Re: [asterisk-users] Asynchronous play music
Hi,
Thanks a lot Steve, Im gonna try it in Asterisk 1.6.
Quyps
On Tue, 2010-03-30 at 12:30 -0700, Steve Edwards wrote:
> > On Tue, 30 Mar 2010, Pham Quy wrote:
> >
> >> Is there anyway to catch DTMF keypress while a music file is playing
> >> without stop the music?
>
> On Mon, 29 Mar 2010, Steve Edwards wrote:
>
> > Have you tried externivr(). I've never used it, but it looks
> > interesting.
>
> I hate it when people post about things they know nothing about...
>
> So I wrote My First ExternalIVR (mfe.c). I'm a 1.2 Luddite, so things may
> be a bit different for the Asterisk version you are using. Comparing the
> documentation, it looks like a lot of useful commands have been added.
>
> The first question I had was "Where do you put the executable?" The
> documentation is silent on this point. ASTAGIDIR seemed a likely guess.
> Guess not. It looks like it has to be an absolute path. Inconvenient, but
> it works.
>
> The actual protocol is trivial, way "simpler" and limited than AGI.
>
> Here's a snippet that shows how to play a file and then "handle" the key
> press events.
>
> // play a file
> printf("S,demo-congrats\n");
> fflush(stdout);
>
> // read events
> while (NULL != fgets(event, sizeof(event), stdin))
> {
> syslog(LOG_ERR, event);
> if ('#' == *event)
> {
> break;
> }
> }
>
> (The full source is at http://www.sedwards.com/mfe.c)
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
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Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource
nik600 wrote: > I was trying to record a call usng Mixmonitor and then convert it > using ffmpeg but the recording file is continuosly growing and ffmpeg > ends the conversion before of the call completion. Here's my quick and easy eagi script: #!/bin/sh cat /dev/fd/3 | sox -t raw -r 8000 -w -s -c 1 - -t raw -r 44100 - vol 2| ffmpeg -f s16le -ar 44100 -ac 1 -i - -ab 32k -f mp3 - | ezstream -c /var/lib/asterisk/ices/stream.mp3.xml It just dumps the audio through sox, to increase the volume a bit, and convert the sample rate, then ffmpeg to encode the mp3, and then ezstream to send it to an icecast server. I could probably skip the sox step, and get ffmpeg to do those adjustments on its own, but for now, I know sox's command line better, so I used that. :) The dialplan is as simple as exten => meetme,n,MeetMe(confname,1qd) put all the members of the conversation in there, exten => mp3stream,n,EAGI(mp3stream.sh) and then put this in as well to start recording. -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Confusion on call forwarding
> After setting promiscredir set to yes * is goign to send the call to > the first desrination on the Contact header. In case others run into the problem, the fix was not to set promiscredir to yes in sip.conf, but instead to set FORWARD_CONTEXT in the globals section of the dialplan. -- _ -- 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 hangup all outging calls after 32 seconds
(Sorry, but my english is not good) Hi, I have a problem with my new asterisk instalation. I search in google but I couldn't find nothing. Here's the thing. Before, we have 2 asterisk servers, each one with a E1 card. one with a Digium TE105 and the another with a A104 and we have a very simple setup. A Linux IBM X4300 Server is running CentOS 5.4 + Asterisk 1.6.2 (one month ago I download the latest ones). No analog cards. I can make calls between my extension perfectly; When I make outgoing calls using the PRI (any E1 PRI) in some phones the call hangup without any reaseon exactly after 32 seconds Where can be the problem or the error? Regards, -- Ing CIP. Alejandro Celi Mariátegui -- Este mensaje ha sido analizado por MailScanner en busca de virus y otros contenidos peligrosos, y se considera que est� limpio. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk system for church call center
On 29 March 2010 21:46, Frank Church wrote: > I have been asked by my church to recommend a VoIP system which can do > the following. > > They do internet radio shows which are sometimes broadcast on radio. > > They are looking for a system which does the following for about 5 > agents, exactly as they have described it. > > 1. Take incoming calls > > 2. Put them on hold if there is no one to handle the call immediately, > or transfer them to an available agent > > 3. Take down their details, and number, (if this can be retrieved and > saved from the caller id, thats better) > > 4. Get them to hold on after taking their details if they still want to hold > > 5. Call them back when the backlog is cleared up. > > I have a fairly good grasp of the hardware and programming part of > Asterisk, having compiled it more than a few times and implemented > A2Billing phone card and call shop system with it. > > But the type of software suited to the Call Center side is where my > knowledge gap lies. > > I am looking for solutions based on the usual Asterisk distributions > like AsteriskNow, trixbox, elastix etc, whether ready packaged or > requiring additional customization. > > > The matter of whether they will use soft phones, or regular phones > with headsets is also something to consider. Soft phones with good > GUI's may be preferred if more cost effective for them, although my > personal preferences are with hard phones. > > Any recommendations - the ease of software for the end users is the > main thing for me, and integration with the database for taking > customers details is the main thing for me. One of the distributions > with SugarCRM comes to mind here. > > Sorry for cross-posting, but ready made and commercially supported > systems are not ruled out, if they come within their budget. > > Regards > > > Frank Church > After there response I will go with some of ready made Asterisk distributions, then consider to go for a commercial supported versions if they do not meet the churches needs. Thanks Frank -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Asterisk system for church call center
On 30 March 2010 02:04, Mark Phillips wrote: > They say confession is good for the soul. Perhaps they are offering a > phone in confessional service? > > Unfortunately the "business" of the church often flies in the face of > the business of the Church. > > > > On 03/29/2010 07:48 PM, Alex Balashov wrote: >> Sounds like the church has strayed from its core competencies and >> invited the money-changers into the temple. >> >> Being the official asterisk-biz harbinger of God's wrath, I suggest an >> intensely commercial platform, for the meek shall inherit the Earth, >> not the 700 Club. Fight the power. >> > O ye of little faith, judge not, that ye be not judged!! > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confusion on call forwarding
