Re: [asterisk-users] Asterisk 1.8.9.3 Problem With Logger

Is this a general issue or just affecting specific versions? Jon Farmer Tel 07795 118140 On 7 March 2012 16:33, Paul Belanger pabelan...@digium.com wrote: On 12-03-07 04:29 AM, Jon Farmer wrote: Hi I have recently upgraded a box to 1.8.9.3 and have noticed that randomly the logger

Re: [asterisk-users] Asterisk 1.8.9.3 Problem With Logger

Hi Just realised this is due to a FIFO blocking. Fixed that and all back to normal. Regards Jon Jon Farmer Tel 07795 118140 On 7 March 2012 16:33, Paul Belanger pabelan...@digium.com wrote: On 12-03-07 04:29 AM, Jon Farmer wrote: Hi I have recently upgraded a box to 1.8.9.3 and have

[asterisk-users] Asterisk 1.8.9.3 Problem With Logger

at other times too, The only way I have managed to get it back is to kill and restart asterisk. Any ideas what is going on. Regards Jon Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Problem With Playing Busy Tone

Hi Since upgrading to 1.8.4.3 my callers no longer hear busy tone when I use playtones(). Here is the CLI output on such a case http://pastebin.com/TMBFhngh Any ideas anyone? Regards Jon -- _ -- Bandwidth and Colocation

Re: [asterisk-users] SIP Trace to troubleshoot one way of communications

On Aug 26, 2011 4:54 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How can I get a SIP trace to troubleshoot a one way of communications? I need to see what is happenning in the packets to know the reason of the problem. Install ngrep on the box. Then type something like. ngrep -tq

[asterisk-users] rtptimeout on 1.8.4

seconds. Anybody got any ideas how to get 1.8.4 rtptimeout correctly? Regards Jon Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

[asterisk-users] RTP and Signalling Dropping

to look next? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] Multiple Registrations

Hi I am researching if there is a practical number of SIP accounts that Asterisk can register against as a UA. I have an idea for a project but it would need to register multiple accounts from multiple providers to work. Regards Jon -- Jon Farmer Tel 07795 118140

Re: [asterisk-users] AGI Delimiter in 1.6

On 16 September 2010 22:23, Barry Miller asterisk-us...@notanet.net wrote: For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work.  See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards Jon -- Jon

Re: [asterisk-users] AGI Delimiter in 1.6

, Jon Farmer wrote: On 16 September 2010 22:23, Barry Millerasterisk-us...@notanet.net wrote: For an interim fix, setting res_agi=1.4 in the [compat] section of asterisk.conf should work. See UPGRADE-1.6.txt . I have tried this but it still complains about the pipe not being a comma. Regards

[asterisk-users] Playing Audio To One Channel

Hi I have a call established and I want to play audio to just one channel on that call. Is this possible? If so, how? My google-fu has failed on this one. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth

Re: [asterisk-users] Playing Audio To One Channel

to the subscriber when call credit is low. However I don't want the other party to hear the message. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Playing Audio To One Channel

the manager interface the extensions get fired and doing a show channels shows the chanspy and playbacks working but I hear nothing. Any ideas? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation

[asterisk-users] AGI Delimiter in 1.6

to access this data. Does anyone have any ideas how I might go about this? Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] AGI Delimiter in 1.6

was pretty tired when I tried so it might just need a fresh set of eyes. Regards Jon -- Jon Farmer Tel 07795 118140 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us

Re: [asterisk-users] AGI Delimiter in 1.6

,AGI(myagi.xx,${VARNAME}) Then you just do your magic on ${VARNAME} Yes, but the problem is I am trying to pass the whole AUTH line which is key=value pairs seperated by commas. e.g. username=myusername, domain=mydomain This breaks when passing to an AGI in 1.6. -- Jon Farmer Tel 07795

[asterisk-users] Remote-Party-ID and selective CLI withold

Hi One of my SIP providers need me to send the Remote-Party-ID with privacy=on to withhold CLI and privacy=off to show CLI. I want the option to withhold CLI selectable by my users. I have set sendprid=yes in the sip.conf but I cant find a way to toggle the privacy between on and off on a per

