Re: [asterisk-users] Asterisk Voicemail changes

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

Re: [asterisk-users] Simplest way of executing a non-blocking (async) python AGI script?

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

Re: [asterisk-users] Autodialer - call simultaneously to both ends

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

Re: [asterisk-users] Is this the future of telephony?

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

Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

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

Re: [asterisk-users] Working around missing libmyodbc in Debian Stretch

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

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

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

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

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

Re: [asterisk-users] app_jack unavailable

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

Re: [asterisk-users] app_jack unavailable

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. > >

Re: [asterisk-users] Dial an extension to modify dialplan

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

Re: [asterisk-users] log incoming calls without answering

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 > >

Re: [asterisk-users] Voicemail asking for login

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 >

Re: [asterisk-users] Voicemail asking for login

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

Re: [asterisk-users] PBX selection

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

Re: [asterisk-users] restart system from extension

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

Re: [asterisk-users] Alphabet character in destination number (CDR)

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

Re: [asterisk-users] Something similar to Doxygen for standard dialplan?

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

Re: [asterisk-users] BUG or ???

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

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

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

Re: [asterisk-users] [OT] Downloading Recqual

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

Re: [asterisk-users] Call List Campaign to an IVR

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

Re: [asterisk-users] Setup DID

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

Re: [asterisk-users] Replacing PBX during a call in progress

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

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

* 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

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

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

Re: [asterisk-users] failing to start asterisk on centos7

** 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

Re: [asterisk-users] failing to start asterisk on centos7

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

Re: [asterisk-users] Asterisk compatibility with SMS services

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

Re: [asterisk-users] _FAX_. extension refuses to work !

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

[asterisk-users] Triggering an AGI script when a queued call is answered

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

Re: [asterisk-users] info about DID.

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

Re: [asterisk-users] Incoming Call by DID

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

Re: [asterisk-users] Hello again

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

Re: [asterisk-users] mysql phonebook

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 => >

Re: [asterisk-users] Multiple phones when one is unregistered

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/user2/user3) > > The idea is that any of the three users can answer the phone to let > someone in. The problem is that

Re: [asterisk-users] Removing mailbox and password prompt for voicemail

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

Re: [asterisk-users] SIP trunk

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

Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

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

Re: [asterisk-users] VoiceMail and SMS

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

Re: [asterisk-users] VoiceMail Audio playing

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

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

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

Re: [asterisk-users] Function SHELL not registered

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

Re: [asterisk-users] rasberry pi

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

Re: [asterisk-users] Function SHELL not registered

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

Re: [asterisk-users] Including doesn't have any effect

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 in

Re: [asterisk-users] Including doesn't have any effect

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)

Re: [asterisk-users] Questions... connecting Asterisk to the World

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

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

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 =>

Re: [asterisk-users] click2call for conferencing two mobile numbers

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

Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6

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

Re: [asterisk-users] my dahdi dont'n start

*** 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

Re: [asterisk-users] "Follow me" with Asterisk that detects cellphone voicemail and similar announcements

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

Re: [asterisk-users] my dahdi dont'n start

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

Re: [asterisk-users] my dahdi dont'n start

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

Re: [asterisk-users] recreating extensions.conf from live dialplan ?

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 knocking up

Re: [asterisk-users] recreating extensions.conf from live dialplan ?

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

Re: [asterisk-users] Best timing source?

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

Re: [asterisk-users] Doing asteriksk with a sip trunk

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.

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

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

Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

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

Re: [asterisk-users] Phone Number Validation

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

Re: [asterisk-users] Mobiles not detecting as BUSY until Dial() timeout completes

On Thursday 24 Mar 2016, Tony Mountifield wrote: > In article <201603241343.24128.asterisk_l...@earthshod.co.uk>, > A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > > When placing a call over a SIP channel to a mobile phone, if the phone is > > engaged, it

Re: [asterisk-users] Updating Asterisk

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

[asterisk-users] Mobiles not detecting as BUSY until Dial() timeout completes

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

Re: [asterisk-users] How to recognize a name spelled letter by letter ?

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

Re: [asterisk-users] How to recognize a name spelled letter by letter ?

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

Re: [asterisk-users] One phone, many names / Was: Loss of devices registration (pjsip)

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

[asterisk-users] *SOLVED* Re: Dialplan question: Variables in GoTo() ?

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

[asterisk-users] Dialplan question: Variables in GoTo() ?

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

Re: [asterisk-users] Dial your phone and contact phone from within outlook?

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

Re: [asterisk-users] Asterisk 13.6.0/The simplest TCP configuration does not work

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: > >

Re: [asterisk-users] SIP URI set 'telephone-context='

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: ". > >

Re: [asterisk-users] 1000 analogue lines with asterisk

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

Re: [asterisk-users] sql schema without alembic

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

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

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 > >>

Re: [asterisk-users] PJSIP Stun/ICE

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

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

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

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 19

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.

Re: [asterisk-users] is there some blocking in 11.21.0

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

Re: [asterisk-users] is there some blocking in 11.21.0

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

Re: [asterisk-users] 488 Not acceptable here

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]

Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)

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

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 8

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

Re: [asterisk-users] no ringing tone with Dial option r

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. >

[asterisk-users] OpenVox G400P / G400E -- a warning

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

Re: [asterisk-users] caller id spoofing/setting on analog

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

Re: [asterisk-users] Asterisk app_mp4 problem

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

Re: [asterisk-users] Single SIP User on multiple location

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

Re: [asterisk-users] Call Queues : linear strategy WITH priority

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 contacted if

Re: [asterisk-users] How many Asterisk deployments?

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 the

Re: [asterisk-users] PTT push to talk solution

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 actually

Re: [asterisk-users] Looking for PRI Card with automatic fail over

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 the

Re: [asterisk-users] Call Center

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 to

Re: [asterisk-users] Windows Asterisk Help

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

Re: [asterisk-users] How to enable group call

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 to

Re: [asterisk-users] Problem no voice

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 no

Re: [asterisk-users] Asterisk SMS

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 via

Re: [asterisk-users] Call Return

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 but if

Re: [asterisk-users] How to handle multiple lines call

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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