After setting promiscredir set to yes * is goign to send the call to the first desrination on the Contact header. --Mensaje original-- De: Richard Kenner Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] Confusion on call forwarding Enviado: 30 Mar, 2010 14:56 > You need promiscredir set to yes on sip.conf And then what do I do in the dialplan? I.e., what context is the redirect number interpreted in? Google searches on this issue show inconsistent and contradictory information. -- _ -- 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 Saludos, Juan E. Rodríguez -- _ -- 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] app_txfax.c
I have installed app_txfax.c and spandsp-0.0.6. spandsp seemed to go fine. when compiling app_txfax.c I get errors. I am using centos x86_64 5.4. [CC] app_txfax.c -> app_txfax.o In file included from app_txfax.c:28: /home/silentm/MessageNet/digium/asterisk-1.4.30/include/asterisk/lock.h: In function ‘ast_mutex_init’: /home/silentm/MessageNet/digium/asterisk-1.4.30/include/asterisk/lock.h:738: warning: implicit declaration of function ‘pthread_mutexattr_settype’ /home/silentm/MessageNet/digium/asterisk-1.4.30/include/asterisk/lock.h:738: error: ‘PTHREAD_MUTEX_RECURSIVE’ undeclared (first use in this function) /home/silentm/MessageNet/digium/asterisk-1.4.30/include/asterisk/lock.h:738: error: (Each undeclared identifier is reported only once /home/silentm/MessageNet/digium/asterisk-1.4.30/include/asterisk/lock.h:738: error: for each function it appears in.) the PTHREAD_MUTEX_RECURSIVE is in /usr/include/pthread.h and around ifdefs for __USE_UNIX98 What do I do about this? Jerry -- _ -- 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] Dropped Calls
> I've written about this issue several times, but have not yet found any > solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones > are primarily Snom 300's but I also have a couple of headset phones > connected to Grandstream HT286 SIP adapters. I have 8 offices, each has > it's own asterisk server all running the same versions of asterisk and > Zaptel. Only difference is that one office uses a Digium TDM 8-port > card and the other branches use 4-port Rhino cards with only 2 ports in > use. What happens is that periodically we will be in a call and the > call will just drop. It's usually within the first couple of minutes of > the call. The calls can be either incoming or outgoing. The phenomenon > affects both the Snoms and the Grandstreams. Along with the dropped > call issue, we periodically have a problem where a person we call or a > person that calls in cannot hear the person in the our office, but the > person in our office can hear the remote person fine. > > All of the phones are on the same physical network as the asterisk > server. There is no NAT, no Firewall, VLAN, etc. between the phones and > the server. I have tried running sip debugs on the calls, but on the > off chance that my logs catch either a drop or a one-way audio, the sip > debug looks like just a normal call. > > Is there any setting that might cause both one-way audio and dropped calls? > > Thanks, > Brent Davidson Join the club. I've experienced the same with various strains on 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). This issue is truly random and debugging reveals nothing. I run an all SIP environment with same results. My solution was to downgrade to another version or switch to 1.2 or 1.6 depending on what features I need for the system. Sorry I couldn't be of any help, but I feel your frustration. JR -- JR Richardson Engineering for the Masses -- _ -- 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] Dropped Calls
On 3/30/2010 3:14 PM, Danny Nicholas wrote: > A few thoughts; > 1. I assume that the * servers aren't on dedicated networks; Do the dropped > or one-way calls occur during high-traffic times or are they concurrent with > large downloads? In my shop, we had to get a router that would prioritize > voice traffic or we would be dead in the water during client file > transmissions. > Asterisk servers are not on a dedicated network, but our total network utilization is less than 10% max at any time. > 2. Don't know about the SNOM or GS phones, but my Polycom phones let you > establish higher packet priorities for voice traffic as well. > I have all the phones, the asterisk server and the core switch set to prioritize RTP and SIP packets at top priority. But I never see any indication of dropped or delayed packets in the logs. > 3. Have you been able to do a "top" during one of these failures? Could be > a memory leak that comes up randomly. > This one is a tough one. When these types of calls occur it is completely random. Sometimes there will be one or two in a row, other times there won't be one for a couple of days. It would take some some serious logging to catch top data at the exact moment one of the calls drops or the one-way audio hits. > 4. Looking at the startup logs, are the cards having to retry several times > to get an IRQ? Digium cards IME can conflict with the Hard Drive (SCSI) > controller, causing problems during heavy I/O periods. > Hope this helps Cards all get an IRQ on the first try. Other data of interest: Our main office only has 8 incoming analog lines, the other offices all only have 2 incoming lines, and there is no correlation between calls in progress and and either of the problems. Sometimes the main office will have two or three in-progress calls and another incoming or outgoing call will experience one-way audio or a disconnect and the others are unaffected. Not even a glitch in the audio. I have had both problems happen to me after hours when I was the only one in the office so the network was completely idle and my call was the only one active. I've been trying to trace this problem for about two years and still have not been able to make any real progress. I guess I should just update to Dahdi and Asterisk 1.6, but I just hate to change a system that is (mostly) working. -- _ -- 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] E1 card w/o echo cancellation
I am planning to use an E1 card from Openvox w/o echo cancellation. Please share yoru experiences about the quality of sound quality w/o hardware echo cancellation. Ehsan -- _ -- 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] Priority based softhangup
Thanks Steve and Danny for your help.
S Mir
On Tue, Mar 30, 2010 at 3:16 PM, Steve Edwards
wrote:
> Un-top-posting...
>
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir
>> shahnawaz
>>
>> Can you please give me idea about Softhang.agi
>
> On Tue, 30 Mar 2010, Danny Nicholas wrote:
>
>> Simple code
>> In PERL -
>> -- agi header stuff --
>> My ($pri, $callno) = @ARGV;
>> -- read database to get priority of lowest active call, return $lowpri,
>> $lowid --
>> if ($pri > $lowpri) {
>> print STDOUT "exec softhangup $lowid\n";
>> -- AMI dial $callno --
>> }
>> Else {
>> Print STDOUT "exec BACKGROUND congested message\n";
>> }
>>
>> You can find real examples on voip-info.org.