Re: [asterisk-users] Remote-Party-ID and selective CLI withold

Subject: Re: [asterisk-users] Remote-Party-ID and selective CLI withold Quoting Jon Farmer [EMAIL PROTECTED]: Hi One of my SIP providers need me to send the Remote-Party-ID with privacy=on to withhold CLI and privacy=off to show CLI. I want the option to withhold CLI selectable by my

[asterisk-users] Linksys 942 Call Transfer

Hi I have a customer who is using Linksys 942 phones. When they try to transfer a call the Asterisk CLI reports that both legs of the call must exist on the server. The call they are trying to transfer then drops. Does anyone know why this is and how to fix it? TIA Regards Jon

Re: [asterisk-users] IVR and MySQL

I write such apps all the time, i have written IVR apps that talk to SQL Server (Sage500), MySQL, our credit/debit card processing system. Regards Jon --- Thiago Maluf [EMAIL PROTECTED] wrote: Hi Fabio, of course that you can. One way to do it is working with app MYSQL(), where you

[asterisk-users] Avaya IP Office DTMF Issue

? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] injecting audio announcements into sip channel

--- Dinesh Nair [EMAIL PROTECTED] wrote: take a look at the L() option to Dial(). The original poster said he need to play different messages at different call durations. In order to do that you would need to dynamically alter LIMIT_WARNING_FILE as the call progressed. Regards Jon Jon

[asterisk-users] AGI DTMF Problem

? Jon Farmer Telford, Shropshire, UK ___ All New Yahoo! Mail – Tired of unwanted email come-ons? Let our SpamGuard protect you. http://uk.docs.yahoo.com/nowyoucan.html

Re: [asterisk-users] problem with asterisk AGI

Set a variable that you can then use GotoIf in the dialplan to branch to the required exten Jon Farmer Telford, Shropshire, UK - Original Message From: prasanth [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, 8 February, 2007 10:06:07 AM Subject: [asterisk-users

Re: [asterisk-users] Dialplan programming vs. AGI vs. ???

controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that hook into our CRM and Accounting systems for fault reporting and credit card payments etc. What you need is the tool for the job like everything in life :-) Jon Farmer Telford, Shropshire, UK

Re: [asterisk-users] PHP AGI script callerid question

Have you tried phpagi http://phpagi.sourceforge.net/ Jon Farmer Telford, Shropshire, UK - Original Message From: Michelle Dupuis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 26 January, 2007 5:52:27

Re: [asterisk-users] Cisco 7970 Unprovisioned

Are you setting the TFTP server address in the DHCP? Are you checking the TFTP log to see what files the phone is requesting and not finding? Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Token PBX [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent

Re: [asterisk-users] How to connect two asterisk server

Hi I would suggest a IAX2 trunk between the two servers. You will need to modify the dialplan to recognise which extensions are on each box and route accordingly. The fact your clients are SIP does not preclude you from using IAX2 to connect the servers. Regards Jon Jon Farmer Telford

Re: [asterisk-users] MWI across multiple servers

. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across

Re: [asterisk-users] MySQL cmd % pattern matching

Try enclosing in single quotes. ie. SELECT name from contacts where tel like '%${number}' Jon Farmer Telford, Shropshire, UK - Original Message From: Garth van Sittert [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent

Re: [asterisk-users] Registration problem

-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=372b2479/ Asterisk is asking the phone to resend the registration with WWW-Authenticate using MD5 hash. Make sure the phone supports this and retry. Or you could turn this option off in the sip.conf. Regards Jon -- Jon Farmer Telford

Re: [asterisk-users] SIP v IAX2

Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK

Re: [asterisk-users] Reception Console

bus: 03 8320 8106 mob: 0434 673 529 ___ --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 Jon Farmer Telford

Re: [asterisk-users] Asterisk and Siemens Legacy PBX

is connected to a ‘real’ ISDN PRI. This is a everyday use for Asterisk :-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] load average with MOH

is the incentive to use native MOH? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [asterisk-users] Check call duration on active call in CLI?

