On Friday 01 Sep 2017, Tim Turpin wrote:
> Is there a way that I can modify the source code for the voicemail
> application? I need to change some of the options in the user’s interface
> to make it work like an existing system that I’m replacing.
$ vi /usr/src/asterisk-*/apps/app_voicemail.c
--
On Friday 30 Jun 2017, Jonathan H wrote:
> What's the simplest, easiest quickest least-code way of firing off an AGI
> with some variable, and then returning to the dialplan?
You have to use the "fork" command. This starts a copy of the process with
all the same internal state including variable
On Monday 26 Jun 2017, Harel wrote:
> Hello List,
> I'm working on an autodialer project.
> At the moment I use the Originate application then I "throw" it to an
> extension where I Dial() the other party and then both legs are bridged.
> The problem is that the Dial() will only run after the Origi
On Friday 16 Jun 2017, Christopher van de Sande wrote:
> So does anyone here think the traditional telephone company will go
> extinct, and voice communication will take place via email like (or
> equal to) sip uri's?
Hardly!
The job of the "traditional telephone company" has always been to conne
On Thursday 15 Jun 2017, Tim S wrote:
> Whatever has been done, if anything, isn't working effectively. At this
> point I'd like to see some response from the mailing list admin about any
> root-cause efforts, AFAIC this is starting to smear the Digium/Asterisk
> brand's ability to handle IT relat
On Thursday 08 Jun 2017, Olivier wrote:
> Hello,
>
> I'm building a new Asterisk system from source on Debian Stretch.
> My building script fails as package libmyodbc is currently missing from
> Debian Stretch repo.
>
> Is there a work around this without leaving MySQL/MariaDB galaxy ?
This is w
On Wednesday 10 May 2017, Steve Edwards wrote:
> On Wed, 10 May 2017, J Montoya or A J Stiles wrote:
> > Presumably your staff carry mobile phones. What about an app that gets
> > the ID of the cell tower to which it is connected, and passes it and the
> > SIM number in a HTT
On Wednesday 10 May 2017, Steve Edwards wrote:
> I have a 'time and attendance' application. Think janitorial or security
> kind of thing where an employee goes from location to location.
>
> They're supposed to 'clock in' when they get to a site using a phone at
> that site to prove they're there
On Wednesday 10 May 2017, andre castro wrote:
> Indeed. apt-get install libjack-dev libresample-dev were not installed.
> libjack-dev libresample-dev , so I installed.
> In the installation of libresample-dev apt-get selected
> 'libresample1-dev' instead of 'libresample-dev'. Not sure if that is a
On Wednesday 10 May 2017, andre castro wrote:
> Hello,
> I am new to Asterisk, so please bear with me.
> I have made a success installation from source of Asterisk 14.4.0 on
> Debian Jessie (8.7). And I am running the Asterisk server, with several
> extensions and dialplans, all working well.
>
>
On Monday 08 May 2017, Frank Vanoni wrote:
> By dialing 4000 or 4001, the dialplan is modified and reloaded
> accordingly.
>
> Is there a better solution?
That's an . interesting . way of doing things!
We would be thinking in terms of using a GLOBAL variable, or an ASTDB entry,
to indi
On Thursday 20 Apr 2017, Fabio Moretti wrote:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound calls without answering them, doing
> something like this:
>
> analog line 1 -+-- asterisk
>
>\_
On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> Yes and [using something like "1571"] works just fine for us. The problem
> is that we are trying
> to deal with the situation where someone calls themselves from another
> phone (internal or external) to pick up their messages. In every other
> ca
On Wednesday 19 Apr 2017, D'Arcy Cain wrote:
> On 2017-04-19 02:39 AM, Pete Mundy wrote:
> > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail
> >
> > show users' I can't see why the vm_authenticate function is
> > failing to read the username :(
>
> I can answer that one. It's
On Monday 17 Apr 2017, Speed Boy wrote:
> Hi all, I'm new to VoIP, now we have a project that needs a
> PBX with client APPs.
> In our team we have argument for choosing PBX. By so far, we
> have following candidates:
>
> A: Open source
>
> 1) Asterisk PBX (http://www.asterisk.org) (with
On Thursday 06 Apr 2017, Atux Atux wrote:
> hi. i would like to be able to reboot the system from my extension. is that
> possible? if yes, how?