>
> mir, please keep in mind the above is meant to "give you an idea," not
> intended as literal code.
>
> You will save yourself a lot of frustration if you use an existing AGI
> "library" for your chosen language*. While conceptually simple, nobody
> gets the AGI protocol right the first time.
>
> *) I write most of my AGIs in C because it is the language I know best and
> you can execute hundreds of AGIs written in C in the time it takes to load
> the Perl or PHP interpreter and parse your script.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- 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
>
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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
And not to mention the need for power over ethernet switches to avoid having many power adpaters lying all over. Don't get me wrong, I'm for IP Phones, however, in this specific scenario that I have, getting an FXS to SIP gateway with 24 ports makes more sense. Thanks for all the pointers. Best Regards, Joseph On Tue, Mar 30, 2010 at 8:39 AM, Andrew Latham wrote: > And to add to this, analog is useful for its distance when running > wall phones in a large warehouse setting... > > > ~ > Andrew "lathama" Latham > lath...@gmail.com > > * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software > * Learn more about Linux http://en.wikipedia.org/wiki/Linux > * Learn more about Tux http://en.wikipedia.org/wiki/Tux > > > > On Tue, Mar 30, 2010 at 11:29 AM, Darrick Hartman > wrote: > > Sometimes you need to look at the cost to pull new wire too, not just the > cost of the phones. There are a few cases where the channel banks + analog > phones make sense, especially when the analog devices are already there. > > Sent from my BlackBerry® wireless device from U.S. Cellular > > > > -Original Message- > > From: hin lee > > Date: Tue, 30 Mar 2010 08:25:19 > > To: Asterisk Users Mailing List - Non-Commercial Discussion< > asterisk-users@lists.digium.com> > > Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Priority based softhangup
Un-top-posting...
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir
> shahnawaz
>
> Can you please give me idea about Softhang.agi
On Tue, 30 Mar 2010, Danny Nicholas wrote:
> Simple code
> In PERL -
> -- agi header stuff --
> My ($pri, $callno) = @ARGV;
> -- read database to get priority of lowest active call, return $lowpri,
> $lowid --
> if ($pri > $lowpri) {
> print STDOUT "exec softhangup $lowid\n";
> -- AMI dial $callno --
> }
> Else {
> Print STDOUT "exec BACKGROUND congested message\n";
> }
>
> You can find real examples on voip-info.org.
mir, please keep in mind the above is meant to "give you an idea," not
intended as literal code.
You will save yourself a lot of frustration if you use an existing AGI
"library" for your chosen language*. While conceptually simple, nobody
gets the AGI protocol right the first time.
*) I write most of my AGIs in C because it is the language I know best and
you can execute hundreds of AGIs written in C in the time it takes to load
the Perl or PHP interpreter and parse your script.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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[asterisk-users] E-mails from Asterisk coming from root
Ever since the upgrade to Asterisk 1.6 the e-mails from Asterisk are coming from r...@. In the voicemail.conf I have fromstring=Asterisk PBX serveremail=asterisk And in my ssmtp.conf root=asterisk However they still come from root@ -- _ -- 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] convert from wav or mp3 to gsm
AIR, * uses wav and gsm with no trouble. Mpg123 plays mp3 format files. You can use LAME and SOX to change files between these formats. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine elharit Sent: Tuesday, March 30, 2010 3:17 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] convert from wav or mp3 to gsm Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- 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] convert from wav or mp3 to gsm
Hello All do you have ant software in order to change the format from mp3 or wav to gsm in order to using it in asterisk file thank you so much for your help and support Best Regards, salah -- _ -- 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] Dropped Calls
A few thoughts; 1. I assume that the * servers aren't on dedicated networks; Do the dropped or one-way calls occur during high-traffic times or are they concurrent with large downloads? In my shop, we had to get a router that would prioritize voice traffic or we would be dead in the water during client file transmissions. 2. Don't know about the SNOM or GS phones, but my Polycom phones let you establish higher packet priorities for voice traffic as well. 3. Have you been able to do a "top" during one of these failures? Could be a memory leak that comes up randomly. 4. Looking at the startup logs, are the cards having to retry several times to get an IRQ? Digium cards IME can conflict with the Hard Drive (SCSI) controller, causing problems during heavy I/O periods. Hope this helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent Davidson Sent: Tuesday, March 30, 2010 2:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dropped Calls I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Priority based softhangup
Simple code
In PERL -
-- agi header stuff --
My ($pri, $callno) = @ARGV;
-- read database to get priority of lowest active call, return $lowpri,
$lowid --
if ($pri > $lowpri) {
print STDOUT "exec softhangup $lowid\n";
-- AMI dial $callno --
}
Else {
Print STDOUT "exec BACKGROUND congested message\n";
}
You can find real examples on voip-info.org.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz
Sent: Tuesday, March 30, 2010 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Priority based softhangup
Thanks Danny,
Can you please give me idea about Softhang.agi
Thanks
Smir
On Tue, Mar 30, 2010 at 1:31 PM, Danny Nicholas wrote:
> AIUI, softhangup is strictly an address-type function. Using Steve's
> suggestion, you could set a key with the priority at dial-time and when a
> congested condition occurred, match the database for the lowest priority
to
> hangup. Something like this:
> Exten => 100,1,noop(prioritized dialing)
> Exten => 100,n,Set(callpri=${DB(Callpri/${EXTEN})})
> Exten => 100,n,Dial..
> Exten => 100-CONGESTION(softhang.agi,${callpri})
>
> Softhang.agi is a program you write to see if the callpri is high enough
to
> cancel another call and dial or send back a message.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
Edwards
> Sent: Tuesday, March 30, 2010 2:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Priority based softhangup
>
> On Tue, 30 Mar 2010, mir shahnawaz wrote:
>
>> Is it possible to softhangup a channel based on priority. I mean I want
>> to put some calls in higher priority lets say 100. If all channels are
>> busy and somebody wants to dial an extension with priority higher than
>> 100. How can softhangup drop a line which has priority less than 100? I
>> will appreciate your valuable help.
