Re: [asterisk-users] ${ANSWEREDTIME} returning null

On Wednesday 17 Sep 2014, Anurag Rana wrote: > in dialplan: > exten=>h,n,NoOp(${DIALLEDPEERNUMBER) > > variable ${DIALLEDPEERNUMBER} is returning null. > > Suggestions please? > > Thanks > > Anurag Rana > http://newbie42.blogspot.in/ Asterisk has it mis-spelled as "DIALEDPEERNUMBER" (sic). Try

Re: [asterisk-users] GSM to GSM call with callerid passthrough

On Wednesday 17 Sep 2014, Rizwan H Qureshi wrote: > Hi All, > I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying > to use for kind of a call intercept between two GSM users. Call comes > through one SIM and goes out through another Sim with our Asterisk in > between to log

Re: [asterisk-users] fail2ban and pjsip in asterisk 12 and 13

(this is not where your reply belongs) On Monday 15 Sep 2014, Rainer Piper wrote: > Hi Patrick, > > github done ;-) > > what is HTH ??? HTH == Hope That Helps. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthsho

Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?

On Thursday 11 Sep 2014, rafa alfurqan wrote: > Hi, > > thank you for your repplied, > > > As you're on Ubuntu, you can begin with > > $ sudo apt-get install phpmyadmin > > i did that, so what i have to do for the configuration in asterisk so i > could remote to asterisk database from phpmyadmin

Re: [asterisk-users] How to use phpmyadmin to remotely access asterisk mysql database?

On Thursday 11 Sep 2014, rafa alfurqan wrote: > Hi, > > Could anyone help me to tell me about how to install and using phpmyadmin > to remotely access asterisk mysql database? > > I'm using asterisk 11.0.1 on ubuntu 10.04 > and mysql-server version is 5.1.73-0ubuntu0.10.04.1 (ubuntu) > > really

Re: [asterisk-users] Pattern Extension not working in Dialplan

On Monday 08 Sep 2014, Anurag Rana wrote: > @A J Stiles : If you could provide an example as you said, It would be very > nice. Thanks. This is excerpted from a dialplan application I wrote. It's actually a PIN entry but should be usable for any general purpose application.

Re: [asterisk-users] Pattern Extension not working in Dialplan

On Sunday 07 Sep 2014, Anurag Rana wrote: > Hi, > > I created a dummy dialplan where I ask the user to enter the age. > > [macro-age] > exten => s,1,Background(my/age) ;;Play recorded message to enter age > exten => s,n,WaitExten(10) > exten => _XX,1,Set(AGE=${EXTEN});; this line is not

Re: [asterisk-users] Asterisk secure fine tune - stop attack

On Thursday 04 Sep 2014, motty cruz wrote: > Hi A J, > believe me, I wish i do as you suggested, however I have a few extensions > outside the office with dynamic IPs, so that is not a possibility. If you know what ISPs they are using, then you can allow just those ISPs' address

Re: [asterisk-users] Asterisk secure fine tune - stop attack

On Thursday 04 Sep 2014, motty cruz wrote: > Hi All, > I see this kind of attack on our Asterisk Server, do you know how to block > that IP? Instead of blocking unwanted IPs, you should be permitting only wanted IPs. -- AJS Note: Originating address only accepts e-mail from list! If replying

Re: [asterisk-users] PJSIP issues with handling incoming calls

On Tuesday 02 Sep 2014, Nick Awesome wrote: > Hello guys. > > Have 2 external numbers that required registration on provider server, > > trunk1: 73432260005@80.75.132.66 > trunk2: 73432260050@80.75.132.66 > > Thing is I can’t figure out how to route them to different IVRs > > by default Asteris

Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

On Tuesday 02 Sep 2014, Jonas Kellens wrote: > On 02-09-14 11:34, Steven Howes wrote: > > On 2 Sep 2014, at 09:03, Jonas Kellens > > > > wrote: > >> So just before hanging up, I add a custom SIP-header : > >> > >> exten => s,n,SIPAddHeader(X-My-Hangup: MaxChan) >

Re: [asterisk-users] Setup Own IP PBX Server

On Monday 01 Sep 2014, Chandran Manikandan wrote: > Hi All, > I would like to Setup own IP PBX Server for our office. > I need to connect our all branch office with head quarter through local > extensions. > I need to receive and make call from our branch office and head quarter > using own DID num