It's possible, with something this in extensions.conf;
exten => 99,1,NoOp(Restarting server now)
exten => 99,n,System(shutdown -r now)
Then dia
On Thursday 30 Mar 2017, Ikka Tirtawidjaja wrote:
> Dear all,
>
> I have PBX with asterisk 13.x
>
> a couple of IPPhone that connect to that asterisk PBX send an alphanumeric
> dialed phone number.
>
> for example, in my CDR table, field DST, it show dialed phone number like
> - 0C81318304632C
On Saturday 18 Mar 2017, Jonathan H wrote:
> Hi, thanks - that looks really good!
>
> I was about to embark on some non-visual stuff using Ragic, but this
> looks great.
>
> Is there a binary anywhere, or any instructions to compile? I've never
> compiled C# code before, and although a quick goog
On Saturday 25 Feb 2017, Антон Сацкий wrote:
> Thanks U Richard
> i know about this solution
> but the main question why "${} substitution containing
> the SHELL is evaluated before anything else"
For the same reason why you do raising to powers before multiplications and
divisions, and all those
On Thursday 16 Feb 2017, Max Grobecker wrote:
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client supports TLS, SRTP and ZRTP:
> http://phonerlite.de/features_en.htm
>
> Yes, the GUI is not that much user friendly as Zoiper is - but a
On Thursday 16 Feb 2017, Olivier wrote:
> Hello,
>
> While googling, I've just discovered Recqual.
> If I'm not mistaken, project's sourceforge site [2] does not host any
> source or binary.
You need to follow the "code" link, copy the line that starts with
"svn checkout ..."
and then just paste
On Thursday 02 Feb 2017, Amelye Chatila wrote:
> Hi,
> I need to make calls to a list of numbers one at a time and once the user
> pick the phone connects to an IVR where I can get few data, after a call
> finishes the 2nd number get called and so forth.
>
> I'm familiar with Asterisk/Elastix but
On Tuesday 24 Jan 2017, Zakir Mahomedy wrote:
> Hi I am trying to setup DDI for one of our servers
> Our Provider has given us one DDI for use for eg 080011.
> On my main server A, I use an IAX trunk to connect to Client Server
> B.When calls come in from the outside world on main server A fo
On Thursday 12 Jan 2017, Telium Technical Support wrote:
> This was asked many years ago but I thought I would check to see if things
> have changed. Is it possible to take over a call in progress - using a
> replacement Asterisk server?
>
> In other words, if 2 user agents are connected through
* THIS IS NOT WHERE YOUR REPLY BELONGS *
On Tuesday 10 Jan 2017, Olivier wrote:
> Historically, I didn't use "install_prereq" but I also used it yesterday.
>
> As make fails with "[LD] libasteriskpj.o -> libasteriskpj.so.2" which is
> the first of its "kind", I still wonder
> if issue c
On Tuesday 10 Jan 2017, Olivier wrote:
> Hello,
>
> I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes.
>
> I followed this:
> cd /usr/src
> wget ... asterisk-13.13.1.tar.gz
> tar zxf asterisk-13.13.1.tar.gz
> cd asterisk-13.13.1
> ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/us
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Monday 12 Dec 2016, christopher kamutumwa wrote:
> Hello support,
>
> Am not winning need your help. ive tried putting a different version of
> asterisk on centos 7 and here are below results, after make config;
>
> [root@localhost asterisk-14.2.1]
On Saturday 10 Dec 2016, christopher kamutumwa wrote:
> ive installed asterisk but below is what am getting proces gets
> killed.please help
>
Make sure you have libncurses5 and its development files installed, otherwise
this can cause crashes.
Also, how much RAM is in your box? Check Asterisk
On Wednesday 30 Nov 2016, Emiliano Vazquez wrote:
> i'm using gammu[1] with a 3g dongle and my own chip with my preffer
> provider. It can send over 700 every our and receive to. I don't know if
> you need asterisk and sms in the same way but with this tool you can make
> everything. It has python
On Wednesday 30 Nov 2016, Michele Pinassi wrote:
>[stuff deleted]
> but on a call directed to, es. FAX_3700 i got:
>
> [Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
> handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
> 'FAX_3700' rejected because extension not fo
Many years ago, I used to have an AGI script that fired on an incoming call,
did some database lookups and ended up raising a notification on the screen of
the person whose phone was ringing, with the details looked up from the
incoming caller ID.