>
> In Asterisk, a "priority" is the "step number" in a dialplan. For example:
>
> exten = *,5, verbose(1,foo)
>
> "5" is the priority.
>
> It sounds like you want to have an "executive" class that can step on an
> "underling's" call. Maybe setting a global channel variable or tracking
> resource usage in a database would work.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- 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
>
>
> --
> _
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> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] Dropped Calls
I've written about this issue several times, but have not yet found any solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones are primarily Snom 300's but I also have a couple of headset phones connected to Grandstream HT286 SIP adapters. I have 8 offices, each has it's own asterisk server all running the same versions of asterisk and Zaptel. Only difference is that one office uses a Digium TDM 8-port card and the other branches use 4-port Rhino cards with only 2 ports in use. What happens is that periodically we will be in a call and the call will just drop. It's usually within the first couple of minutes of the call. The calls can be either incoming or outgoing. The phenomenon affects both the Snoms and the Grandstreams. Along with the dropped call issue, we periodically have a problem where a person we call or a person that calls in cannot hear the person in the our office, but the person in our office can hear the remote person fine. All of the phones are on the same physical network as the asterisk server. There is no NAT, no Firewall, VLAN, etc. between the phones and the server. I have tried running sip debugs on the calls, but on the off chance that my logs catch either a drop or a one-way audio, the sip debug looks like just a normal call. Is there any setting that might cause both one-way audio and dropped calls? Thanks, Brent Davidson -- _ -- 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] Priority based softhangup
Thanks Danny,
Can you please give me idea about Softhang.agi
Thanks
Smir
On Tue, Mar 30, 2010 at 1:31 PM, Danny Nicholas wrote:
> AIUI, softhangup is strictly an address-type function. Using Steve's
> suggestion, you could set a key with the priority at dial-time and when a
> congested condition occurred, match the database for the lowest priority to
> hangup. Something like this:
> Exten => 100,1,noop(prioritized dialing)
> Exten => 100,n,Set(callpri=${DB(Callpri/${EXTEN})})
> Exten => 100,n,Dial..
> Exten => 100-CONGESTION(softhang.agi,${callpri})
>
> Softhang.agi is a program you write to see if the callpri is high enough to
> cancel another call and dial or send back a message.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
> Sent: Tuesday, March 30, 2010 2:09 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Priority based softhangup
>
> On Tue, 30 Mar 2010, mir shahnawaz wrote:
>
>> Is it possible to softhangup a channel based on priority. I mean I want
>> to put some calls in higher priority lets say 100. If all channels are
>> busy and somebody wants to dial an extension with priority higher than
>> 100. How can softhangup drop a line which has priority less than 100? I
>> will appreciate your valuable help.
>
> In Asterisk, a "priority" is the "step number" in a dialplan. For example:
>
> exten = *,5, verbose(1,foo)
>
> "5" is the priority.
>
> It sounds like you want to have an "executive" class that can step on an
> "underling's" call. Maybe setting a global channel variable or tracking
> resource usage in a database would work.
>
> --
> Thanks in advance,
> -
> Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
> Newline Fax: +1-760-731-3000
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Asynchronous play music
Guess you hate me about half the time :). Gonna try mfe.c with 1.4.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, March 30, 2010 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asynchronous play music
> On Tue, 30 Mar 2010, Pham Quy wrote:
>
>> Is there anyway to catch DTMF keypress while a music file is playing
>> without stop the music?
On Mon, 29 Mar 2010, Steve Edwards wrote:
> Have you tried externivr(). I've never used it, but it looks
> interesting.
I hate it when people post about things they know nothing about...
So I wrote My First ExternalIVR (mfe.c). I'm a 1.2 Luddite, so things may
be a bit different for the Asterisk version you are using. Comparing the
documentation, it looks like a lot of useful commands have been added.
The first question I had was "Where do you put the executable?" The
documentation is silent on this point. ASTAGIDIR seemed a likely guess.
Guess not. It looks like it has to be an absolute path. Inconvenient, but
it works.
The actual protocol is trivial, way "simpler" and limited than AGI.
Here's a snippet that shows how to play a file and then "handle" the key
press events.
// play a file
printf("S,demo-congrats\n");
fflush(stdout);
// read events
while (NULL != fgets(event, sizeof(event), stdin))
{
syslog(LOG_ERR, event);
if ('#' == *event)
{
break;
}
}
(The full source is at http://www.sedwards.com/mfe.c)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Priority based softhangup
AIUI, softhangup is strictly an address-type function. Using Steve's
suggestion, you could set a key with the priority at dial-time and when a
congested condition occurred, match the database for the lowest priority to
hangup. Something like this:
Exten => 100,1,noop(prioritized dialing)
Exten => 100,n,Set(callpri=${DB(Callpri/${EXTEN})})
Exten => 100,n,Dial..
Exten => 100-CONGESTION(softhang.agi,${callpri})
Softhang.agi is a program you write to see if the callpri is high enough to
cancel another call and dial or send back a message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, March 30, 2010 2:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Priority based softhangup
On Tue, 30 Mar 2010, mir shahnawaz wrote:
> Is it possible to softhangup a channel based on priority. I mean I want
> to put some calls in higher priority lets say 100. If all channels are
> busy and somebody wants to dial an extension with priority higher than
> 100. How can softhangup drop a line which has priority less than 100? I
> will appreciate your valuable help.
In Asterisk, a "priority" is the "step number" in a dialplan. For example:
exten = *,5,verbose(1,foo)
"5" is the priority.
It sounds like you want to have an "executive" class that can step on an
"underling's" call. Maybe setting a global channel variable or tracking
resource usage in a database would work.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] Asynchronous play music
> On Tue, 30 Mar 2010, Pham Quy wrote:
>
>> Is there anyway to catch DTMF keypress while a music file is playing
>> without stop the music?
