Re: [asterisk-users] Asterisk 11 and stdexten written in AEL invoked by pbx_config
I wonder if anyone in the list find useful to write stdexten in AEL instead of in extensions.conf syntax. I do, but probably I'm just alone. On Wed, Oct 31, 2012 at 11:44 AM, Octavio Ruiz wrote: > Almost two years ago, a change between how AEL code is built into > Asterisk dialplan between minor versions made clear the need to > provide a sane entry point into AEL subroutines and that's how > AELSub() born. > > With Asterisk 11 release, they way [stdexten] at extensions.conf is > invoked changed from Macro to Gosub using the 'missing context > feature' and this caused that any stdexten written in anything else > but extensions.conf (AEL, LUA, etc, being these not able to define an > arbitrary priority) will not work. > > The only way to workaround this is to fallback to Macro() and write > macro-contexts in AEL with the stack limit implications of them so I'm > proposing to add to asterisk.conf configuration the ability to invoke > stdexten using AELSub() so stdexten can be again be written in AEL > mantaining real backward compatiblity as it did the fact that you are > able to fallback to Macro. > > ;stdexten = gosub ; How to invoke the extensions.conf stdexten. > ; macro - Invoke the stdexten using a macro as > ; done by legacy Asterisk versions. > ; aelsub - Invoke the stdexten sutbroutine using AELSub > ; when stdexten is defined in AEL. > ; gosub - Invoke the stdexten using a gosub as > ; documented in extensions.conf.sample. > > I've already started this conversation on the development lists, you > can follow up it at: > > http://lists.digium.com/pipermail/asterisk-dev/2012-August/05.html > > and there is a working patch submited to JIRA here: > > https://issues.asterisk.org/jira/browse/ASTERISK-20355 > > I would like to read your comments. > > Best, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11 and stdexten written in AEL invoked by pbx_config
Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten written in anything else but extensions.conf (AEL, LUA, etc, being these not able to define an arbitrary priority) will not work. The only way to workaround this is to fallback to Macro() and write macro-contexts in AEL with the stack limit implications of them so I'm proposing to add to asterisk.conf configuration the ability to invoke stdexten using AELSub() so stdexten can be again be written in AEL mantaining real backward compatiblity as it did the fact that you are able to fallback to Macro. ;stdexten = gosub ; How to invoke the extensions.conf stdexten. ; macro - Invoke the stdexten using a macro as ; done by legacy Asterisk versions. ; aelsub - Invoke the stdexten sutbroutine using AELSub ; when stdexten is defined in AEL. ; gosub - Invoke the stdexten using a gosub as ; documented in extensions.conf.sample. I've already started this conversation on the development lists, you can follow up it at: http://lists.digium.com/pipermail/asterisk-dev/2012-August/05.html and there is a working patch submited to JIRA here: https://issues.asterisk.org/jira/browse/ASTERISK-20355 I would like to read your comments. Best, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Congestion in Outgoing call through PRI
On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman <[EMAIL PROTECTED]> wrote: > Octavio Ruiz wrote: >> On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan <[EMAIL PROTECTED]> wrote: >> The output of a >> CLI> pri intese debug >> at Asterisk CLI before make a test call would be very useful, libPRI >> 1.4.7 is just fine. > I am amazed no one else have suggested trying a different phone type > like an IAX2 softphone. (if i am right, this will work) For me is complete clear that -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) the Zap channel is the one which returns the congestion status, not the other leg (whatever the technology is). Anyway, if he try both options nobody is going to be hurt. I forgot completely mention (and carefully read their zaptel.conf configuration and see dchan=16 declared rather than hardhdlc=16 ) that probably their issue is already solved and documented just right here: http://wiki.sangoma.com/Asterisk-FAQ#hardhdlc Shariq, can you tell us your wanrouter + zaptel version? -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Congestion in Outgoing call through PRI
On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan <[EMAIL PROTECTED]> wrote: > When i dial out any number through PRI it gives the following error every > time, while incoming calls works fine > I have sangoma E1 PRI card. The output of a CLI> pri intese debug at Asterisk CLI before make a test call would be very useful, libPRI 1.4.7 is just fine. -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selectively disable echo cancellation?
On Tue, Sep 2, 2008 at 6:16 PM, Ken D'Ambrosio <[EMAIL PROTECTED]> wrote:
> Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
> currently passing through some of my in-bound calls to a legacy PBX (which
> I hope to eventually replace). That being said, until I do, I'd like to
> kill echo cancellation for the passed-through calls -- I don't want to
> mess with their fax reception.
>
> Any idea how to do this?
Is
echocancelwhenbridged=no
inside zapata.conf what are you looking for?