All that fell by the wayside when Debian Squeez
On Thursday 27 Oct 2016, KyD wrote:
> Hi!
>
> I need to make a dialplan by DID.
>
> where it gets the asterisk values did? from sip headers or ... ?
>
> Thanks!
It will all be taken care of for you, so you don't have to do anything special
for calls to a direct inbound number. When a call com
On Wednesday 26 Oct 2016, KyD wrote:
> Hi,
>
> My sip provider gave me 2 numbers for the incoming call via pstn.
>
> nro1 = 12341234
> nro2 = 45674567
>
> I have a dialplan for each.
> if i put this on my dialplan:
>
> exten => s,1,Dial(SIP/1001)
> exten => Hangup()
>
> Works!
>
> But if i pu
On Friday 30 Sep 2016, aaberga/gmail wrote:
> Hi,
>
> after a long pause (Asterisk 1.8 times), I have started again playing with
> VOIP. A lot has changed since last time I did setup an Asterisk system!
>
> So I am asking for some help.
[stuff deleted]
> [2102]
> type=endpoint
> context=internal
On Thursday 15 Sep 2016, tux john wrote:
> hi. i am running asterisk 11 and i am using astdb to store all my contacts
> and their numbers. so everytime they call me, i can see their name on the
> screen of the phone. i am making use of the following to retrieve the name
> from the astdb exten =>
>
On Tuesday 30 Aug 2016, D'Arcy J.M. Cain wrote:
> I have an extension that looks like this:
>
> exten => 55,1,Verbose(Door buzzer calling)
> same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
>
> The idea is that any of the three users can answer the phone to let
> someone in. The problem i
On Thursday 04 Aug 2016, Nabeel wrote:
> On 30 July 2016 at 19:32, D'Arcy J.M. Cain wrote:
> > Not playing the prompt changes nothing. If someone presses '*' while
> > listening to your answer message then they are in your mailbox. You
> > better have a password that they need to enter to contin
On Tuesday 26 Jul 2016, Jerry Geis wrote:
> It seems I am not getting any digits coming over a SIP trunk.
>
> How can I match "anything" or "nothing" and start my extension.
>
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
>
> but if 55 does not come across and it appears to b
On Wednesday 20 Jul 2016, Yves biganiro wrote:
> Hi all
>
> Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial
> system ,
Is this some kind of one-stop, pre-prepared distribution with Linux, Asterisk,
DAHDI, a web server and some custom scripts, that all installs from o
On Friday 15 Jul 2016, Joaquin Alzola wrote:
> Hi Guys
>
> I am asking too many questions because we would like to use Asterisk first
> as a proof of Concept and check from there were it goes.
>
> - Does the Voicemail have the option of SMS notification on new drop
> messages (we have an SMSC so
On Friday 15 Jul 2016, Joaquin Alzola wrote:
> Hi Madushan
>
> Maybe I was not clear …. After SIP negotiation and SDP set up on the
> VoiceMail Server ….
>
> Is there a file to specify a MGw (the machine that deliver RTP packages to
> end user)?
No. The VoiceMail server takes care of all that
On Thursday 14 Jul 2016, Joshua Colp wrote:
> Carlos Chavez wrote:
> > Until Asterisk 11 I could use sip.conf to set defaults for all phones
> > (language, dtmf, vmexten, etc) and just leave many fields in the
> > database as NULL. What would be the proper way to do this for Asterisk
> > 13 and PJS
On Wednesday 06 Jul 2016, Michael Jepson wrote:
> Adding live_dangerously did the trick. Thanks! But how dangerous is
> Asterisk living now ?
I must admit, still using an ancient Asterisk version, I didn't know about
live_dangerously. But it sort of makes sense.
It is somewhat dangerous to ha
On Wednesday 06 Jul 2016, John Novack wrote:
> AstLinux can be remotely managed with the GUI,
> which unlike other Asterisk GUI's the conf files are not modified by the
> GUI and can be edited "by the book" AstLinux will NOT work with a Pi
> though. It is not for the ARM processor.