Re: [asterisk-users] Billing software: Other than A2Billing because of the problem with the analogue channels

On Thursday 21 Aug 2014, bilal ghayyad wrote: > Hello; > > I am facing a trouble with A2Billing when using analogue lines because the > channels are not closing properly when dialing happen through A2Billing > (it seems the dialing scenario including the hangup is not handled > properly through A2

Re: [asterisk-users] Asterisk 12 on Debian Wheezy

On Tuesday 12 Aug 2014, Olivier wrote: > Hello, > > A couple of questions in relation with Asterisk 12 on Debian Wheezy. > > 1. Can paquet libpjproject-dev (from wheezy-backport) be installed as > the sole binary to add PJSIP stack to Asterisk 12 (compiled from > source) ? > > 2. When compiling

Re: [asterisk-users] Asterisk support for Bittorrent Bleep

On Monday 11 Aug 2014, Farid Fadaie wrote: > Hello, > > Full disclosure: my name is Farid Fadaie and I'm in charge of BitTorrent > Bleep (a private P2P SIP-based messaging application in early alpha) > http://blog.bittorrent.com/2014/07/30/building-an-engine-for-decentralized- > communications/ >

Re: [asterisk-users] The plain old PBX functionality

On Friday 08 Aug 2014, Gergo Csibra wrote: > Hi, > > back in the old analog telephony days there was "digital" PBX-es and > digital "system" phonesets. This phonesets have had many individual > illuminatable buttons connected with extensions. The PBX can show on > the buttons if some extension is

[asterisk-users] *SOLVED* Re: Anyone have any experience with inbound SIP trunks from Simwood?

On Wednesday 06 Aug 2014, I wrote: > I'm trying -- unsuccessfully! -- to configure an inbound trunk with > Simwood, and I was hoping someone on this list might have managed to do > this. > > I have configured some numbers to route to a SIP endpoint > %e164@customer's server > and convinced the c

[asterisk-users] Anyone have any experience with inbound SIP trunks from Simwood?

I'm trying -- unsuccessfully! -- to configure an inbound trunk with Simwood, and I was hoping someone on this list might have managed to do this. I have configured some numbers to route to a SIP endpoint %e164@customer's server and convinced the customer to open up UDP ports 5060 and 1 - 20

Re: [asterisk-users] Message Waiting indicator setup in ELASTIX ?

On Monday 04 Aug 2014, upendra wrote: > Hi, > > i wanted to know that if i have a message indicator SIP phone , then MWI > will work in ELASTIX ?? > > Let me know the Details of MWI and how test it. As long as the "message waiting" indicator can be controlled via SIP messages, it should Just Wo

Re: [asterisk-users] Asterisk 12.4.0 not able to install pjsip

(This is not where your reply belongs) On Friday 01 Aug 2014, Sameer Rathod wrote: > Hi Matthew, > > I know that no one is bounded to solve the issue for me. > I am new to asterisk that's why asking for help only. Pardon me if I did > something wrong. > > Please let me know where do I get config

Re: [asterisk-users] *SOLVED* SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

I have now fixed this issue, and am posting this for the benefit of anyone else who may be suffering with a similar problem. It was, as I suspected all along, a subtle misconfiguration at this end. The fix was to give the SIP trunk its own configuration stanza in sip.conf as follows; [sip_trun

Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

On Thursday 31 Jul 2014, James Thomas wrote: > Is the quality the same incoming from mobile as outgoing to mobile? It's a one-way trunk (outgoing only). Anyway, I've now fixed it, with help from the trunk provider. Details to follow in a separate message. -- AJS Note: Originating address o

[asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

I'm having a problem with a new SIP trunk. Calls within the UK to fixed lines are fine, but calls to mobiles have noticeably poorer audio quality. I thought it might have been a codec issue; we have used G.726 for internal and external calls (over primary ISDN and GSM). So I tried allowing "a

Re: [asterisk-users] incoming calls fall into echo test mode

On Saturday 19 Jul 2014, Norman Molhant wrote: > I tried many things on our FreePBX box and found out > the problem seems somehow linked with the customer's > extension (or phone number), not his inbound route > (changing the latter has no effect on the problem). > > Creating a new extension with

