[asterisk-users] Voice Mail beep / tone detection
Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing Software http://www.ictbroadcast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voice Mail beep / tone detection
Hi Carlos
Yes, I know about AMD & silence detection but my query relates to beep /
tone detection, AMD algorithm works on silence detection and it will not
able to detect Voice Mail beeps / tones
Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
Unified Communication Telemarketing Software
http://www.ictbroadcast.com
On Sun, Aug 5, 2012 at 10:58 PM, Carlos Rojas wrote:
> Hello
>
> You will need to do, something like
>
> [outbound]
>
> exten => s,1,NoCDR
>
> exten => s,n,AMD
>
> exten => s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
>
> exten => s,n(mach),WaitForSilence(2500)
>
> exten => s,n,Playback(message-when-machine)
>
> exten => s,n,Hangup
>
> exten => s,n(humn),WaitForSilence(500)
>
> exten => s,n,Playback(message-when-human)
>
> exten => s,n,Hangup
>
>
> On Sun, Aug 5, 2012 at 12:52 PM, tahir almas
> wrote:
> > Though asterisk support AMD which is based on silence detection but I did
> > not found support of tone / beep detection in asterisk to record a voice
> > message for answering machines after detecting tone
> >
> > Will appreciate any help in this regard
> >
> > Best Regards
> > Tahir Almas
> >
> > Managing Partner
> > ICT Innovations
> > http://www.ictinnovations.com
> > Leveraging open source in ICT
> >
> > Unified Communication Telemarketing Software
> > http://www.ictbroadcast.com
> >
> >
> > --
> > _
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> > New to Asterisk? Join us for a live introductory webinar every Thurs:
> >http://www.asterisk.org/hello
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Voice Mail beep / tone detection
I found some past work of Roger Schreiter contributed through asterisk
application "Mwanalyze" but not sure about its current status
Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
On Sun, Aug 5, 2012 at 11:37 PM, tahir almas wrote:
> Hi Carlos
>
> Yes, I know about AMD & silence detection but my query relates to beep /
> tone detection, AMD algorithm works on silence detection and it will not
> able to detect Voice Mail beeps / tones
>
> Regards
> *Tahir Almas*
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com
> Leveraging open source in ICT
>
> Unified Communication Telemarketing Software
> http://www.ictbroadcast.com
>
>
> On Sun, Aug 5, 2012 at 10:58 PM, Carlos Rojas wrote:
>
>> Hello
>>
>> You will need to do, something like
>>
>> [outbound]
>>
>> exten => s,1,NoCDR
>>
>> exten => s,n,AMD
>>
>> exten => s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
>>
>> exten => s,n(mach),WaitForSilence(2500)
>>
>> exten => s,n,Playback(message-when-machine)
>>
>> exten => s,n,Hangup
>>
>> exten => s,n(humn),WaitForSilence(500)
>>
>> exten => s,n,Playback(message-when-human)
>>
>> exten => s,n,Hangup
>>
>>
>> On Sun, Aug 5, 2012 at 12:52 PM, tahir almas
>> wrote:
>> > Though asterisk support AMD which is based on silence detection but I
>> did
>> > not found support of tone / beep detection in asterisk to record a
>> voice
>> > message for answering machines after detecting tone
>> >
>> > Will appreciate any help in this regard
>> >
>> > Best Regards
>> > Tahir Almas
>> >
>> > Managing Partner
>> > ICT Innovations
>> > http://www.ictinnovations.com
>> > Leveraging open source in ICT
>> >
>> > Unified Communication Telemarketing Software
>> > http://www.ictbroadcast.com
>> >
>> >
>> > --
>> > _
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> > New to Asterisk? Join us for a live introductory webinar every Thurs:
>> >http://www.asterisk.org/hello
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] best free fax solution with asterisk
I will recommend to give ICTFAX http://www.ictfax.org a chance , ICTFAX is based on spandsp and old version work with asterisk http://www.ictfax.org Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Mon, Aug 13, 2012 at 7:08 AM, Bryant Zimmerman wrote: > James is this inbound or outbound faxing that is running at 95%. We see > about 94% success on inbound faxes, but were not satisfied with that so we > started doing some research into the issue to find that the bulk of the > fails were actually voice calls, or robo dialers calling fax numbers. Once > we threw out those we get about 98% The last 2% include some calls that > might not have been faxes, but we were not able to eliminate all of them. > Out bound runs at about 98% success. > Thanks > > Bryant Zimmerman (ZK Tech Inc.) > 616-855-1030 Ext. 2003 > > > -- > *From*: "Steve Underwood" > *Sent*: Sunday, August 12, 2012 3:56 AM > *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > *Subject*: Re: [asterisk-users] best free fax solution with asterisk > > > On 08/12/2012 10:32 AM, James Sharp wrote: > > On 8/11/2012 8:05 AM, virendra bhati wrote: > >> Hi team, > >> > >> I want to configure fax with asterisk. there a lot of fax link i found > >> by google but not working perfectly. my setup as follow > >> > >> asterisk 10.x > >> centos 5.8 > >> > >> Want to used T.38 with SpanDSP... > >> > >> Please suggest me the best way. and how to test FoIP ? > > > > I use Asterisk 10.3.1, SpanDSP 0.0.6, and Ubuntu 11.10 connecting to > > Gafachi.com. It works with probably 95% success rate talking via T.38. > 95% is pretty bad. Do you know if the failures are mostly during the > initial negotiation, or somewhere in the actual FAX exchange? > > Steve > > > > -- > _ > -- 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] accept email and make phone call?