What stops it f
On Monday 04 Jul 2016, Michael Jepson wrote:
> Hi all,
>
> I am getting the following error when starting asterisk:
> pbx_functions.c: Function SHELL not registered
>
> Some of my conf files use a SHELL command, which used to work with an older
> version of asterisk, but now with version 13.9.1 I
On Monday 06 Jun 2016, Markus wrote:
> Hi AJ,
> Am 06.06.2016 um 10:14 schrieb A J Stiles:
> > But why not call an AGI script, have this check the caller ID against a
> > MySQL database and return a status -- blocked or not -- in a variable?
> > Then you can manage individ
On Saturday 04 Jun 2016, Markus wrote:
> Hi list,
>
> n00b question, but I can't figure it out:
>
> [callthrough]
> exten => _+X.,1,NoOp(nothing here)
> #include "blockedall.conf"
> exten => _+X.,n(hangup),Hangup
> exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" =
> "anonymous"]?nocli:cli)
On Saturday 14 May 2016, Stefan Becker wrote:
> Greetings,
>
> asterisk list and community,
>
> I have a problem in how our telefon switch (Siemens HiCOM)
> "talks" with my new configured Asterisk server (V.11.18.0)
>
> without my Asterisks server in the middle
>
> <--> Siemens HiCOM <-ISD
On Monday 09 May 2016, Jonathan H wrote:
> . {stuff deleted} .
> [streamdemo]
> exten => s,1,Answer
> exten => s,2,BackGround(menu)
> exten => s,3,WaitExten
> exten => s,4,Goto(s,2)
> exten =>
> _[2,3,4,5],1,Dial(Local/${EXTEN}@play-radio,,G(play-radio^${EXTEN}^2))
> exten => _[2,3,4,5],2,G
On Friday 06 May 2016, Alok Srivastava wrote:
> Dear List
> wanna configure click2call in such a manner that my asterisk box call two
> mobile numbers and connect both numbers to talk. I have configured voip
> gateway, my internal and external calls are working fine.
> please help ,
You ought to b
On Wednesday 04 May 2016, Mamadou NGOM wrote:
> Hello everybody,
> When I call my extension the agi script don't work well. when I look at
> the cli, that is what I have:
> [stuff deleted]
> AGI Tx >> agi_arg_1: 56
> AGI Tx >>
> AGI Rx << SET VARIABLE ** 2
> AGI Tx >> 510 Invalid or unknown
*** THIS IS NOT WHERE YOUR REPLY BELONGS ***
On Friday 29 Apr 2016, Mamadou NGOM wrote:
> Hello,
> I have not resolved my problem.I renamed my dahdi file "mv dahdi.bash
> dahdi " in the directory /etc/init.d, but it doesn'nt work yet. the same
> error after the command /etc/init.d/dahdi start
On Thursday 28 Apr 2016, Robin Kipp wrote:
> Hi all,
>
> sorry if the subject is a bit confusing, but I just couldn’t think of a
> good way of better describing the situation…
>
> Basically, I travel a lot and have several SIM cards for my phone from
> local carriers. What I’d like to do now is t
On Thursday 28 Apr 2016, Mamadou NGOM wrote:
> Hello,
> it doesn't work my dahdi yet .for information, i use debian 8 .
> I put the file dahdi.bash in /etc/init.d and I gave it the permission 755
> but i have the same error: bash: /etc/init.d/dahdi: No such file or
> directory
You need to name
On Tuesday 26 Apr 2016, Mamadou NGOM wrote:
> Hello,
>
> Having installed DAHDI to be able to use the meetme() application , when I
> start the dahdi service it generates me the following error: -bash:
> /etc/init.d/dahdi: No such file or directory
> I need help please.
You are using a distributi
On Wednesday 13 Apr 2016, Jeremy Kister wrote:
> On 4/13/16 11:57 AM, A J Stiles wrote:
> > You could try
> > *CLI> dialplan show
>
> Between my older backup and dialplan show, I guess that's my best shot.
>
> Thanks :D
I'll have a go this lunchtime at kn
On Wednesday 13 Apr 2016, Jeremy Kister wrote:
> with the slip of a finger, i destroyed by extensions.conf (grep -i >
> extensions.conf)
>
> I have a backup that is dozens of hours of code old.