Re: [asterisk-users] Simultaneous Ring

On Friday 18 Jul 2014, Haley,Scott A wrote: > That worked. I had to use the *two* underscores in the agi script where I > was setting the values. Thanks. Glad you got it working in the end! I always like to use plenty of NoOp() statements to make sure the variables I'm setting are correct, espec

Re: [asterisk-users] Simultaneous Ring

On Friday 18 Jul 2014, Haley,Scott A wrote: > I have this working but I have one problem. I need to grab values from > variables that I have set in the calling context to dial. How would I do > that. I think you need to prefix your variable names with *two* underscores, to make them indefinitely

Re: [asterisk-users] Simultaneous Ring

On Wednesday 16 Jul 2014, Haley,Scott A wrote: > I have a need to issue a dial command to a number: > > same => n,Dial(${DIALGROUP1},${TIMER1},t) > > After a number of seconds, let's say 10 seconds. I want to dial another set > of numbers while continuing to ring, or interrupting the first group

Re: [asterisk-users] Digium E1 card stops working til disconnect machine power cord

On Thursday 10 Jul 2014, Ismael Gil wrote: > Hi there, > > In one of my asterisk installation, there is a Digium E1 pri card > connected. The asterisk and card are working properly. > The problem we have is that when a storm occurs in the area, the card > stops working, and E1 lines connected no

Re: [asterisk-users] Call rating software

On Wednesday 02 Jul 2014, Andrew Colin wrote: > Can you try maybe assist with this, as I have tried for ages and still cant > get it right. Firstly, have you got CDR working and writing to some sort of database? We use cdr_mysql; although the more modern recommendation is to use cdr_odbc (whic

Re: [asterisk-users] Call rating software

On Wednesday 02 Jul 2014, Sameer Rathod wrote: > Hi, > > I am facing issue in bypassing asterisk for audio call > can anyone help in packet to packet bridging I had posted the logs in > previous mail > If required again then please let me know Then why are you replying to a thread which, evidentl

Re: [asterisk-users] Call rating software

On Tuesday 01 Jul 2014, andrew Colin wrote: > Hi Guys > > Does anyone know of any good cdr rating software. > > I am looking for something that I can pull reports by extension. > Not a full billing solution like a2billing. Have you thought of rolling your own? It's not hard to write a program

Re: [asterisk-users] Best approach in asterisk configuration

On Monday 30 Jun 2014, sylvain GOTRI wrote: > Hi , > I have asterisk 1.8.5 installed on Centos 6. Now I want to configure my > PBX to work in my network. I see that I can do this with asterisk files > or use database like mysql to do it (realtime) > I want to know what is the best way and what can

Re: [asterisk-users] SugarAsterisk vs. ________

On Thursday 19 Jun 2014, thufir wrote: > http://www.voip-info.org/wiki/view/Asterisk+CRM+Integration > > lists a few options. I'm looking for, literally, the simplest FOSS CRM > for "click to dial" functionality, but don't know where to start. > > > > thanks, > > Thufir The Free version of S

Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)

On Thursday 05 Jun 2014, Mojtaba wrote: > My scenario is (2) After doing some tests with my own hardware, I'm now convinced that this is actually normal behaviour: As far as Asterisk is concerned, a call is deemed "answered" as soon as the hardware seizes the line. It is only "not answered" i

Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)

On Wednesday 04 Jun 2014, Mojtaba wrote: > Thank you for your replying. > Is there any way so that i could found the far end user pick up phone? I > could use Wait() function in dialplan but i dont how long (secend) > should be wait! > Thanks with Regards.Mojtaba I'm confused now. Please describe

Re: [asterisk-users] Channel is answered by FXO card before callee answered the phone(pick up phone)

On Wednesday 04 Jun 2014, Mojtaba wrote: > Hello Experts. > Im working with Asterisk PBXand freeswitch PBX. > I have a challenge with FXO card in Asterisk and i could not solve it yet. > I hope you could guide me in this regards. > When i want route the call to FXO channels, Before the callee answe