Already implemented Email to Fax in ICTFAX http://www.ictfax.org using both sendmail and drupal mail handler module , you need to modify Fax part with Voice call Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Fri, Sep 21, 2012 at 3:06 PM, Joseph Acquisto wrote: > >>> On 9/21/2012 at 4:00 AM, Jeremy Kister > wrote: > > On 9/20/2012 1:31 PM, Joseph Acquisto wrote: > >> Any ideas on how asterisk could accept an email (such as an email to > SMS or > > "num...@mybox.org" sort of thing) and make a phone > >> call to a specific number and make an announcement? > > > > that's actually what my jkSMS package does. > > > > i don't know if it'd be useful out of the box, depending on what you're > > trying to do. > > > > http://jeremy.kister.net/code/asterisk/jkSMS > > > > Jeremy Kister > > http://jeremy.kister.net./ > > > > I will take a look at it and certainly look at all the other suggestions > as well. > > Thanks to all for your response. > > joe a. > > > -- > _ > -- 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] Auto dialer scripts and software
ICTDialer http://www.ictdialer.org is free and open source dialer suitable for mentioned requirments Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Fri, May 24, 2013 at 1:26 AM, Ron Wheeler wrote: > One of the tricks used in Canada was to call the other party's > supporters and pretend that you are from their favorite party and piss them > off. > You can also give them misleading information such as a phony change in > voting location. > > It appears to work since the guys who did this won the election. > > If you hire a company outside your country to do this, you can make it > hard to detect and impossible to prosecute. > > Ron > > > On 23/05/2013 3:40 PM, cjwstudios wrote: > > As long as you're dialing a screened registered voter list and don't call > .gov or .edu, you're fine. > > > On Wed, May 22, 2013 at 5:54 AM, Don Kelly wrote: > >> Calls on behalf of political candidates are generally legal--even to >> people >> on the "do not call" lists. It doesn't seem to be possible to pass >> legislation preventing them. >> >> --Don >> >> >> >> >> -Original Message- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris >> Bagnall >> Sent: Wednesday, May 22, 2013 6:48 AM >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Auto dialer scripts and software >> >> On 22/5/13 10:54 am, A J Stiles wrote: >> > You do know that sort of thing is against the law -- or at least >> > requires a permit from the authorities -- in most civilised countries, >> right? >> >> And it's worth adding that even if it is legal in your country, you're >> almost guaranteed to offend/annoy your target audience. Recorded calls >> always do. >> >> Kind regards, >> >> Chris >> -- >> This email is made from 100% recycled electrons >> >> -- >> _ >> -- 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 > > > > -- > Ron Wheeler > President > Artifact Software Inc > email: rwhee...@artifact-software.com > skype: ronaldmwheeler > phone: 866-970-2435, ext 102 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send Fax from Asterisk
it support email to fax and fax to email, you are looking for *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Wed, Aug 15, 2012 at 11:35 PM, Ahmed Munir wrote: > Thanks for sharing the link. Actually I'm looking for a different approach > without installing/using third party i.e. a user sends an email to Asterisk > (which is also running mail service), as Asterisk receives the mail where > the mail contains attachment and subject contains destination number, > Asterisk will download the file and capture the number and later send fax > to destination number just like '.call' file. > > Does anyone worked on this scenario? If yes/no, please let me know at > earliest. > > > > > please check it. might be it will help >> >> http://ictfax.org/content/installation-guide >> >> On Tue, Aug 14, 2012 at 7:20 PM, Ahmed Munir > >wrote: >> >> > Hi, >> > >> > I would like to know, anyone who worked in Email to Fax scenario? If so >> > please share the idea for implementing it. >> > >> > As on other hand I configured Asterisk for inbound Fax which is working >> > good i.e. later forward the fax via email but don't know how can I >> > implement for outbound fax in this case. >> > >> > Please advice. >> > >> > -- >> > Regards, >> > >> > Ahmed Munir Chohan >> > >> > >> Thanks and regards >> >> Virendra Bhati >> +91-9718500594 >> Asterisk Developer >> E-mail-: virbh...@gmail.com >> Skype id:- virbhati2 >> New Delhi(India) >> [image: View my profile on >> LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755> >> > > -- > Regards, > > Ahmed Munir Chohan > > > > -- > _ > -- 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] Text to Speech Engine