>
> is there a way i can use the asterisk cli (or some other asterisky
> method) to recreate that exten
On Tuesday 05 Apr 2016, Mamadou NGOM wrote:
> Hello,
> I am doing a configuration for connecting my server asterisk to a SIP
> provider. I ask if somebody can give me a basic code or a link to begin
> well; Thanks
Rule One: Start your own topics -- don't jump in on someone else's, unless
it
On Thursday 31 Mar 2016, Mamadou NGOM wrote:
> Hello !
> I ask if it is necessary to install DAHDI and LIBPRI if we want to connect
> our asterisk to an operator SIP (trunk SIP). Someone for helping me.
> thanks !!!
No.
DAHDI is a library for hardware interfaces to POTS, ISDN and mobile lines.
On Wednesday 30 Mar 2016, Vitor Mazuco wrote:
> Humm thanks for your reply,
>
> Do you know whats is step for I can transform this card link a fax modem?
Start with the specification document for the modulation scheme you want to
implement, and the DAHDI Source Code for the card you want to use.
On Wednesday 30 Mar 2016, Vitor Mazuco wrote:
> Hi!
>
> Is possible to use X100p TDM400P, Tdm410p, Tdm400, A400p, Ax400p or
> any others digium card FXO for use Fax modem?
Yes, in theory it is entirely possible to use an FXO card driven by software
as a modem (and indeed, this is exactly what W
On Tuesday 29 Mar 2016, Rizwan H Qureshi wrote:
> Hi Everyone,
> I need to develop a service which tells me whether a given phone number is
> in service and is valid or not. It can be international number. This is
> basically to clean the list of leads we have. Is there any service which
> can give
On Thursday 24 Mar 2016, Tony Mountifield wrote:
> In article <201603241343.24128.asterisk_l...@earthshod.co.uk>,
> A J Stiles wrote:
> > When placing a call over a SIP channel to a mobile phone, if the phone is
> > engaged, it does not return a BUSY status straig
On Thursday 24 Mar 2016, Mamadou NGOM wrote:
> Hello,
> I am asking if it is possible to left from a version to another one of
> asterisk without reinstalling it. I would like to say for example is
> there a linux command which allows us to left version 12 to 13. Passage
> from a version to an ot
I'm not sure if this is an Asterisk thing, a handset thing or a telco thing,
so please be gentle with me if this is not the right place to ask .
When placing a call over a SIP channel to a mobile phone, if the phone is
engaged, it does not return a BUSY status straightaway. Rather, I get a
On Wednesday 23 Mar 2016, Olivier wrote:
> I'm thinking about something to delegate provisionning to end users:
> a new employee joins the company, the system I'm after let him enter his
> own name himself, once for all.
This is generally good, because it means less work for you :) Just be
care
On Wednesday 23 Mar 2016, Olivier wrote:
> Hello,
>
> I'm wonddering if it is possible, with Asterisk and any third party module
> or service, to build the following feature:
>
> - caller dials a given extension dedicated to a given language (german,
> english, ...)
> - Asterisk plays a welcome a
On Monday 21 Mar 2016, somsad khan wrote:
> Hello guys,
>
> I need some help.
>
> I have a client coming who wants to assign 5 different numbers to one
> virtual employee SIP phone at his desk or softphone (Zoiper).
>
> which I can assign for the incoming or outgoing both.
>
> but the problem i
On Thursday 10 Mar 2016, Joshua Colp wrote:
> I wrote:
> > I can't seem to find a definitive answer on this, and I really don't want
> > to risk breaking a production server to find out; so I am going to try
> > asking this here, and maybe anyone else in the same situation searching
> > the archive
I can't seem to find a definitive answer on this, and I really don't want to
risk breaking a production server to find out; so I am going to try asking this
here, and maybe anyone else in the same situation searching the archives
sometime in future will find the answer I get.
Can you use variab
On Wednesday 02 Mar 2016, Ryan, Travis wrote:
> I am wondering what the best solution is for initiating a call from Outlook
> Contacts. I imagine something that would start the call from the outlook
> card (or similar) and then dial the user's extension and the contact's
> phone number and place th
On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote:
> OK. Let me ask this. Is anything else necessary, except choosing TCP as the
> preferred protocol on the client, to make TCP w Asterisk work? At the
> moment, I have only changed one line in pjsip.conf from my working UDP
> setup:
>
> [transport-
On Wednesday 17 Feb 2016, imperium broadcast wrote:
> I kinda have it working with chan_sip.
