentication:
you can use the binaries in /usr/libexec/dovecot to start an imap (or
Pop3) session with the permission of the relevant user.
authuser=root ?
Yeah, in the real world that would be exceedingly stupid. On the other
hand this is basically a single user system with not much interesting
on it. Once it is known to work with the all-powerful root, I can then
focus on a dedicated, less privileged account...
Thanks,
Dan
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[asterisk-users] How to connect two asterisk server
Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients.. plz anyone suggest me Regards, Thiru ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The Good, Bad and Scam VoIP Providers
Dovid B [EMAIL PROTECTED] wrote: A PI that does asterisk on the side ?? WTF ?? Do you have a list of business types that are not allowed to use VoIP? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Say who is using the PSTN?
I use an SPA3000 to connect to the PSTN (SIP/pstn). Since I only have one line,
if it is in use, and someone else tries to dial out, they get the
all-outgoing-lines-unavailable message played.
I'd like to find a way to instead tell them which extension is using the PSTN
line. I know that info is available in the manager API, but I have no idea how
to get access to it from either the dialplan or an AGI script.
The relevant portion of my dialplan looks like this:
[macro-dialtopstn]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],120,WT)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,Hangup
exten = s-BUSY,1,Playtones(busy)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Play(number-not-answering)
exten = s-NOANSWER,2,Hangup
exten = s-CONGESTION,1,Playback(all-outgoing-lines-unavailable)
exten = s-CONGESTION,2,Hangup
exten = s-CHANUNAVAIL,1,Goto(s-CONGESTION,1)
Has anyone written anything like this, or have any suggestions on the easiest
way to do it? I've searched and haven't found anything like this so far.
Thanks,
Grant Emsley
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