Here is list of top multi language TTS engines 1. Acapela 2. Ivona 3. Loguendo 4. Cepstral. As per my information, they all work with open source Asterisk however please contact with their support for more information Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Wed, Jan 15, 2014 at 9:15 PM, Thorsten Göllner wrote: > Take a look at http://www.ispeech.org/ > > I implemented Speech-Recognition. The API is well documented and easy. > > Am 10.01.2014 21:16, schrieb Jai Rangi: > > Hello, > > Anyone know good quality text to speach engine for building IVRs for > asterisk. Open-source will be nice, but I wont mind paying for thing really > good. > > Regards, > -Jai > > > > -- > _ > -- 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] G729 Licensing Revisited - I'm Sorry!
> > 1) We do not perform any transcoding whatsoever. All recordings, and > voice mail are in G729, > and allow=g729 for all peers and in sip.conf. Is there anything else > we need to perform "g729 passthrough". More importantly are we still > liable? Given that most vendors support G729, why do some still > require the need to transcode? > As earlier referred following quote from their site "DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 patent holders for using their algorithm" You have to pay royalty fee for using their algorithm and it does not matter whether you are trans-coding or not however there is no restriction to pay their royalty fee under testing / evaluation environment. 2) If we decide that we require to purchase licenses, can we purchase > 23 licenses and continue to use the open source version? > I do't think there is any restriction to use open source version when you paid their roylity fee > Darryl Said > > The real question is: is there really any choice other than Digium for > the licence? Due to > > the dual licensing of the asterisk code, even if you could license the > codec elsewhere, you > might be violating Digium's OSS license when you > don't but their commercial asterisk > > license. > > This only applies to the commercial versions of the codec right? We > are still ok in respect to Digium's OSS license with the open source > should we decide to continue using that version? > Yes it is . I do't think there is confilit between GPL license and g.729 patent fee Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- 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] Which tool to automatically restart Asterisk ?
Why not to use Fail2ban
https://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
http://www.ictbroadcast.com
Leveraging open source in ICT
On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen
wrote:
> On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote:
> > Hi, Oliver.
> >
> > Maybe something like this (add this script to your crontab):
> >
> > 8<--
> >
> > #!/bin/bash
> > #
> > # File: asterisk-watchdog.sh
> > # Date: 2015.05.26
> > # Build:v1.0
> > # Brief:Secuencia para monitorizar procesos.
> > #
> > # ${PATH}: Variable de entorno con las rutas a los ejecutables.
> > PATH=/bin:/sbin:/usr/bin:/usr/sbin
> >
> > # ${DAEMON}: Demonio a monitorizar.
> > DAEMON="asterisk"
> >
> > # ${MSG}: Cuerpo del mensaje a enviar por mail.
> > MSG="$(date '+%F %T'): ${DAEMON} se ha caido!"
> >
> > pidof ${DAEMON} > /dev/null 2>&1
> >
> > [ $? -ne 0 ] && { echo ${MSG}; service ${DAEMON} start; }
> >
> > exit 0
>
> Both Debian 8 and Centos 7 have systemd. Systemd gives you this type of
> monitoring almost for free (see previous reply).
>
> Using cron is generally not a good idea here:
>
> 1. No way to stop Asterisk when you need it.
>
> 2. If Asterisk has failed, it may take up to a minute to restart it.