Re: [asterisk-users] SMS Capabilities

On Friday 16 May 2014, Jayson Devor wrote: > Hello Everyone, > > We have an order for SMS messaging. Can you gents and ladies be kind enough > to > disclose if SMS is possible using Asterisk? What is a quick way to test a > `Hello World` > to my cell. Finally, do all service providers support SMS

Re: [asterisk-users] AMR installation error

On Wednesday 30 Apr 2014, [Digital^Dude] ® wrote: > make gives this: > > codec_amr.c: In function 'amrtolin_sample': > codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this > function) > codec_amr.c:227: error: (Each undeclared identifier is reported only once > codec_amr.c:227:

Re: [asterisk-users] Asterisk -rx, how expensive is it? Should you avoid "spamming" it?

On Thursday 24 Apr 2014, Mikael Fredin wrote: > I will look into netcat as well, thank you There's not much to look into, really! It's just a command-line tool for connecting STDIN and STDOUT to a network socket. $ echo -e "WIBBLE\nWIBBLE\nWIBBLE" | nc somehost.co.uk 3245 will send WIBBLE WIBB

Re: [asterisk-users] cdr viewer for csv

On Thursday 24 Apr 2014, binary dreamer wrote: > really nice. but could tell me the way, play? I think we have gone as far as we can with this matter on this list, which is strictly for non-commercial discussion only. If you would still like to contact me off-list, please change the underscore i

Re: [asterisk-users] cdr viewer for csv

On Thursday 24 Apr 2014, binary dreamer wrote: > already logrotate is doing the file split every month. > how do you serve it in a webpage and which CGI script? You need a web server. You say you already have nginx; I'm not familiar with this, but it probably will do what you need. Read the doc

Re: [asterisk-users] cdr viewer for csv

On Thursday 24 Apr 2014, binary dreamer wrote: > hello everyone. > I am running asterisk and all of my CDRs are in the default csv. > the system is so limited to ram (only 256) and I cannot run MySQL or any > other program to give CDRs a fancy view. > at the moment the only other software running i

Re: [asterisk-users] Open Source Asterisk Polling Solution

On Wednesday 23 Apr 2014, Steve Edwards wrote: > On Tue, 22 Apr 2014, A J Stiles wrote: > > ...so absolutely *do not* pay money for a solution, and *do* insist on > > the Source Code and Modification Rights. > > Even an obvious and simple solution has value if it exceeds th

Re: [asterisk-users] Help with a bug

On Wednesday 23 Apr 2014, CDR wrote: > Dear friends > I filed a bug > https://issues.asterisk.org/jira/browse/ASTERISK-23656 > but I am wondering if somebody can figure a workaround. I am stuck > trying to deliver an application. > The case is this: A Record is executed and an immediate Playback >

Re: [asterisk-users] Open Source Asterisk Polling Solution

On Monday 21 Apr 2014, Nick Cameo wrote: > Hello Everyone, > > We are looking for a simple open source auto dialer with "polling" > capabilities. What we would like is a program that we can upload > leads to, and have asterisk: > > i) Dial numbers > ii) Play pre-recorded > iii) If user presses on

Re: [asterisk-users] Asterisk on OSX

On Friday 11 Apr 2014, Manu wrote: > Hi, > I used asterisk on Debian7 and it was good experience. > Now, i'm using osx on mac mini. > I'd like to install asterisk 12. > I tried to compile it and after lot of searches, I got it. > All sip accounts log in. I can call but I haven't any sounds. > - for

Re: [asterisk-users] Asterisk 1.6

On Friday 04 Apr 2014, Michelle Dupuis wrote: > Take a look a SecAst from www.generationd.com > > It does everything fail2ban does and more, including blocking users by > geography (we exclude all of Asia and Africa), detection of break-in > patterns (even if someone g

Re: [asterisk-users] process asterisk stop

On Thursday 03 Apr 2014, Павел Чашков wrote: > Could it be due to the version ubuntu? > Tried to put the asterisk 11 with ubuntu 10.04 - the same error occurs > intermittently. It's possible that this could be a libc issue. 2.11.1 is a very old version, after all. I would suggest upgrading your