Hermann Wecke wrote: voiplist wrote: Is there a command to check the call duration of an active call in the CLI? show channels verbose show channel channel_id_originating_channel shows among other things Elapsed Time: 0h2m47s Regards Jon -- Jon Farmer Telford, Shropshire, UK

Re: [asterisk-users] Asterisk and Siemens Legacy PBX

SIP phone 101 or if the numbers from the siemens follow a pattern ie they all start with 12 then you could use exten = _12,1,Dial(SIP/101) If you check the extensions.conf page at www.voip-info.org/wiki you will see loads of examples on how to construct a dialplan HTH Jon -- Jon

[asterisk-users] Sony Ericsson F250m, Sipura 3000 and Asterisk

use the F250M. If anyone has any pointers on this I would be grateful. Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Sony Ericsson F250m, Sipura 3000 and Asterisk

-- Jon Farmer Telford, Shropshire, UK ___ --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] Dial plan question

and fairly straightforward way of doing what you require. Regards Jon Jon Farmer Telford, Shropshire, UK ___ All new Yahoo! Mail The new Interface is stunning in its simplicity and ease of use. - PC Magazine

Re: [asterisk-users] Call queue drops call after 1 min

the CLI output when this happens and the relevant portions of your extension.conf and queues.conf Regards Jon Jon Farmer Telford, Shropshire, UK ___ Copy addresses and emails from any email account to Yahoo! Mail - quick

Re: [Asterisk-Users] How to retrieve voicemail

}) HTH Jon -- Jon Farmer Telford, Shropshire, UK ___ --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] PHP-AGI help

Yes you have a parse error in your PHP when I saved it locally and run it from the command line I got syntax error, unexpected '[', expecting ']' in test.php on line 33 Jon FarmerTelford, Shropshire, UK - Original Message From: Matthew Warren [EMAIL PROTECTED] To:

Re: [Asterisk-Users] Dumping queue_log to MySQL

We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.RegardsJonJon FarmerTelford, Shropshire, UK- Original Message From: Kevin Savoy [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion

[Asterisk-Users] voicemail dialout

Hi How do i disable dialling out from voicemail? -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

Re: [Asterisk-Users] voicemail dialout

Doug Lytle wrote: It's enabled/disabled via the voicemail.conf I have commented out dialout=from-vm but the option is still given even though any number dialled results in unobtainable. So I dont want the option given. -- Jon Farmer Telford, Shropshire, UK

Re: [Asterisk-Users] voicemail dialout

Doug Lytle wrote: You'll also need to do a stop/start of Asterisk. Done that also, no difference -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] voicemail dialout

; greetings. The default is no. ; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory ; The default is no. Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] voicemail dialout

') -- Playing 'vm-opts' (language 'en') -- Playing 'vm-tomakecall' (language 'en') -- Playing 'vm-starmain' (language 'en') -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

Re: [Asterisk-Users] voicemail dialout

was not offered to you or it did not work when offered. I am also on 1.2.7.1 -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] billing realtime

which I don't want. Anyone know of a system that allows concurrent calls? -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] billing realtime

Nick Hoffman wrote: Hi Jon. If a customer has 10 minutes of call credit left and he makes 2 concurrent calls, how do you know to cut off the 2 calls at the 5 minute mark rather than cut off both calls after 10 minutes? That is the problem I am asking about :-) -- Jon Farmer Telford

Re: [Asterisk-Users] Re: billing realtime

at the current time per account. If the credit expires for that account it hangs up all channels for that account. The only problem at the moment is I can't figure away to dynamically play a warning to the callers. -- Jon Farmer Telford, Shropshire, UK

Re: [Asterisk-Users] Re: billing realtime

Won't the called party hear the warning as well if you do that? Jon FarmerTelford, Shropshire, UK - Original Message From: Tony Mountifield [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 26 April, 2006 3:08:18 PM Subject: [Asterisk-Users] Re: billing realtime

Re: [Asterisk-Users] Camp on?