>
> --
>Tzafrir Cohen
> icq#16849755 jabber:tzafrir.co...@xorcom.com
> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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Re: [asterisk-users] Which tool to automatically restart Asterisk ?
Sorry , I forget it for another monitoring tool monit that we have
used in our production systems to restart asterisk in case of asterisk
crash or halt.
I have attached a monit configuration for your reference. it will work
almost in all cases
This configuration will check Asterisk for following
1. will check for Asterisk process.
2. will check Asterisk via AMI
3. will check Asterisk by sending a SIP request
You simply need to install monit and place attached file on your server as
/etc/monit.d/asterisk.conf and then restart monit service daemon
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictbroadcast.com
http://www.ictinnovations.com
Leveraging open source in ICT
On Thu, Feb 23, 2017 at 3:45 PM, Olivier wrote:
>
>
> 2017-02-21 14:09 GMT+01:00 Tahir Almas :
>
>> Why not to use Fail2ban https://www.voip-info.org/wiki
>> /view/Fail2Ban+%28with+iptables%29+And+Asterisk
>>
>> How would fail2ban detect that Asterisk needs to be restarted ?
>
>
>>
>>
>> *Tahir Almas*
>>
>> Managing Partner
>> ICT Innovations
>> http://www.ictinnovations.com
>> http://www.ictbroadcast.com
>> Leveraging open source in ICT
>>
>>
>>
>> On Tue, Feb 21, 2017 at 12:28 AM, Tzafrir Cohen > > wrote:
>>
>>> On Mon, Feb 20, 2017 at 11:36:24AM -0300, Victor Villarreal wrote:
>>> > Hi, Oliver.
>>> >
>>> > Maybe something like this (add this script to your crontab):
>>> >
>>> > 8<--
>>> >
>>> > #!/bin/bash
>>> > #
>>> > # File: asterisk-watchdog.sh
>>> > # Date: 2015.05.26
>>> > # Build:v1.0
>>> > # Brief:Secuencia para monitorizar procesos.
>>> > #
>>> > # ${PATH}: Variable de entorno con las rutas a los ejecutables.
>>> > PATH=/bin:/sbin:/usr/bin:/usr/sbin
>>> >
>>> > # ${DAEMON}: Demonio a monitorizar.
>>> > DAEMON="asterisk"
>>> >
>>> > # ${MSG}: Cuerpo del mensaje a enviar por mail.
>>> > MSG="$(date '+%F %T'): ${DAEMON} se ha caido!"
>>> >
>>> > pidof ${DAEMON} > /dev/null 2>&1
>>> >
>>> > [ $? -ne 0 ] && { echo ${MSG}; service ${DAEMON} start; }
>>> >
>>> > exit 0
>>>
>>> Both Debian 8 and Centos 7 have systemd. Systemd gives you this type of
>>> monitoring almost for free (see previous reply).
>>>
>>> Using cron is generally not a good idea here:
>>>
>>> 1. No way to stop Asterisk when you need it.
>>>
>>> 2. If Asterisk has failed, it may take up to a minute to restart it.
>>>
>>> --
>>>Tzafrir Cohen
>>> icq#16849755 jabber:tzafrir.co...@xorcom.com
>>> +972-50-7952406 mailto:tzafrir.co...@xorcom.com
>>> http://www.xorcom.com
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-
Re: [asterisk-users] Which tool to automatically restart Asterisk ?