If not, what I figured out is if you run
System(wan_ec_client wanpipe1 disable ${VALUE}) ;
in your dialplan logic [perhaps inside a macro called with the M()
option for Dial()] would do the trick.
Don't forget that you obtain "Zap/${VALUE}-1" from ${CHANNEL} (using
some variable stripping) and to run
System(wan_ec_client wanpipe1 enable ${VALUE}) ;
at Hangup.
Regards,
--
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 14-087790
Mobile: (+52 1 55) 41-351242
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Re: [asterisk-users] How to get a clean, basic configuration?
On Fri, Feb 22, 2008 at 10:15 AM, Tilghman Lesher <[EMAIL PROTECTED]> wrote: > On Friday 22 February 2008 04:55:13 Vincent wrote: >> On Thu, 21 Feb 2008 22:04:41 +0200, Tzafrir Cohen wrote: >> >For the brave: use modules.conf without 'autoload = yes'. This promises >> >you many hours of interesting dialplan debugging. Enjoy. >> >> Yup, that's what I anticipated, which is why I was asking which >> modules I can _safely_ remove without breaking things :-) > Generally, the rule is that you can't remove any of the res_* Based on experience you almost always need res_features.so, otherwise you will experience crashes. app_dial and many chan_* depends on it. -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any idea how making Asterisk "transparent"?
On Fri, Dec 7, 2007 at 6:43 AM, dave cantera <[EMAIL PROTECTED]> wrote: > artifex, > if you want call recording transparently, check out orecX.com they > have a commercial and an open source SIP call recording package... no > zap recording If you are using sangoma hardwarde it's possible to do a voice RTP tap for OrecX http://wiki.sangoma.com/wanpipe-voice-rtp-tap -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- 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] PRI Setup problem
On Mon, Oct 1, 2007 at 7:23 PM, Alvin Austin <[EMAIL PROTECTED]> wrote: > Hi everyone, Hi Alvin, > I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI > (Megalink) circuit and having some trouble getting it to handshake. > Thanks for any help or suggestions to diagnose this problem. I was trying to figure out on that issue and find a something reasonable and I found it. As of branch 1.4 of zaptel, the option for hardware HDLC framing has been integrated and IIRC since 3.2.2 version, wanpipe uses it rather than their own patch, so it's needed to change all those ... dchan=16 ... from zaptel.conf to ... hardhdlc=16 ... in order to pass the HDLC framing from the Sangoma's HDLC engine to the D channel. That explains why when you changed version prior the mentioned above, everything worked (Considering that this thread is from Oct 1, 2007) > The problem is that Asterisk has trouble bringing up the link. I see > the following lines every couple of minutes: > > == Primary D-Channel on span 1 up > == Primary D-Channel on span 1 up > == Primary D-Channel on span 1 up Since 2008-03-08, (chan_zap revision #106945) you will see that just once because the channel state itself is not changing. > q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED > q921.c:664 q921_dchannel_up: q921_state now is > Q921_LINK_CONNECTION_ESTABLISHED Those messages indicate that layer 2 is trying to reinitialize because there are no response from the other side due the missing working D-Channel. Regards, -- Octavio H. Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 155) 5514-087790 Mobile: (+52 155) 5541-351242 ___ -- 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] PRI cards, Digium vs. Sangoma
On Mon, Mar 31, 2008 at 12:58 PM, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > There are a number of > ways to order that. In most distributions it is done by explicit > ordering. In some it is done by dependencies. Gentoo is one of them, where if you run directly from CLI /etc/init.d/zaptel stop and wanrouter script have the zaptel dependency first wanrouter is going to be stopped as a direct dependency of zaptel. http://www.gentoo.org/doc/en/handbook/handbook-x86.xml?part=2&chap=4 Have you any other in mind? -- Octavio H. Ruiz Cervera ___ -- 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] PRI cards, Digium vs. Sangoma
On Sun, Aug 26, 2007 at 9:49 AM, Doug Lytle <[EMAIL PROTECTED]> wrote: > I have, on many occasions, had kernel > panics when trying to shut down wanrouter. I don't have this 'fear' > with Digium cards. I never have had those issues if you don't execute zaptel init.d script, because it tries to unmod all zaptel dependant modules including wanrouter which need to be unmoded with wanrouter script. (A matter of order in the unload process). Perhaps this tip helps you avoiding that fear. This makes an auto-reboot after a kernel panic occurs. /etc/sysctl.conf: kernel.panic = 1 OR echo 1 > /proc/sys/kernel/panic OR pass panic=1 as a kernel parameter in your grub.conf/lilo.conf -- Octavio H. Ruiz Cervera ___ -- 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] Best "Console" phone?