>
> Dial(SIP/+${EXTEN}\;phone-context=+44@10.10.10.10;user=phone)
> But it doesn't include the user=phone at the end when dialling out.
>
> "To: ".
>
> even adding
> usereqphone=yes
> to the sip.conf doe
On Wednesday 17 Feb 2016, Goke Aruna wrote:
> Hello all,
> Can someone recommend what hardware to use for a 1000 analogue line
> capacity asterisk PABX?
>
> Regards
A PCI express card with four primary rate ISDN ports, each linked up to a
channel bank, will give you 120 analogue lines. So you w
On Thursday 04 Feb 2016, Marek Červenka wrote:
> hi,
>
> is there way to get SQL schema for Asterisk 13.7.0 without alembic?
> thanks
Assuming you already have Asterisk up and running, you can just use
$ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ...
will print (on STDOUT, so you can ju
On Wednesday 27 Jan 2016, James Cloos wrote:
> I gave up switching my edge asterisk to pjsip at least twice because I
> couldn't figure out how to configure it properly for a dynamic ip. And
> I sent a note to one of the lists at least on the 2nd attempt.
>
> That install doesn't need nat for si
On Wednesday 27 Jan 2016, Marek Červenka wrote:
> Dne 27.1.2016 v 13:14 A J Stiles napsal(a):
> > On Wednesday 27 Jan 2016, Marek Červenka wrote:
> >> hi,
> >>
> >> i have strange problem with asterisk 13 mixmonitor, recording to wav
> >> (centos6)
On Wednesday 27 Jan 2016, Marek Červenka wrote:
> hi,
>
> i have strange problem with asterisk 13 mixmonitor, recording to wav
> (centos6)
> when the system is under load, there are sometimes missing syllable
>
> there arent BIG spikes on cpus
> recordings are to ramdisk (/dev/shm)
>
> any hints
On Monday 25 Jan 2016, waqas.mehmood90 wrote:
> I am working on asterisk ivr .i am facing problrm in crontab.when i run
> example it give bash 5:command not found then i check and found that no
> crontab for root user kindly guide me please
Hello, is that the vet? One of my animals is poorly. Wh
On Thursday 21 Jan 2016, Jerry Geis wrote:
> >Not really. Very little info to go on so far. You need to give us
> >more detail of what you are doing with AGI and AMI.
>
> Sorry - let me try again...
>
>
> I am basically doing the following:
> 1) calling a phone SIP/401 upon answer run an AGI for
On Thursday 21 Jan 2016, Jerry Geis wrote:
> I am using the AMI interface to start calls.
>
> At one point I have a 10 second delay "Wait(10)" in the dialplan...
> During this time it "seems" that if I then connect with the manager
> interface
> and place a call that nothing happens till the 10 se
On Wednesday 20 Jan 2016, bilal ghayyad wrote:
> Hello List;
> I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and
> I am getting the following debug, can someone advise me about the
> solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE
> . [stuff deleted] .
On Monday 18 Jan 2016, Matthew Murphy wrote:
> Hi everyone,
>
> I am getting a segmentation fault (seems to occur randomly) using Asterisk
> 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that
> libmysqlclient is complaining about when doing a query in
> ps_endpoint_id_ips. We are usi
On Wednesday 13 Jan 2016, waqas.mehmood90 wrote:
> How to get user extention no in agi php scrip from which he's calling on
> ivr i am using cid and able to get his name but not his extention no
> please help me
Within the dialplan, what you are looking for would be ${CALLERID(num)} . So
you cou
On Tuesday 03 Nov 2015, sean darcy wrote:
> On 11/01/2015 12:38 PM, sean darcy wrote:
> > I'm not getting any ringing when I use option r with Dial:
> >
> > Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in
> > new stack
> >
> > Otherwise all works. The call goes through, good audio.
>
I recently had a nightmare building up some servers with these OpenVox cards.
Although I have used them successfully in the past, the chan_extra driver
building process has always been highly temperamental (although to be fair,
they have always worked fine once any necessary tweaking was done).