Thanks for your suggestions *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com http://www.ictbroadcast.com On Mon, Feb 27, 2017 at 6:38 AM, Tzafrir Cohen wrote: > On Mon, Feb 27, 2017 at 06:00:30PM +0500, Tahir Almas wrote: > > Sorry , I forget it for another monitoring tool monit that we have > > used in our production systems to restart asterisk in case of > asterisk > > crash or halt. > > [snip] > > Some notes regarding the asterisk monit configuration: > > > check process asterisk with pidfile /var/run/asterisk/asterisk.pid > > group asterisk > > start program = "/bin/bash -c 'ulimit -n 16386 && > /etc/init.d/asterisk start'" > > If you use systemd, this ulimit will have no effect: when you restart a > service, it is restarted from a separate systemd context (cgroup) and > not directly under your own. > > It would generalyl be a good idea not to embed such settings in your > scripts and rather put them in a proper configuration file. What happens > in you happen to run '/etc/init.d/asterisk restart'? It seems that all's > well, until you're suddenly out of file descriptors. > > > stop program = "/etc/init.d/asterisk stop" > > if does not exist for 2 cycles then restart > > if failed port 5060 type udp protocol SIP > > and target "011@127.0.0.1" maxforward 10 > > for 2 cycles then restart > > if failed host 127.0.0.1 port 5038 with timeout 15 seconds for 2 > cycles then restart > > if 5 restarts within 5 cycles then timeout > > Nice. > > Also: what happens when you run 'core stop now' from within asterisk? > > -- >Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk. > org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Opensips Integration with Asterisk, opensips as load balancer and registrar and asterisk as media server
I like to share our experiment about integration of opensips and Asterisk with asterisk as media server and opensips as load balancer and registrar . following you will find details . improvements and suggestions will be welcomed http://www.ictinnovations.com/opensips-as-load-balancer-and-register-with-asterisk-servers-integeration Tahir Almas CEO ICT innovations http://www.ictinnovations.com http://www.ictbroadcast.com http://www.ictfax.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] salesforce opencti
We have successfully integrated Asterisk based autodialer ICTBroadcast with Salesforce CRM http://ictbroadcast.com/Integeration-of-ICTBRoadcast-auto-dialer-with-SalesForce-CRM-to-run-Voice-Broadcasting-campaigns-directly-from-Salesforce-CRM . Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Tue, Nov 13, 2012 at 6:57 PM Marek Cervenka wrote: > hello, > > do you have someone connector to salesforce? > http://wiki.developerforce.com/page/Open_CTI > > i need it for Mac OS X (the web/javascript way, not the old CTI Adapter > way) > > i'm using Asterisk 1.8 > > thanks > > -- > --- > Marek Cervenka > === > > > > -- > _ > -- 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 -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ICTBroadcast Enterprise Edition , Asterisk based telemarketing platform , Limited free licenses available
Pleased to offer absolutely free licenses of ICTBroadcast Enterprise Edition 2 channels, 5 channels , 10 channels and 50 channels licenses Register now https://service.ictinnovations.com/cart.php?gid=1 http://www.ictbroadcast.com Regars *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New features released in ICTBroadcast
Following new features are now supported by asterisk based telemarketing software Auto subscription / registration after call recipient press a key in voice broadcasting https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer There will be restriction to call a number in off time accordingly to timezone of destination number automatically https://www.ictbroadcast.com/Time-Zone-based-restrictions-on-telemarketing-campaigns-ICTBroadcast-autodialer-scheduling Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New features released in ICTBroadcast
ok , noted regards On Wed, 19 Dec 2018, 18:26 Social Boh Please STOP send this kind of messages. Use only > asterisk-...@lists.digium.com list > > Thank you > > --- > I'm SoCIaL, MayBe > > El 19/12/2018 a las 06:00, Tahir Almas escribió: > > Following new features are now supported by asterisk based > telemarketing software > > Auto subscription / registration after call recipient press a key in voice > broadcasting > > > https://www.ictbroadcast.com/Subscription-Campaign-to-automatically-register-customers-at-website-with-Voice-broadcasting-Autodialer > > There will be restriction to call a number in off time accordingly to > timezone of destination number automatically > > > https://www.ictbroadcast.com/Time-Zone-based-restrictions-on-telemarketing-campaigns-ICTBroadcast-autodialer-scheduling > > Regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ICTAgent Ver 0.6 released, A chrome extension based web phone
Just released a new version of ICTAgent with voice calling and Fax support using ICTCore as backend communications framework also tested with any other gateway. Here is project link https://github.com/ictinnovations/ictagent regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ICTBroadcast Version 4.2 released