On Mon, 29 Mar 2010, Steve Edwards wrote:
> Have you tried externivr(). I've never used it, but it looks
> interesting.
I hate it when people post about things they know nothing about...
So I wrote My First ExternalIVR (mfe.c). I'm a 1.2 Luddite, so things may
be a bit different for the Asterisk version you are using. Comparing the
documentation, it looks like a lot of useful commands have been added.
The first question I had was "Where do you put the executable?" The
documentation is silent on this point. ASTAGIDIR seemed a likely guess.
Guess not. It looks like it has to be an absolute path. Inconvenient, but
it works.
The actual protocol is trivial, way "simpler" and limited than AGI.
Here's a snippet that shows how to play a file and then "handle" the key
press events.
// play a file
printf("S,demo-congrats\n");
fflush(stdout);
// read events
while (NULL != fgets(event, sizeof(event), stdin))
{
syslog(LOG_ERR, event);
if ('#' == *event)
{
break;
}
}
(The full source is at http://www.sedwards.com/mfe.c)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
_
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Re: [asterisk-users] Priority based softhangup
On Tue, 30 Mar 2010, mir shahnawaz wrote: > Is it possible to softhangup a channel based on priority. I mean I want > to put some calls in higher priority lets say 100. If all channels are > busy and somebody wants to dial an extension with priority higher than > 100. How can softhangup drop a line which has priority less than 100? I > will appreciate your valuable help. In Asterisk, a "priority" is the "step number" in a dialplan. For example: exten = *,5,verbose(1,foo) "5" is the priority. It sounds like you want to have an "executive" class that can step on an "underling's" call. Maybe setting a global channel variable or tracking resource usage in a database would work. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Confusion on call forwarding
> You need promiscredir set to yes on sip.conf And then what do I do in the dialplan? I.e., what context is the redirect number interpreted in? Google searches on this issue show inconsistent and contradictory information. -- _ -- 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] DAHDI 2.2.1, Asterisk 1.6.2.6 - Channel unacceptable (6)
Hi, we have a problem with a Digium TE122 card, DAHDI 2.2.1 and Asterisk 1.6.2.6 in the UK. Outgoing calls do work without problems, but each incoming call fails: Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 1/0x1) (Originator) Message type: RELEASE (77) [08 02 82 86] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Channel unacceptable (6), class = Normal Event (0) ] Processing IE 8 (cs0, Cause) q931.c:3801 q931_receive: call 1 on channel 1 enters state 0 (Null) Channel 0/1, span 1 got hangup, cause 6 The dial plan just consists of a Ringing() and a Wait(2). Most configuration errors probably should affect outgoing calls as well. A working configruation example would be very helpfull. Any help appreciated -- Stefan Tichy ( asterisk2 at pi4tel dot de ) -- _ -- 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] Priority based softhangup
Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir -- _ -- 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] dnd not working correctly (resolved)
Dave figured it out.
looks like i didn't configure the sounds/extra sounds when i did the asterisk
compile.
Thanks Dave.
Date: Tue, 30 Mar 2010 10:42:48 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
yes, that's the default location unless there's any change in
/etc/asterisk/asterisk.conf
I think it should be there, cause when it is not asterisk complains with a
message letting you know it wasn't able to find it.
>I dont have a working server to look at so i didn't know if i was even looking
>in the right place.
Pls look for them in the server you are actually having the problems with cause
I can't remember that sound file being on the official's asterisk release.
Alyed
2010/3/30 Ott Rose
where are those sound files kept? i looked last night in
/var/lib/asterisk/sounds and i didn't see anything named do-not-disturb.
if its supposed to be in there then thats a problem. I dont have a working
server to look at so i didn't know if i was even looking in the right place.
Date: Mon, 29 Mar 2010 23:58:43 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
I'm not an Amportal expert so all I can say from:
> -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
"do-not-disturb&activated") in new stack
> -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
"hangupcall,") in new stack
is that Asterisk is playing the "do-not-disturb&activated" file (apparently
without errors) and then the next instruction is to hangup the call, hence
Asterisk hangs it up.
Just to be sure play this sound file independently.
Sorry but other than this there's little I can do, maybe someone else has
experience with this.
Alyed
2010/3/29 Ott Rose
i posted this on the freepbx site. here is the response
"from the trace, everything is working. Check your asterisk log for file
errors playing back the audio, could be your sound files are not
installed or messed up."
so i checked /etc/log/asterisk/full
and in vi full i did /error and /117 (my ext) and /activate didn't really
find anything
i didn't see anything but i might be over looking it. I did grep error full and
it returned some errors but not related to dnd as far as i can tell. is there
some place else to look, a better way to search that file, or can someone tell
me what i am looking for?
Date: Fri, 26 Mar 2010 18:34:46 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
Seems like an Amportal configration problem not and Asterisk issue. Maybe you
should try in one of the FreePBX users list.
Alyed
2010/3/26 Ott Rose
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Extension Changed 117[ext-local] new state InUse for Notify User 102
== Extension Changed 117[ext-local] new state InUse for Notify User 103
== Extension Changed 117[ext-local] new state InUse for Notify User 114
-- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in new
stack
-- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in new
stack
-- Executing [...@from-internal:3] Macro("SIP/117-01f6",
"user-callerid,") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
"AMPUSER=117") in new stack
-- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
"0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
"1?Set(REALCALLERIDNUM=117)") in new stack
-- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
"AMPUSER=117") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
"AMPUSERCIDNAME=My Name") in new stack
-- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
"0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
"AMPUSERCID=117") in new stack
-- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
"CALLERID(all)="My Name" <117>") in new stack
-- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
"0?continue") in new stack
-- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
"__TTL=64") in new stack
-- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6", "Using
CallerID "My Name" <117>") in new stack
-- Executing [...@from-internal:4] GotoIf(
Re: [asterisk-users] Confusion on call forwarding
You need promiscredir set to yes on sip.conf --Mensaje original-- De: Richard Kenner Remitente: asterisk-users-boun...@lists.digium.com Para: asterisk-users@lists.digium.com Responder a: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [asterisk-users] Confusion on call forwarding Enviado: 30 Mar, 2010 13:38 I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue. -- _ -- 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 Saludos, Juan E. Rodríguez -- _ -- 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] Confusion on call forwarding
I'm confused. What does Asterisk do when it gets a 302 with a new number to forward to? Is there anything I have to do in the dialplan to make this work? I can't find any clear documentation on this issue. -- _ -- 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] dnd not working correctly
yes, that's the default location unless there's any change in
/etc/asterisk/asterisk.conf
I think it should be there, cause when it is not asterisk complains with a
message letting you know it wasn't able to find it.