On Friday 25 Sep 2015, Ryan, Travis wrote:
> I've not used analog for quite some time. It seems it's not possible in
> asterisk to spoof a phone number/name on an analog call?
Probably not if you are using an analogue FXO connection to the exchange;
because there is no standardised way of communi
On Thursday 10 Sep 2015, 陳伯濤 wrote:
> Hi,
>
> I install Asterisk app_mp4, and use mp4save to record mp4 video file, then
> we can play the recorded mp4 file by using mp4play. But the recorded mp4
> file can not be played by MS media player or Quick Time Player. And we
> download mp4 file from inte
On Wednesday 02 Sep 2015, Avanish Shahi wrote:
> Now I’m trying to solve following problem. I have a requirement that
> each employee should have SIP phone at home, SIP phone in office,
> cell phone with same user.
>
>
> I want all those 3 phones to be “one extension”. So, if someone calls
> our
On Wednesday 12 Aug 2015, Jonas Kellens wrote:
> Hello
>
> I was wondering of it is possible to have Queue Agents with the same
> priority (penalty) but with a certain order ?
>
> So I have 20 Agents.
>
> Agent 1 till Agent 10 has penalty 1.
>
> Agent 11 till Agent 15 has penalty 2.
> (only con
On Friday 07 Aug 2015, Tech Support wrote:
> All;
>
> I know that there is no way to determine an exact number, or even a
> close number, but does anyone know a ballpark figure of how many Asterisk
> deployments are out there worldwide? How about the percentage of Asterisk
> PBX's compared to
On Thursday 06 Aug 2015, Jerry Geis wrote:
> I am looking for a push to talk solution does anyone know of a good
> PTT phone one that works with asterisk.
Um . Asterisk supports full-duplex telephony, so there's no need for any
of that "over to you, roger and out" business -- you can actuall
On Monday 03 Aug 2015, Eric Klein wrote:
> Hi all,
>
> Strange request, I have a customer where we are putting an Asterisk PBX in
> front of a legacy (non-VoIP) PBX. One of the requirements it that the
> Asterisk PBX have 2 PRI ports (on towards the legacy PBX and one towards
> the carrier) with t
On Saturday 01 Aug 2015, Murthy Gandikota wrote:
> Hi All
>
> Has anyone used Asterisk for a Call Center operation? What I mean is: given
> a list of phone numbers, can Asterisk dial each number, play a message and
> accept some DTMF?
Yes it can, very easily. But before you go too far, you need
On Wednesday 29 Jul 2015, Murthy Gandikota wrote:
> Hi All,As
> Downloaded latest version of Asterisk from www.asteriskwin32.com and
> installed on Windows 7.
Why?
Trying to get Asterisk to run on Windows is like trying to teach a gerbil to
bark. It's an extraordinary effort, and the result is
On Wednesday 15 Jul 2015, Luca Bertoncello wrote:
> But it seems, that I found the problem, adding:
>
> disallow=all
> allow=g729
>
> to the configuration of the peer for this number...
You need the following;
disallow=all
allow=alaw
in the configuration for *every* device. There is literally
On Thursday 16 Jul 2015, Thyda ENG wrote:
> I would like to see how can we config the asterisk to enable calling to
> multiple SIP number at the same time?
If you want to have a number that will call several phones when dialled, you
can do it in the Dial() command. The following example refers t
On Friday 10 Jul 2015, Thyda ENG wrote:
> Dear Sir,
>
> Does the asterisk support SMS feature ?
> If it does how can we config that ?
> I am waiting for your reply,Thank.
You need a suitable GSM card. We have used the OpenVox G400P / E400E series.
This has a facility for sending SMS directly v
On Wednesday 08 Jul 2015, Andrew Colin wrote:
> Hi Guys
>
>
>
> I am trying to write a macro for a call return so for example
>
> Anyone in the company transfers a call to another extension and it is not
> answered etc it must return to the person who did the transfer
>
> I have got it working
On Wednesday 08 Jul 2015, Thyda ENG wrote:
> Hi,
>
> I am new to asterisk, I have set up the asterisk server and successfully I
> could make the dialplan between 2 SIPs but when there are more than two
> sips calling each other, my dialplan seems doing the wrong routing to the
> sip. Do i need to
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