Pleased to announce the release of asterisk based auto dialer and call center software solution https://www.ictbroadcast.com/ICTBroadcast-version-4.2-released-contact-center-software ICTBroadcast is a complete inbound and outbound call center software solution with support of unified communications, predictive dialling, auto dialling, power dialling, purview and progressive dialling as well as manual dialling support. https://www.ictbroadcast.com/ICTBroadcast-version-4.2-released-contact-center-software Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICTBroadcast Version 4.2 released
Sorry , posted here being it as asterisk based project, it would not happen again regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Sat, Aug 14, 2021 at 4:29 PM Social Boh wrote: > He do this always y still have access to this list. > > Very bad behaviour. > > --- > I'm SoCIaL, MayBe > > El 14/08/2021 a las 4:50 a. m., Antony Stone escribió: > > On Saturday 14 August 2021 at 11:43:55, Tahir Almas wrote: > > > >> Pleased to announce the release of asterisk based auto dialer and call > >> center software solution > > Please note that you have posted this advertisement to the mailing list > > "Asterisk Users - Non-Commerical Discussion". > > > > > > Regards, > > > > > > Antony. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICTBroadcast Version 4.2 released
The first post was about an open source github project with no commercial benefit at all and this one , I admitted my mistake and apologized. what is the lie here regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Sat, Aug 14, 2021 at 4:48 PM Social Boh wrote: > BLA BLA BLA... liar > > --- > I'm SoCIaL, MayBe > > El 14/08/2021 a las 6:38 a. m., Tahir Almas escribió: > > Sorry , posted here being it as asterisk based project, it would not > happen again > > regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > On Sat, Aug 14, 2021 at 4:29 PM Social Boh wrote: > >> He do this always y still have access to this list. >> >> Very bad behaviour. >> >> --- >> I'm SoCIaL, MayBe >> >> El 14/08/2021 a las 4:50 a. m., Antony Stone escribió: >> > On Saturday 14 August 2021 at 11:43:55, Tahir Almas wrote: >> > >> >> Pleased to announce the release of asterisk based auto dialer and call >> >> center software solution >> > Please note that you have posted this advertisement to the mailing list >> > "Asterisk Users - Non-Commerical Discussion". >> > >> > >> > Regards, >> > >> > >> > Antony. >> > >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ICTBroadcast Version 4.2 released
Check the post about ICTBroadcast 4.1 release, only posted on biz list and it has not been posted here regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Sat, Aug 14, 2021 at 5:03 PM Tahir Almas wrote: > The first post was about an open source github project with no commercial > benefit at all and this one , I admitted my mistake and apologized. what > is the lie here > > regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > > > On Sat, Aug 14, 2021 at 4:48 PM Social Boh wrote: > >> BLA BLA BLA... liar >> >> --- >> I'm SoCIaL, MayBe >> >> El 14/08/2021 a las 6:38 a. m., Tahir Almas escribió: >> >> Sorry , posted here being it as asterisk based project, it would not >> happen again >> >> regards >> *Tahir Almas* >> >> Managing Partner >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> >> On Sat, Aug 14, 2021 at 4:29 PM Social Boh wrote: >> >>> He do this always y still have access to this list. >>> >>> Very bad behaviour. >>> >>> --- >>> I'm SoCIaL, MayBe >>> >>> El 14/08/2021 a las 4:50 a. m., Antony Stone escribió: >>> > On Saturday 14 August 2021 at 11:43:55, Tahir Almas wrote: >>> > >>> >> Pleased to announce the release of asterisk based auto dialer and call >>> >> center software solution >>> > Please note that you have posted this advertisement to the mailing list >>> > "Asterisk Users - Non-Commerical Discussion". >>> > >>> > >>> > Regards, >>> > >>> > >>> > Antony. >>> > >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> Check out the new Asterisk community forum at: >>> https://community.asterisk.org/ >>> >>> New to Asterisk? Start here: >>> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk RPM with dubuginfo/core-dump/backtrace
I need help that how to compile asterisk version 16.x with backtrace flag , debug or coredump flags. I tried many times but it has not been compiled successfully. Would be thankful if someone could help us how to use these flags in compilation that work with asterisk 16.x successfully. i am using source rpm from this repo https://ast.tucny.com/repo/asterisk-16/el7/ Thanks *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Community Forum
Thanks for invitation to asterisk community forum regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Tue, Jul 5, 2022 at 2:31 PM Joshua C. Colp wrote: > Kia ora, > > Kind of a random email here but thought I'd remind everyone of the > community forums at https://community.asterisk.org/ which see more > activity than the mailing list. If you've got questions/issues, you may > find an answer there instead of the mailing list. > > Cheers, > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cps limit of asterisk
What is maximum cps limit of a good asterisk server (single node ) ? regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cps limit of asterisk