>I dont have a working server to look at so i didn't know if i was even
looking in the right place.
Pls look for them in the server you are actually having the problems with
cause I can't remember that sound file being on the official's asterisk
release.
Alyed
2010/3/30 Ott Rose
> where are those sound files kept? i looked last night in
> /var/lib/asterisk/sounds and i didn't see anything named do-not-disturb.
>
> if its supposed to be in there then thats a problem. I dont have a working
> server to look at so i didn't know if i was even looking in the right place.
>
> --
> Date: Mon, 29 Mar 2010 23:58:43 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> I'm not an Amportal expert so all I can say from:
>
> > -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
> "do-not-disturb&activated") in new stack
> > -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
> "hangupcall,") in new stack
>
> is that Asterisk is playing the "do-not-disturb&activated" file (apparently
> without errors) and then the next instruction is to hangup the call, hence
> Asterisk hangs it up.
>
> Just to be sure play this sound file independently.
>
> Sorry but other than this there's little I can do, maybe someone else has
> experience with this.
>
> Alyed
>
>
> 2010/3/29 Ott Rose
>
>
> i posted this on the freepbx site. here is the response
>
>
> "from the trace, everything is working. Check your asterisk log for file
> errors playing back the audio, could be your sound files are not installed
> or messed up."
>
>
>
> so i checked /etc/log/asterisk/full
>
> and in vi full i did /error and /117 (my ext) and /activate didn't
> really find anything
>
> i didn't see anything but i might be over looking it. I did grep error full
> and it returned some errors but not related to dnd as far as i can tell. is
> there some place else to look, a better way to search that file, or can
> someone tell me what i am looking for?
>
>
>
>
> --
> Date: Fri, 26 Mar 2010 18:34:46 -0600
> From: al...@vivoxie.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] dnd not working correctly
>
> Seems like an Amportal configration problem not and Asterisk issue. Maybe
> you should try in one of the FreePBX users list.
>
> Alyed
>
>
>
> 2010/3/26 Ott Rose
>
> i have posted this question couple of times and never really got any hits
> i wasn't able to provide any debug info
>
> Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid =
> 3309)
> Verbosity is at least 4
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> == Using SIP VRTP TOS bits 136
> == Using SIP VRTP CoS mark 6
> == Extension Changed 117[ext-local] new state InUse for Notify User 102
> == Extension Changed 117[ext-local] new state InUse for Notify User 103
> == Extension Changed 117[ext-local] new state InUse for Notify User 114
> -- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in
> new stack
> -- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in
> new stack
> -- Executing [...@from-internal:3] Macro("SIP/117-01f6",
> "user-callerid,") in new stack
> -- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
> "1?Set(REALCALLERIDNUM=117)") in new stack
> -- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
> "AMPUSER=117") in new stack
> -- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
> "AMPUSERCIDNAME=My Name") in new stack
> -- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
> "0?report") in new stack
> -- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
> "AMPUSERCID=117") in new stack
> -- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
> "CALLERID(all)="My Name" <117>") in new stack
> -- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
> "0?continue") in new stack
> -- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
> "__TTL=64") in new stack
> -- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
> "1?continue") in new stack
> -- Goto (macro-user-callerid,s,18)
> -- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6",
> "Using CallerID "My Name" <117>") in new stack
> -- Executing [...@from-internal:4] GotoIf("SIP/117-01f6",
> "1?activate:deactivate") in new stack
> -- Goto (from-internal,*76,5)
> -- Exec
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
On Sun, 28 Mar 2010, Joseph Begumisa wrote: > Hi, > > Can anyone recommend a 24 fxs port voip gateway that has worked well with > asterisk? I have a couple of analog handsets that I want to hookup to my > asterisk server? Any tested and tried product recommendations are welcome. > Thanks. Any reason to not use a plug-in card? TDM2400P or , etc. ? Gordon -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
And to add to this, analog is useful for its distance when running wall phones in a large warehouse setting... ~ Andrew "lathama" Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Mar 30, 2010 at 11:29 AM, Darrick Hartman wrote: > Sometimes you need to look at the cost to pull new wire too, not just the > cost of the phones. There are a few cases where the channel banks + analog > phones make sense, especially when the analog devices are already there. > Sent from my BlackBerry® wireless device from U.S. Cellular > > -Original Message- > From: hin lee > Date: Tue, 30 Mar 2010 08:25:19 > To: Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Sometimes you need to look at the cost to pull new wire too, not just the cost of the phones. There are a few cases where the channel banks + analog phones make sense, especially when the analog devices are already there. Sent from my BlackBerry® wireless device from U.S. Cellular -Original Message- From: hin lee Date: Tue, 30 Mar 2010 08:25:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FAX 2 mail configuration