I believe what you refer to is called Ring Back When Free at least thats how I know it in the UK. Regards Jon Jon FarmerTelford, Shropshire, UK - Original Message From: Patrick [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 26 April, 2006 3:25:12 PM Subject:

[Asterisk-Users] Asterisk and SER hangup issue

. Anybody got any ideas? Regards Jon -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Variables

running... If each call had a uniq session id i could easily just use that http://www.voip-info.org/wiki/view/Asterisk+variables See section about variable inheritance. Regards Jon -- Jon Farmer Telford, Shropshire ___ --Bandwidth

[Asterisk-Users] Call Forward and AGI

is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a way around this? -- Jon Farmer Telford, Shropshire ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Manager API Help

, it is a class that implements the manager API. -- Jon Farmer Telford, Shropshire ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [Asterisk-Users] Cisco 7960 problems

Paul A Brown wrote: Hi All, Not sure if this is a phone problem or an Asterisk problem. Basically after a period of time (around 30 minutes but not too sure of the time) the phone no longer delivers any sounds. What I mean by that is. if I pick up the phone after a reset I get a

Re: [Asterisk-Users] Cisco 7960 problems

Paul A Brown wrote: Do you have a sccp config example I could look at http://www.voip-info.org/wiki/view/SCCP-HOWTO2 Regards Jon ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] Dial Plan Logic Problem

/asterisk-users I sorted it, I needed to include the campon context before the mainmenu context in the default context. Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Photos – NEW, now offering a quality print

[Asterisk-Users] Dial Plan Logic Problem

,GotoIf($[${LOOPER} 4]?mainmenu,s,2) exten = s,7,Goto(huntgroups,0,1) exten = t,1,GotoIf($[${LOOPER} 4]?mainmenu,s,2) exten = t,2,Hangup exten = i,1,Goto(mainmenu,s,1) exten = 1,1,Goto(sales,s,1) exten = 2,1,Goto(finance,s,1) exten = 0,1,Goto(huntgroups,0,1) exten = #,1,Goto(mainmenu,s,1) Jon

[Asterisk-Users] Problems with wcte11xp module

is lost. If I rmmod the driver then audio returns. What is going on? Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide

Re: [Asterisk-Users] time update (7905)

in the SIPDefault.cnf. Then if you have told the phones to use NTP they update automatically. Regards Jon Jon Farmer Telford, Shropshire, UK ___ To help you stay safe and secure online, we've developed the all new Yahoo

Re: [Asterisk-Users] Problems with wcte11xp module

--- [EMAIL PROTECTED] wrote: It means that you are loading the digium card up with incorrect values. I had it happen to me recently. Aha I wonder that, are you referring to the span definition in the /etc/zaptel.conf? Regards Jon Jon Farmer Telford, Shropshire, UK

Re: [Asterisk-Users] Problems with wcte11xp module

--- [EMAIL PROTECTED] wrote: It means that you are loading the digium card up with incorrect values. Ok when I modprobe wcte11xp I get the following message ZT_CHANCONFIG failed on channel 26: No such device or address Any ideas? Regards Jon Jon Farmer Telford, Shropshire, UK

Re: [Asterisk-Users] PHP/AGI Problem

Michael Stearne wrote: Jon, What version of PHPAGI are you using? I am starting a PHPAGI app and want to know whether to use 1.12 or 2.0CVS. I am using 1.12 Regards Jon ___ How much free photo storage do you

Re: [Asterisk-Users] PHP/AGI Problem

Benjamin West wrote: Michael, The version, in the context of Jon's problem, was irrelevant. Jon's problem was due to a small bug in his code, and not related to PHPAGI. Hi Benjamin, Actually I would say it was more to do with my lack of understanding with how Asterisk AGI worked and my

[Asterisk-Users] Problem with SIP peer registration

Hi I am trying to get 2 incoming SIP accounts working from 2 different providers. One is sipgate.co.uk and the other is voipuser.org. If I load the Register command seperate they will both register phone and incoming works. If I try to load them both only sipgate registers. Anybody got any