What cps and maximum concurrent calls on following hardware configuration with freeswitch possible ? I need approximate speculation not exact or real time results Intel E-2286 G 2 x 960 SSD , 6 cores / 12 threads @4 ghz , 128 GB Memory and no limitation on internet bandwidth *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Thu, Nov 24, 2022 at 12:49 AM Joshua C. Colp wrote: > The only answer is the one you benchmark yourself for your environment, > deployment, and usage. > > On Wed, Nov 23, 2022 at 3:02 PM Tahir Almas Dhesi < > ta...@ictinnovations.com> wrote: > >> What is maximum cps limit of a good asterisk server (single node ) ? >> >> regards >> *Tahir Almas* >> >> Managing Partner >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cps limit of asterisk
Sorry for typing error , I mean asterisk regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Thu, Nov 24, 2022 at 3:22 PM Jon Bonilla (Manwe) wrote: > El Thu, 24 Nov 2022 15:08:12 +0500 > Tahir Almas Dhesi escribió: > > > What cps and maximum concurrent calls on following hardware configuration > > with freeswitch possible ? > > > > I need approximate speculation not exact or real time results > > > > Intel E-2286 G 2 x 960 SSD , 6 cores / 12 threads @4 ghz , 128 GB > > Memory and no limitation on internet bandwidth > > > > > > This is not a freeswitch list. This is the asterisk list. > > Why don't you test yourself and share your results with us? > > thank you > > > > -- > PekePBX, the multitenant PBX solution > https://pekepbx.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cps limit of asterisk
G.711 voice calls origination , no transcoding , sip + rtp regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Fri, Nov 25, 2022 at 6:24 PM Tahir Almas Dhesi wrote: > Sorry for typing error , I mean asterisk > > regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > > On Thu, Nov 24, 2022 at 3:22 PM Jon Bonilla (Manwe) > wrote: > >> El Thu, 24 Nov 2022 15:08:12 +0500 >> Tahir Almas Dhesi escribió: >> >> > What cps and maximum concurrent calls on following hardware >> configuration >> > with freeswitch possible ? >> > >> > I need approximate speculation not exact or real time results >> > >> > Intel E-2286 G 2 x 960 SSD , 6 cores / 12 threads @4 ghz , 128 GB >> > Memory and no limitation on internet bandwidth >> > >> > >> >> This is not a freeswitch list. This is the asterisk list. >> >> Why don't you test yourself and share your results with us? >> >> thank you >> >> >> >> -- >> PekePBX, the multitenant PBX solution >> https://pekepbx.com >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? Start here: >> https://wiki.asterisk.org/wiki/display/AST/Getting+Started >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G.729 Annex B or AB support in Asterisk
Does Asterisk support G.729 annex B or AB codec ? regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Memory leak
A memory leak can be challenging to diagnose, but there are several steps you can take to identify the cause within your Asterisk/FreePBX setup. Here's a general approach to help you find the module or component responsible for the memory leak: Monitor System Resources Identify Suspected Modules Enable Debugging and logging Stress Testing and Isolating Modules Update and Patch Modules Reach Out for support 1. - Tahir Almas Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Sun, Jul 9, 2023 at 2:48 AM wrote: > On 7/8/2023 5:32 PM, Federico wrote: > > > > I am using Asterisk 16.30 inside Freepbx, with commercial modules, > > purchased from Sangoma and Symphony. After a few hours my memory usage > > reaches 900 GB, no kidding, in a box with 1 TB of RAM. The question > > is: how can I determine what is causing the memory leak? Can somebody > > send me instructions to find out what module is killing my box? > > FreePBX is 100% updated. > > > > There is some documentation on the wiki, here: > > https://docs.asterisk.org/latest/Development/Debugging/Memory-Leak-Debugging/ > > The easiest way is recompile with MALLOC_DEBUG. You can then use "memory > show summary" to narrow it down to a module, and "memory show > allocations" to narrow it down to specific leaks. > > NA > > > Before I contact a vendor I need understand what module is responsible > > for this. > > > > Alternatively, I would hire a consultant to login remotely via SSH and > > do the work. I am an businessman, not an engineer. > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Recommended sip providers
Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended sip providers
Only outbound to USA so no DID Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Mon, Nov 20, 2023 at 4:18 PM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > > > Interested to know good wholesale SIP providers for 15k concurrent calls > > You might want to specify a bit more detail, such as: > > - which country are you located in > - do you require inbound DDIs (if so, in which region/s)? > - which countries' Caller ID/s do you need to present? > > Antony. > > -- > These clients are often infected by viruses or other malware and need to > be > fixed. If not, the user at that client needs to be fixed... > > - Henrik Nordstrom, on Squid users' mailing list > >Please reply to the > list; > please *don't* CC > me. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