Re: [asterisk-users] Asterisk CLI Banner

On Friday 28 Mar 2014, Haider Khalil wrote: > Thank you Thorsten Göllner. > Matthew, > What does violating license of Asterisk means ? Does it means I won't be > able to use any commercial modules or asterisk commercially ? I thought it > was open and anyone can change the code ? Haider Nothing, *

Re: [asterisk-users] Spammer direct replying to those posting on the users list

On Wednesday 26 Mar 2014, Tzafrir Cohen wrote: > [Intentionally ignoring the Reply-to header in this reply. And yes, this > is on-list] > > What if I wanted to reply to one of your messages off-list? > > My message would end up in asterisk_unwanted. And it did -- which is why I have added a spec

Re: [asterisk-users] Spammer direct replying to those posting on the users list

On Tuesday 25 Mar 2014, Digium's Asterisk Development Team wrote: > We apparently have a spam bot subscribed to the list and replying > *directly* to anyone who posts on the list. The e-mail address I use for this mailing list is asterisk_l...@earthshod.co.uk ; so I used the following procmail re

Re: [asterisk-users] Skip ./configure when source directory has not changed

On Monday 24 Mar 2014, Olivier wrote: > Is there a smart way to accelerate things a bit and skip ./configure when > source files have not changed since last configure command was previously > run ? That probably isn't what you really want to do. The most common reason why a build fails is because

[asterisk-users] Replying to Posts

(If you want to reply to this message, this is not where your reply goes) Please, for the benefit of anyone reading the archives in search of answers to a question, when replying to messages on this list, can everyone try to follow the natural flow of conversation? That is, position your reply

Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

On Friday 28 Feb 2014, Tahir Almas wrote: > 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 wh

Re: [asterisk-users] Hacking attempt, Asterisk 1.4

On Thursday 20 Feb 2014, Brynjolfur Thorvardsson wrote: > Every few weeks we get an attack that lasts about a minute or two, > resulting in our AGI script being overloaded. > > What happens is that somebody seems to be trying to connect from our server > – in my cdrs log I can see that they use a

Re: [asterisk-users] Asterisk NAT

On Wednesday 19 Feb 2014, Gholamreza Sabery wrote: > Hello, a few days ago I sent a question: > > http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html > > but no one answered me! I just want to know is it possible or not? There is a bit of a tendency on this list to ignore

Re: [asterisk-users] Dialer software for Asterisk...

On Friday 14 Feb 2014, Carlos Chavez wrote: > [stuiff omitted] > Does anyone know of a dialer for Asterisk that can > take several phone numbers for the same contact and if any of those > answers it will not try the other numbers? You can do that in your dialplan, without any additional software!

Re: [asterisk-users] ConfBridge on asterisk 11

On Friday 14 Feb 2014, Jerry Geis wrote: > I believe I am running an AGI (to put users in a conf) before the > confbridge is built. So the users are not really get in the conf... > > exten X,1,run agi to put users in conf > exten X,n,ConfBridge() > > How do I have in the dial plan ConfBridge() an

Re: [asterisk-users] Want Queues to ignore mobile operators voice mails and continue ringing...?

On Friday 14 Feb 2014, Tiago Geada wrote: > Hi all, > > > How does one detect the 'divert' to voicemail? > > Say we have PRI lines and as wel as SIP Trunks to connect to mobile phones. > How can asterisk know if the call is being diverted?? It can't. But you know (from the STD code) whether

Re: [asterisk-users] Asterisk Not Starting after YUM Update

On Wednesday 12 Feb 2014, Aldo Bergamini wrote: > Hi List, > > it feels silly, but here I am. > > My Asterisk box is useless, after running a long delayed yum update (Centos > box). If you got a new kernel as part of the upgrade, you will need to rebuild at least DAHDI and maybe Asterisk. Just

Re: [asterisk-users] Rejecting a call as if the extension does not exist.