On 1/27/08, Michelle Dupuis <[EMAIL PROTECTED]> wrote: > The Aastra's also have a range of interested firmware bugs that > support/development just can't seem to fix. Do a search for aastra > hang/lockup and you will find what I mean. Have you ever achieved those hangup cases? Which firmware/model are you using? -- Octavio H. Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Mobile: (+55 155) 5514-087790 Mobile: (+55 155) 5541-351242 ___ -- 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] POE draw on Aastra 480i
> > Allen Casteran wrote: > > Anyone know what the POE draw is for the Aastra 480i phones? > > We have switches that will do 15 watts on 12 ports but only do 7.7 watts on > > all 24 ports. > > A Cisco 3560 switch will do 15.6 watts on all 24 ports. > > Just trying to find out if we need that much power. > Drew wrote: > According to Aastra tech support, 5 watts (peak) per 480i. > We are testing five phones running on a Linksys SRW208P that will only > support full 15W on up to > 4 of 8 ports. I can power up the switch while all phones are connected > without any issues. > I would expect your lower power switch will provide ample power. But, PoE class does not matter? Did you plug five Aastra phones? I'm suspicious about how that scenario worked, I mean, as far as i know Aastra phones should register as a zero PoE class, that means it would reserve up to 12.94 watts no matter how many watts uses. So, my guess here is even if the phone use only 5 watts, the switch already reserved 12.94 watts for it. I would love to see what happens if you plug a sixth phone or figure out if you used an Aastra phone. Can you tell us what model/brand you used? Dimensioning PoE devices over capable switches has been a new issue which involves many factors like those described before. Regards, PD. Sorry about the original thread break off, I've been unable to find the original one. -- Octavio H. Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Mobile: (+55 155) 5514-087790 Mobile: (+55 155) 5541-351242 ___ -- 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] queues
>Is there anyway to setup a queue with only one agent (device) which is >always logged in. So when a call hits that queue the device will ring (if >not already on a call) or will be put in the queue if the call is already >in place? Sure, in queues.conf you can add many type of members (not just agents) like SIP or Local channels. So you don't need to use AgentLogin/CallBackLogin ej. [recepcion] musicclass = default monitor-format = wav49 strategy = ringall timeout = 15 retry = 2 autopause = no maxlen = 3 context = voicemail setinterfacevar = yes announce-frequency = 15 periodic-announce-frequency = 0 announce-holdtime = yes announce-round-seconds = 10 joinempty = strict leavewhenempty = strict eventwhencalled = yes eventmemberstatus = yes ringinuse = yes timeoutrestart = no member => SIP/9001,1 member => SIP/9005,2 -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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: [asterisk-dev] Find the name of queue
> i´m trying find in the codes of asterisk as change the name of file created
> after that a extension dial for a queue.
> Has someone some sugestion for obtain this name of queue(150)?
Use Queue monitor rather than Agent monitor.
agents.conf
recordagentcalls=no
queues.conf
monitor-type = MixMonitor
monitor-format = gsm
And define MONITOR_FILENAME in your dialplan
(just an example)
[macro-queues]
exten s,1,MONITOR_FILENAME=${ARG1}
exten s,n,Queue(${ARG1})
--
I like your SNOOPY POSTER!!
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[asterisk-users] deprecated ALERT_INFO var andAMI's Originate command
Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I don't have that option, rather than calling a Local/ channel to SetSIPheaders() and Dial(). I don't want to do it in that fashion 'cos I like (and have) to have completely separated dial plan logic (extensions.conf) and external applications via AMI. Regards, -- I'm having a MID-WEEK CRISIS! ___ --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] Call load balancing
> I've got a system I'm putting together to handle IVR calls with *
> I have one head system that terminates two PRIs. It routes the calls from
> the PRIs to * boxes using IAX I'm planning on having four or five * boxes.
> The * boxes run AGI scripts to process the IVR calls. Can I load balance the
> routing if I have five calls each of the IVR * boxes gets two call and the
> next call would go to the system that currently has the lowest number of
> calls?
Another approach: what about load-balance (in terms of redundancy and
scalability) the AGI app's and just the AGIs with FastAGI? So your
IVR application can be separated from your * boxes and they (the * boxes)
dont have to ve overloaded with your AGI apps.
Your head system receive the two PRIs and in dial-plan logic you can (maybe
using RANDOM() or something more deterministic like a counter)
[just an example]:
exten s,1,Answer
exten s,n,Random(50:next)
exten s,n,AGI(agi://asterisk1/${VAR1}|${VAR2})
exten s,n,Hangup
exten s,n,AGI(agi://asterisk2/${VAR1}|${VAR2})
exten s,n,Hangup
--
Honi soit la vache qui rit.