Hello Im trying to configure Fax2Mail in my Asterisk 1.4.23.1 server, wich receievs the Faxes through a SIP trunk. I found a lot of solutions in voip-info.org So, I would like to know what's the best free Fax2Mail solution and if I really need to install Dahdi or Zaptel. Thanks Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
With the price of FXS gateway, why not just get SIP phones? Polycom 330 is around $60-$110 a piece. From: mir shahnawaz To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tue, March 30, 2010 6:46:20 AM Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk Xorcom XR005 is highly recommended. They work great. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=120550484883&ssPageName=STRK:MESELX:IT Smir On Tue, Mar 30, 2010 at 7:34 AM, Juan Miguel wrote: > Hello Joseph > > I recommend that you use The Mediatrix 4100 Series are very good. > > Juan M. > > 2010/3/28 Joseph Begumisa >> >> Hi, >> Can anyone recommend a 24 fxs port voip gateway that has worked well with >> asterisk? I have a couple of analog handsets that I want to hookup to my >> asterisk server? Any tested and tried product recommendations are welcome. >> Thanks. >> Best Regards, >> >> Joseph >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue a dialplan when the client hang up the call
you might want to use the extension h. See http://www.voip-info.org/wiki/view/Asterisk+config+extensions.conf . huu giang schrieb: > Hi all, > > When a user make a call to Asterisk, and when user hang up the call at > any point of the conversation, Asterisk will stop Diaplan > intermediately. > > At this situation, Are there any way to make Asterisk continue > execute the Diaplan ?, so Asterisk can do something like that delete > temporary file, .. etc. > > Thanks in advance, > Giang > > -- _ -- 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] Minimalize jitter in VoIP calls
Hello list, I have set the tos-settings in sip.conf as recommended at http://www.voip-info.org/wiki/view/Asterisk+sip+tos : sip.conf tos_sip cs3 sip.conf tos_audio ef But there is still jitter and audio delay. What other measures can I take ?? Zoiper softphone --> D-Link router --> ADSL (ISP) --> Asterisk-server --> ITSP --> rest of the world The same TOS-settings for sip and audio are set in the Zoiper softphone. On the workstation there is some minimal web browsing, no hardcore downloading or file transfer. Kind regards. On Tue, 2010-03-23 at 17:21 +0100, jonas kellens wrote: > Hello list, > > what can I do to minimalize the jitter in SIP-calls at server level ? > > If at local network level, there is a VoIP-router and their is a > physical network dedicated to IP-phones, but there is still jitter. > > When using a Hosted Asterisk server, which settings on the > Asterisk-server can minimalize the jitter between the VoIP-router and > the Asterisk-server on the public internet ?? > > > Kind regards, > > Jonas. -- _ -- 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] 24 FXS Port Voip Gateway and Asterisk
Xorcom XR005 is highly recommended. They work great. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=120550484883&ssPageName=STRK:MESELX:IT Smir On Tue, Mar 30, 2010 at 7:34 AM, Juan Miguel wrote: > Hello Joseph > > I recommend that you use The Mediatrix 4100 Series are very good. > > Juan M. > > 2010/3/28 Joseph Begumisa >> >> Hi, >> Can anyone recommend a 24 fxs port voip gateway that has worked well with >> asterisk? I have a couple of analog handsets that I want to hookup to my >> asterisk server? Any tested and tried product recommendations are welcome. >> Thanks. >> Best Regards, >> >> Joseph >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dnd not working correctly
where are those sound files kept? i looked last night in
/var/lib/asterisk/sounds and i didn't see anything named do-not-disturb.
if its supposed to be in there then thats a problem. I dont have a working
server to look at so i didn't know if i was even looking in the right place.
Date: Mon, 29 Mar 2010 23:58:43 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
I'm not an Amportal expert so all I can say from:
> -- Executing [...@from-internal:8] Playback("SIP/117-01f6",
"do-not-disturb&activated") in new stack
> -- Executing [...@from-internal:9] Macro("SIP/117-01f6",
"hangupcall,") in new stack
is that Asterisk is playing the "do-not-disturb&activated" file (apparently
without errors) and then the next instruction is to hangup the call, hence
Asterisk hangs it up.
Just to be sure play this sound file independently.
Sorry but other than this there's little I can do, maybe someone else has
experience with this.
Alyed
2010/3/29 Ott Rose
i posted this on the freepbx site. here is the response
"from the trace, everything is working. Check your asterisk log for file
errors playing back the audio, could be your sound files are not
installed or messed up."
so i checked /etc/log/asterisk/full
and in vi full i did /error and /117 (my ext) and /activate didn't really
find anything
i didn't see anything but i might be over looking it. I did grep error full and
it returned some errors but not related to dnd as far as i can tell. is there
some place else to look, a better way to search that file, or can someone tell
me what i am looking for?
Date: Fri, 26 Mar 2010 18:34:46 -0600
From: al...@vivoxie.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] dnd not working correctly
Seems like an Amportal configration problem not and Asterisk issue. Maybe you
should try in one of the FreePBX users list.