On Friday 07 February 2014, John Kiniston wrote: > I'm trying to address a problem with users transferring to invalid > destinations. > > In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to > a context with a extension defined below to set some CDR variables before > the ca

Re: [asterisk-users] Integration with outlook

* THIS IS NOT WHERE RESPONSES BELONG * On Tuesday 28 January 2014, John Kiniston wrote: > On Tue, Jan 28, 2014 at 12:13 PM, bilal ghayyad wrote: > > Hello; > > > > Is there a method "way" to be able to dial the phone number through > > asterisk from the outlook email contact? > > > >

Re: [asterisk-users] Integration with outlook

On Tuesday 28 January 2014, bilal ghayyad wrote: > Hello; > > Is there a method "way" to be able to dial the phone number through > asterisk from the outlook email contact? If Outlook works anything like the KDE mail / news / calendar app Kontact, then you can specify a command to be run wheneve

Re: [asterisk-users] AsteriskNOW with AX1600P card

On Monday 27 January 2014, Fernando Pizarro wrote: > Hi all! > > I'm new with telephony cards and DAHDI drivers. I have installed > Asterisk NOW 3.0.0 and update to Asterisk 11.7.0, modules are update too. > > I'm following the installation guide of Atcom [1] for AX1600P analogic > card, modules

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

On Friday 24 January 2014, Mike wrote: > On 14-01-24 11:16 AM, Amit wrote: > > If I assume that Asterisk will write data on disk every second for > > each call, I will need disk array to support minimum of 500 IOPS. > > Where as if Asterisk push data every 2 seconds, I can deal with array > > suppo

Re: [asterisk-users] Register => plain text password

On Wednesday 22 January 2014, José Pablo Méndez Soto wrote: > Hello, > > Is there anyway to encrypt or scramble a bit the secret used to register > with a provider? Im talking about the > > register => fromuser@fromdomain:secret@host > > directive in > sip.conf

Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]

On Friday 17 January 2014, Olivier wrote: > 2014/1/16 A J Stiles > > If you need to install something on several boxen, you can make your own > > .deb > > package -fairly- easily -- > > For a complete packager beginner, how much time would it (very roughly) >

Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]

On Thursday 16 January 2014, Olivier wrote: > Thanks for replying. > > So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian > setup the alternatives are either : > - to install it from source > - tto build my own custom package removing this asterisk dependency (is it > easy o

Re: [asterisk-users] Asterisk QOS

On Tuesday 14 January 2014, richard.seg...@marisec.ca wrote: > I asked this on the list over the weekend, and likely missed a few people > inboxes. > > I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes > I have asterisk sending QOS data to the console. It seems I get QO

Re: [asterisk-users] screen capture for asterisk call center solution

On Friday 20 December 2013, Goke M Aruna wrote: > hello AJ, > Can I benefit from that code of yours? > regards As it was written for a very specific application, it's extremely unlikely that it will suit your environment. If you are wanting me to write some application-specific software for you

Re: [asterisk-users] screen capture for asterisk call center solution

* THIS IS NOT WHERE YOUR REPLY GOES * On Friday 20 December 2013, Goke M Aruna wrote: > Thanks AJ, > The capturing of agent activities on their desktop by the supervisor. > Regards Ah. That is really nothing to do with Asterisk. We run our business on a custom in-house app (written b

Re: [asterisk-users] screen capture for asterisk call center solution

On Friday 20 December 2013, Goke M Aruna wrote: > Thank you AJ, > Just want to know from people who uses asterisk as call center solution, > how and what screen capture solution / applications are in use. What do you mean by "screen capture" ? -- AJS Answers come *after* questions. -- ___

Re: [asterisk-users] screen capture for asterisk call center solution

On Friday 20 December 2013, Goke M Aruna wrote: > Hi all, > Can someone suggest a good solution on "agent screen capture" in asterisk > call center? open source preferred but cheap offer is welcome too. ??? What are you trying to do? Asterisk is *just* a telephony construction kit. It doesn't k

Re: [asterisk-users] Asterisk on Windows

On Wednesday 04 December 2013, CDR wrote: > Digium is 100% lost in the map. If they would come up with a Paid > version of Asterisk, one that would use the .NET framework in Windows, > something simple to install, they could go public on the product. Why would they? They already have it working w

Re: [asterisk-users] Asterisk SIP server on windows

On Wednesday 04 December 2013, Ruddy Gbaguidi wrote: > Hi all, > > I need to build an application that will be an SIP server program that will > run on Linux and Windows. > > The sip server need only some features such as be able to : > > - Register sip endpoints > > - Answer