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Re: [asterisk-users] Asterisk queue and agents
>Have a question for the group >If I have an agent is on the phone outside of the queue should that person >still get queue calls ? >Doing a show agents online I see Available however show hints I see inuse. There is a ringinuse feature for SIP devices on 1.4.X which is what you are looking for. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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] play music while continue executing dial plan
Perhaps MusicOnHold() app? > Is there any application can let the dial plan to execute while > playing music? Say I have a lot of command to do in the dial plan but > I don't want to keep silence while execution of dial plan. I notice > Background(file) can play music but it will hold until pressing a key. > I want something like background and it plays music with continuing > execute the rest of the command in dial plan. -- YOW!!! I am having fun!!! ___ --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] API: how to bridge originated call?
> in my case I want a user to be on-line all the time - the system will > dial and connect them and, when they're done, they select the next one. > what I'm doing now is putting them into a loop with a g-option on the > dial. the number it dials is set thru the api. if the number's not > set it waits one second and loops again. Why not use an agent channel AgentLogin()? Or I missunderstood your need? -- Honi soit la vache qui rit. ___ --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] CLI output to file
> Thanks! > But the information in /var/log/asterisk/messages is much different > from the messages in CLI. I want to log the message in CLI to file > for easy debugging. It is the same, see the "levels" in logger.conf, copy the console config to the messages config: (just for example) console => notice,warning,error,verbose,dtmf messages => notice,warning,error,verbose,dtmf And you will have the same output. :) > On 12/18/06, Octavio Ruiz (Ta^3) <[EMAIL PROTECTED]> wrote: > >> Hi all, > >> How can I redirect the CLI output to file without viewing it on > >> screen? Is it possible. > >Read and edit /etc/asterisk/logger.conf > >You should have already that output at /var/log/asterisk/messages. -- Octavio Ruiz Cervera Neocenter, SA. de CV. http://www.neocenter.com/ Soluciones para Centros de Contacto y Telefonía IP Tel.: (+52 55) 8590-9000 Ext. 9016 Cel.: (+55 55) 5514-087790 ___ --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] CLI output to file
> Hi all, > How can I redirect the CLI output to file without viewing it on > screen? Is it possible. Read and edit /etc/asterisk/logger.conf You should have already that output at /var/log/asterisk/messages. -- Oh, wow! Look at the moon! ___ --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] AgentCallbackLogin deprecated?
> > Yes, AgentCallbackLogin is deprecated, but it will not be removed > > until after 1.4. > > Is there an isolated example somewhere of how to use existing dialplan > logic and dynamic queue membership to simulate the current behaviour? http://svn.digium.com/view/asterisk/trunk/doc/queues-with-callback-members.txt > What about generation of statistics for callcentre monitoring? If this > is not taking place through chan_agent, won't it be reinventing the > wheel to have to simulate this behaviour, too? Why? Seems that reinventing the well was the agentcallbacklogin implementation, when it could be happend in dialplan logic. -- May I ask a question? ___ --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] queue management
> That's pretty easy and included in the basic * implementation - you tell > the queue not to accept users and play a message after the queue command > terminates. > l. Don't forget to analyze the QUEUESTATUS variable. :) -- Sign my PETITION. ___ --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] Can AGI do this?
Bret Schuhmacher, who happens to be smarter than you, thinks:
> Please pardon the absolute noob questions. Someone has asked me to
> interface with Asterisk and have it dial 4 numbers in succession
I dont understand this:
> to have
> it track down an on-call person.
> If not, is it possible to write an AGI program that gets all 4 numbers,
> then somehow hands them one-by-one to Asterisk? If so, how does
> Asterisk manage the communication of "failed to complete the call" with
> the AGI app? Does the AGI just monitor stdin looking for status
> messages and returns the next number?
>
> If Asterisk/AGI can do both, is the first method better than the
> second? It certainly seems easier.
The AGI script can die when it finished they job, probably I'm
missunderstanding what you want but maybe this examples helps:
exten => 123,1,Answer
exten => 123,n,AGI(somescript)
exten => 123,n,Dial(${TRUNK}/${FOURDIGITNUMBER})
exten => 123,n,Hangup
somescript: just sends AGI command SETVAR FOURDIGITNUMBER 2468
[context]
exten => s,1,Answer
exten => s,n... (set timeouts, et all).
exten => s,n,Background(audio)
exten => s,n,WaitExten
exten => _,1,Dial(${EXTEN})
exten => t,..
exten => h,...
--
Onward through the fog.
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