Alyed
2010/3/26 Ott Rose
i have posted this question couple of times and never really got any hits i
wasn't able to provide any debug info
Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309)
Verbosity is at least 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
== Extension Changed 117[ext-local] new state InUse for Notify User 102
== Extension Changed 117[ext-local] new state InUse for Notify User 103
== Extension Changed 117[ext-local] new state InUse for Notify User 114
-- Executing [...@from-internal:1] Answer("SIP/117-01f6", "") in new
stack
-- Executing [...@from-internal:2] Wait("SIP/117-01f6", "1") in new
stack
-- Executing [...@from-internal:3] Macro("SIP/117-01f6",
"user-callerid,") in new stack
-- Executing [...@macro-user-callerid:1] Set("SIP/117-01f6",
"AMPUSER=117") in new stack
-- Executing [...@macro-user-callerid:2] GotoIf("SIP/117-01f6",
"0?report") in new stack
-- Executing [...@macro-user-callerid:3] ExecIf("SIP/117-01f6",
"1?Set(REALCALLERIDNUM=117)") in new stack
-- Executing [...@macro-user-callerid:4] Set("SIP/117-01f6",
"AMPUSER=117") in new stack
-- Executing [...@macro-user-callerid:5] Set("SIP/117-01f6",
"AMPUSERCIDNAME=My Name") in new stack
-- Executing [...@macro-user-callerid:6] GotoIf("SIP/117-01f6",
"0?report") in new stack
-- Executing [...@macro-user-callerid:7] Set("SIP/117-01f6",
"AMPUSERCID=117") in new stack
-- Executing [...@macro-user-callerid:8] Set("SIP/117-01f6",
"CALLERID(all)="My Name" <117>") in new stack
-- Executing [...@macro-user-callerid:9] GotoIf("SIP/117-01f6",
"0?continue") in new stack
-- Executing [...@macro-user-callerid:10] Set("SIP/117-01f6",
"__TTL=64") in new stack
-- Executing [...@macro-user-callerid:11] GotoIf("SIP/117-01f6",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,18)
-- Executing [...@macro-user-callerid:18] NoOp("SIP/117-01f6", "Using
CallerID "My Name" <117>") in new stack
-- Executing [...@from-internal:4] GotoIf("SIP/117-01f6",
"1?activate:deactivate") in new stack
-- Goto (from-internal,*76,5)
-- Executing [...@from-internal:5] Set("SIP/117-01f6",
"DB(DND/117)=YES") in new stack
-- Executing [...@from-internal:6] Set("SIP/117-01f6", "STATE=BUSY") in
new stack
-- Executing [...@from-internal:7] Gosub("SIP/117-01f6",
"app-dnd-toggle,sstate,1") in new stack
-- Executing [sst...@app-dnd-toggle:1] Set("SIP/117-01f6",
"DEVICE_STATE(Custom:DND117)=BUSY") in new stack
-- Executing [sst...@app-dnd-toggle:2] Set("SIP/117-01f6",
"DEVICES=117") in new stack
-- Executing [sst...@app-dnd-toggle:3] GotoIf("SIP/117-01f6",
"0?return") in new stack
== Extension Changed 117[ext-local] new state Busy for Notify User 102
-- Executing [sst...@app-
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Hello Joseph I recommend that you use *The Mediatrix 4100 Series* are very good. Juan M. 2010/3/28 Joseph Begumisa > Hi, > > Can anyone recommend a 24 fxs port voip gateway that has worked well with > asterisk? I have a couple of analog handsets that I want to hookup to my > asterisk server? Any tested and tried product recommendations are welcome. > Thanks. > > Best Regards, > > Joseph > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
I have used Rhinos for a while and they are very stable and work well with
asterisk... You need a T1 port port though
I also just bought a Xorcom and that is working very well too... (This is USB
so no need for a hardware card)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Lamanna
Sent: Sunday, March 28, 2010 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
On Sun, Mar 28, 2010 at 8:28 PM, Steve Edwards
wrote:
> On Sun, 28 Mar 2010, Joseph Begumisa wrote:
>
>> Can anyone recommend a 24 fxs port voip gateway that has worked well
>> with asterisk? I have a couple of analog handsets that I want to
>> hookup to my asterisk server? Any tested and tried product recommendations
>> are welcome.
>> Thanks.
>
> Adtran channel banks are a great "trailing edge" technology. You can
> get them off Ebay for pennies on the original dollar and they are
> built like a tank.
>
> ("voip gateway" is not very specific. If you meant SIP or IAX, you
> might want to specify which.)
I've actually had decent success with the GXW-4024 (FXS <-> SIP) from
Grandstream which is probably one of the cheapest 24 FXS port boxes you'll find
out there.
-- James
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk realtime ldap:active directory
hello, I try to query an active directory from asterisk. I do not see how to do it because I just started in asterisk. In extensions.conf switch => Realtime/defa...@realtime_ext In res_ldap.conf [general] host = ip_active directory protocol = 3 port = 389 dn = ou=voip,dc=test,dc=com pass = administrator_pass_activedirectory user = ou=Users,cn=Administrateur,dc=test,dc=com URL = ldap://ip_activedirectory:389/ou=voip,dc=test,dc=com [extensions] attribute = name =>uid attribute = accountcode => uid attribute = callerid => cn In extconfig.conf [settings] realtime_ext => ldap,dc=test,dc=com,extensions sipusers => ldap,dc=mydomain,dc=com,sipusers sippeers => ldap,dc=mydomain,dc=com,sippeers when i try to register save a user I get the following message : Warning[1952]: config.c:2025 find_engine:Realtime mapping for 'sippers' found to engine 'ldap' but the engine is not available Notice[1952] : chan_sip.c:21500 handle_request_register:Registration from '"user"' failed for 'ip_activedirectory' no matching peer found. I see no potential source or error in the configuration if you have a idea i'm willing Thank You Best Regards Mickael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Asterisk system for church call center
On 30 March 2010 02:04, Mark Phillips wrote: > They say confession is good for the soul. Perhaps they are offering a > phone in confessional service? > > Unfortunately the "business" of the church often flies in the face of > the business of the Church. > > I think you'll find a lot of Church Based Charities, Consoling and Advise Lines needing very high capacity phone systems out there. Asterisk due to it being Free is an ideal solution for this purpose. The Church primary commission is to tell the world. So use of modern technology is the ideal method of getting this job done. Its not about money its about evangelism. Peter. > > On 03/29/2010 07:48 PM, Alex Balashov wrote: >> Sounds like the church has strayed from its core competencies and >> invited the money-changers into the temple. >> >> Being the official asterisk-biz harbinger of God's wrath, I suggest an >> intensely commercial platform, for the meek shall inherit the Earth, >> not the 700 Club. Fight the power. >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can install and use Async AGI
Hi, How can I install Async AGI and use it in Asterisk 1.6 ?? Quyps -- _ -- 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] Asynchronous play music
Thanks! I've never tried either, But there rarely are documents or examples about this function. Do you know any good one, please point me some. Quyps On Tue, Mar 30, 2010 at 12:06 PM, Steve Edwards wrote: > On Tue, 30 Mar 2010, Pham Quy wrote: > > > Is there anyway to catch DTMF keypress while a music file is playing > > without stop the music? > > Have you tried externivr(). I've never used it, but it looks interesting. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