Re: [asterisk-users] issue with speech in IVR

On Thursday 28 November 2013, Salaheddine Elharit wrote: > hi > i follow your dialplan but the issue still the same ican't stop the speech > and go to another context > > any other idea please > > best regards . Well, the Background() application should definitely allow you to interrupt a soun

Re: [asterisk-users] issue with speech in IVR

On 28/11/13 15:36, Salaheddine Elharit wrote: hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context any other idea please best regards . It sounds as thgough the DTMF tones are not being sent in a way that Asterisk is seeing . What typ

Re: [asterisk-users] issue with speech in IVR

On Wednesday 27 November 2013, Salaheddine Elharit wrote: > hello list > > i have an IVR menu in asterisk 1.4 > > [stuff deleted] > > my problem when the customor call the number 600 and press 1 in order to go > to the project menu he must wait all the speech music1 music2 and music 3 > > if t

Re: [asterisk-users] Asterisk 1.8.24 : illegal instruction

On Wednesday 20 November 2013, Jonas Kellens wrote: > Hello, > > I have installed asterisk 1.8.24 (from source) but I can not start up > Asterisk : > > > [root@sip32 admin]# /usr/sbin/asterisk -r > Illegal instruction Are you using a VIA C6/C7 processor (often found soldered to tiny motherboa

Re: [asterisk-users] Make phone ring through webserver using Asterisk

On Saturday 16 November 2013, akhilesh chand wrote: > What is the easiest way? And how can it be implemented? > i don't care if i can hear something, it's enough that it rings Just inject a callfile into /var/spool/asterisk/outgoing/ . One end is the extension you want to ring, the other end is

Re: [asterisk-users] Recurring SIP problem with asterisk 11.6 & 11.7

On Wednesday 13 November 2013, Jeremy Kister wrote: > On 11/12/2013 8:46 PM, Duncan Turnbull wrote: > > Any chance DNS is dying about the same time the problem occurs > > good idea, but I don't use DNS anywhere in Asterisk. well, except for > sip.conf:externhost. it's all IP addresses. That doe

Re: [asterisk-users] two steps when calling from web!

(re-ordered correctly) On Tuesday 12 November 2013, akhilesh chand wrote: > I'm making a call from web(click to dial) and able to successfully dial to > number but problem with when i dial a number call goes to first client and > after that call come into my softphone show me "Answer" and "Declin

Re: [asterisk-users] two steps when calling from web!

On Friday 08 November 2013, akhilesh chand wrote: > When I calling a number from web, my softphone show me "Answer" and > "Decline" bottoms, and then I have to click Answer to call the number. it > seems it is two step to calling the number. If I type the number direct to > my client softphone, it

Re: [asterisk-users] No matching peers message has gone (1.8.23.1)

Hi Ish, I assume you are using Fail2Ban to monitor the logs for dictionary attacks - If so, the following regex should work for 1.8: Registration from '.*' failed for '(:[0-9]{1,5})?' - Wrong password Registration from '.*' failed for '(:[0-9]{1,5})?' - No matching peer found Registration from

Re: [asterisk-users] Redirect a GSM call through Wifi to a SIP phone

On Friday 01 November 2013, Sil wrote: > I want to know if it's possible to use a mobile phone application for > redirect automatically incoming calls of a GSM phone connected to Wifi > network to a Sip phone. > I've try to use different mobile phones SIP clients without any success. > No one of th

Re: [asterisk-users] issue with dahdi_channels.conf

On Thursday 31 October 2013, Salaheddine Elharit wrote: > Hello list > > > i have an issue with my dahdi_channels.conf > > in span 1 when i use it like below i can do my outband calls without issue > > ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER) > group=0,11 > context=from-pstn > swi

Re: [asterisk-users] issue after install dahdi

On Tuesday 22 October 2013, Salaheddine Elharit wrote: > hello yes this is a fresh install > > [trunkgroups] > trunkgroup => 1,16 > spanmap => 1,1,1 > > [channels] > #include dahdi-channels.conf > > context=default > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=ye

Re: [asterisk-users] Asterisk-12 issue after successful installation

On Monday 21 October 2013, virendra bhati wrote: > Hi Team, > > I have installed asterisk-12 Beta but when I try to asterisk start then get > below issue. > > *[root@cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r > asterisk: error while loading shared libraries: libjansson.so.4: cannot > ope