Re: [Asterisk-Users] PHP/AGI Problem

Moises Silva wrote: what version of asterisk you are using? i had some problems with agi until i upgrade to asterisk 1.0.7 I am also using version 1.0.7. I installed it from the Xorcom CD ISO if you run simple agi scripts works? try using this 2, one is php, the other one is C, that

Re: [Asterisk-Users] PHP/AGI Problem

Matthew Boehm wrote: You using phpagi v2? Some of your functions are built in I believe. No, I am not using phpagi v2, I will take a look, None of my AGI's loop but if someone hangs up in the middle of script execution the script dies. I'm almost 99% sure of that. All of my

Re: [Asterisk-Users] PHP/AGI Problem

Benjamin West wrote: So if the user stays on the line, the php script never blocks or hangs, and the phone call terminates correctly, including the php script. However, if the user hangs up the phone, your php script never times out because it gets stuck in a state that doesn't count towards

Re: [Asterisk-Users] Little Php question

Ronald wrote: When starting the script I get a parse error (unexpected t_string) in line 15 which is the Exten line Can anybody help me out. (I have minimal php knowledge, so Im turning to you all) Change fputs($socket, Exten: 12345678\r\n\); to fputs($socket, Exten: 12345678\r\n);

Re: [Asterisk-Users] PHP/AGI Problem

Alex Barnes wrote: Currently after: $res = $agi-agi_getdtmf(1,1,$term,$prompt=FALSE); You test for no DTMF and then simply return null. Instead you could call the other piece of code you have: $status = $agi-agi_channel_status($agi-request[agi_channel]); $agi-conlog(Status code: .

[Asterisk-Users] PHP/AGI Problem

/usr/share/asterisk/agi-bin/test.php They only way to get rid of it is to killall -9 it. Any ideas how I can get asterisk to kill the script if the caller hangs up? Regards Jon Farmer ___ How much free photo storage do

RE: [Asterisk-Users] PHP/AGI Problem

if I very time I poll for DTMF I check the channel status and exit() the script if the line is anything but UP. Regards Jon Jon Farmer Skype: viperdude_uk ___ How much free photo storage do you get? Store your holiday

RE: [Asterisk-Users] PHP/AGI Problem

--- Alex Barnes [EMAIL PROTECTED] wrote: Although this isnt a substitute for a correctly terminating script, I would have thought that the PHP 'maximum script execution time' variable would kick-in and kill the script eventually. Well I have already tried that I have the first line of

Re: [Asterisk-Users] MoH: mpg123 problems

--- yusuf [EMAIL PROTECTED] wrote: Hey all, I have read on voip-info.org that to configure MoH asterisk requires the use of mpg123. I have installed mpg123 and restarted asterisk. But, when i put a call on hold i get this error: May 25 14:13:03 WARNING[1872]: res_musiconhold.c:865

Re: [Asterisk-Users] PHP/AGI Problem

Well, that is your problem. Don't use deadagi. DeadAGI is for use if you want to continue processing after the call hangs up. That is why your scripts are continuing to run. Use regular AGI. I get the same behaviour if I use deadagi or just agi Regards Jon

RE: [Asterisk-Users] PHP/AGI Problem

Jon Jon Farmer Skype: viperdude_uk ___ Yahoo! Messenger - NEW crystal clear PC to PC calling worldwide with voicemail http://uk.messenger.yahoo.com ___ Asterisk

Re: [Asterisk-Users] PHP/AGI Problem

Matthew Boehm wrote: Well then you got something screwed up somewhere. I've got many PHP AGI's and as soon as the caller hangs up the script terminates. This is what I use: exten = _80059974XX,1,AGI(line_counter.php) Works like a champ. And yes, I'm using phpagi.php as well. Matthew,

Re: [Asterisk-Users] PHP/AGI Problem

Moises Silva wrote: could you post the script, the output of the script in the asterisk console and which asterisk version are you working with? See below This is just a proof of concept script so its a bit basic... #!/usr/bin/php -q ?php set_time_limit(30); require phpagi.php;