Re: [asterisk-users] Access PBX from internet - best practice

On Thursday 17 October 2013, richard.seg...@marisec.ca wrote: > The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under > this scenario. Basically this setup is for people who are traveling, and > may be using a smart phone at an airport (or something similar). The idea > is th

Re: [asterisk-users] What linux distro most popular for Asterisk

On Wednesday 16 October 2013, Michelle Dupuis wrote: > Is there a recent survey of that Linux distro and version people are using > for the Asterisk installations? I recall seeing a pie chart over a year > ago (I think on a wiki but I can't find it again)also hoping for > something more curren

Re: [asterisk-users] How to disable Internal call ?

On Wednesday 16 October 2013, akhilesh chand wrote: > Dear All, > > I want to disable internal call facility.Means agent(4002) does not make > call to agent(4003) or other extensions. Just configure extension 4002 to be in a context which doesn't have an extension matching 4003. (And make sure

Re: [asterisk-users] is g729 codec free? or under license???

On Saturday 12 October 2013, s m wrote: > i have freebsd 8.2 and asterisk 1.8.22 but there is no compatible > codec for Xeon Intel in the list. it means that i can't use codec g729 on > my system??? or can i use codec for another type of hardware for my system? > anyone has any experience? Xeon is

Re: [asterisk-users] asterisk 11.6 nat problem

Jeremy Kister wrote: > > using asterisk 11.6.0-rc1 i just converted my "nat=yes" to > "nat=auto_force_rport,auto_comedia" > > I have my asterisk box on the same subnet as a cisco 1760 (vgw1). > > a few times per day, Asterisk thinks vgw1 is dead (by qualify/options). > A 'sip reload' always

Re: [asterisk-users] GSM to SIP Adapter

On Friday 11 October 2013, Tarek Sawah wrote: > Greetings,I'm looking for a really cheap GSM-SIP gateway, Single channel > (one SIM card). any suggestions? > > Tarek Sawah We've been using OpenVox G400P cards (PCI; there is also a G400E, which is PCI express for newer motherboards). Sends and

Re: [asterisk-users] Read Telnet Packet

On Friday 11 October 2013, akhilesh chand wrote: > Dear All, > > I want to read telnet packet continuously whenever a new call is originated > and store into a variable after that pass into window server. I have > written a Perl script to read telnet packet but problem is that whenever I > execute

Re: [asterisk-users] utils.c: fwrite() returned error: Broken pipe how to solve it ???

On Thursday 10 October 2013, akhilesh chand wrote: > Dear all, > > I want to make call through socket i have set code given below: > > #!/usr/bin/perl -w > > use IO::Socket::INET; > > > sub asterisk_command () > { > # my $command=$_[0]; > my > $ami=IO::Socket::INET->ne

Re: [asterisk-users] How to disable call transfers?

On Tuesday 08 October 2013, akhilesh chand wrote: > Dear All, > > > I want to disable call transfers internally.Means agent(4002) does not > transfer call to agent(4003) or other extensions. > But i want to create two extensions as supervisor who are able to take a > internal call.Suppose to agen

Re: [asterisk-users] Extensions fail to register themselves when all trunks are unreachable.

On Tuesday 01 October 2013, Francesco Namuri wrote: > Hi, > my asterisk server has a strange behavior when all trunks are > unreachable (for example due to internet connection), it doesn't accept > registrations from any internal extension, and, obviously, I can't do > any internal call. Asterisk

Re: [asterisk-users] OpenVox G400P network registration problems ** SOLVED **

It took an OpenVox engineer to sort out this obscure problem in the end, but it was pretty much as I suspected: the Quectel M20 GSM module serving span 1 was stuck in an unusual state, in which it would only operate in the 900 MHz band. Fine for O2, Vodafone and Tesco; but no good for T-Mobile

Re: [asterisk-users] The call is established but without exchanged voice packets

Asmaa Ahmed wrote: > > Indeed I missed your previous message! > After changing the externip, it worked successfully... The sip > session is established with the complete three-way handshake, and > the voice packet is exchanged with no problem! > > Many thanks. Asmaa, That's great news!! I

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