Re: [asterisk-users] asterisk-users Digest, Vol 221, Issue 2

Have you tried using the EVAL function? On Tue, Jan 24, 2023, 7:38 PM wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 17

Re: Get a SHAKEN Identity Token (Alexander Perkins) Saint Michael 1:06 PM (0 minutes ago) to Asterisk Please look at this https://issues.asterisk.org/jira/browse/ASTERISK-28924 I have a solution that works for any version of Asterisk, if interested contact me at venefax at the Google mail service

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 7

Stir Shaken Asterisk cannot do that, but my company can give you Stir Shaken for Asterisk, via ODBC, any version. Please contact me via email venefax at the google mail system Philip Orleans On Fri, Jan 8, 2021 at 1:00 PM wrote: > Send asterisk-users mailing list submissions to > asteris

Re: [asterisk-users] asterisk-users Digest, Vol 193, Issue 15

memory vs disk cache > This is an issue that has plagued Asterisk since day one. Basically there >> is no solution available because there is no way to set aside memory to be >> kept from a growing disk cache. I did some research and this looks like a >> bad design from the Kernel people. Meanwhil

Re: [asterisk-users] asterisk-users Confbridge

Sorry about the trouble. Unsubscribed that user from the mailing lists. Matthew Fredrickson On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote: > > I'm online on this site! > So contact me in my profile: > here > -- > _ > -- Bandwi

Re: [asterisk-users] asterisk-users Confbridge

I'm online on this site! So contact me in my profile: galleries.daswanitailors.com here -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://communit

Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

In article <874506323.2924334.1567645810...@mail.yahoo.com>, bilal ghayyad wrote: > > Thank you a lot for your kindly help and reply. Actually it helped me a > lot.I was using _X. in the extensions.conf at > the trunkinbound context.Can you advise me what is the difference between _X. > and s?

Re: [asterisk-users] asterisk-users Digest, Vol 181, Issue 3

Thank you a lot for your kindly help and reply. Actually it helped me a lot.I was using _X. in the extensions.conf at the trunkinbound context.Can you advise me what is the difference between _X. and s? In other words, when it is better to use s and when it is better to use _X.? Again, I am ful

Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

. From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Israel Gottlieb Sent: Monday, July 1, 2019 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1 how about sticking in a pbx

Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1

how about sticking in a pbx between [c] and [h] so when [h] hangsup you send to [s] if that is 3rd party else i dont see how you could redirect [c] at all else maybe ask them to have [h] redirect [c] to [s] then [h] will also be out of the call On Mon, Jul 1, 2019, 20:03 Send asterisk-users mail

Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote: > Joshua > Is there a way in PJSIP to send the audio between the parties always, > unless one of the parties is behind a NAT? > A session refresh would work. > That my only problem with PJSIP. This is routine in the old sip channel. Any such

Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

Joshua Is there a way in PJSIP to send the audio between the parties always, unless one of the parties is behind a NAT? A session refresh would work. That my only problem with PJSIP. This is routine in the old sip channel. On Sat, May 25, 2019 at 1:03 PM wrote: > Send asterisk-users mailing list

Re: [asterisk-users] Asterisk users survey

On Tue, Mar 12, 2019, at 3:05 AM, Stefan Viljoen wrote: > Hi Joshua > > Does the survey imply that there are big changes coming for Asterisk? > > E. g. features or facilities will be dropped / deprecated from the open > source version in new releases, big changes to existing facilities / > pro

Re: [asterisk-users] Asterisk users survey

Hi Joshua Does the survey imply that there are big changes coming for Asterisk? E. g. features or facilities will be dropped / deprecated from the open source version in new releases, big changes to existing facilities / protocols, what is supported officialy by Digium for the official version

Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 9

Sebastian, Well, this can't be problem with trunk because:1. Call coming from outside, so trunk works2. sip show registry shows it registered. Trunk allows for 2 channels which is not a problem here either It's just weird that out of 4 queue member only 2 being called and log doesn't show anythin

Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 1

Expose some logs (full file in /etc/asterisk/). Also you can active CDR debug in cli to see exactly what is going on there. Asterisk 13 brings many changes in CDR procedures, but I am using it for a while and I don’t have any problem. > On 2 Nov 2018, at 18:00, asterisk-users-requ...@lists.digi

Re: [asterisk-users] asterisk-users Digest, Vol 168, Issue 14

The limit 10 is just an assumption. For real number I've set in my conference server is 350 max participants to join. As per the load testing I've performed on the server, after 400+ participants voice is getting choppy so I've set the max limit to 350 globally for safe reason. If 5 other particip

Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 21

Hi, I sent the requested information. I always get this responde: Response: Success Message: Timeout Set But keep the old timeout, interestingly, decreasing the timeout works perfectly. The problem is increasing. > Which verson? Version Asterisk 1.8.32. > > Show us what command you are sending?

Re: [asterisk-users] asterisk-users Digest, Vol 167, Issue 17

Hi Daniel Thanks for the reply! Yes, turns out it was all my fault, I had a line feed character (0x0a a.k.a printf("\n")) in one of the Asterisk channel variables passed via system() / shell() to my target script. It seems 13.22.0 (I'm using the same version as you) reacts to a line feed in t

Re: [asterisk-users] asterisk-users Digest, Vol 160, Issue 5

Get Outlook for Android From: asterisk-users-boun...@lists.digium.com on behalf of asterisk-users-requ...@lists.digium.com Sent: Monday, November 6, 2017 6:00:01 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 160, Issue 5 Send asterisk-use

Re: [asterisk-users] asterisk-users Digest, Vol 153, Issue 28

Install app_mysql.so after then check your dialplan . On Sun, Apr 23, 2017 at 10:30 PM, wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listi

Re: [asterisk-users] asterisk-users Digest, Vol 152, Issue 31

Dear Saint Michael, I will be grateful if you could introduce me to the Company that offers the translation service. I am really interested in google voice. Sincerely, Michael Codjoe On 29 March 2017 at 17:00, wrote: > Send asterisk-users mailing list submissions to > asterisk-users@

Re: [asterisk-users] asterisk-users Digest, Vol 151, Issue 23

Theory: The carrier is not responding with 100 Trying in the expected time. Hence, Asterisk is sending the INVITE again. On Wed, Feb 22, 2017 at 1:00 PM, wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the Wor

Re: [asterisk-users] asterisk-users Digest, Vol 150, Issue 17

hi, Do you edit your voicemail.conf? [default] 1091=(number to access your voicemail in your phone ex: 1234) 2017-01-25 16:00 GMT-02:00 : > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit

Re: [asterisk-users] asterisk-users Digest, Vol 147, Issue 5

Hi all ! Thanks for your feedback and sory for the delay. Respond: > Date: Mon, 3 Oct 2016 21:05:55 -0300 > From: Marcelo Terres > > I think that you need the dev files too. In Debian 8, the package is > libmysqlclient-dev. > > But Debian 8 uses libmysqlclient-18. Where did you get the 20 ? > >

Re: [asterisk-users] asterisk-users Digest, Vol 140, Issue 15

It does not work. That was the first think I tried. Maybe we need a patch? I don't want to file a bug if there is a workaround. On Sat, Mar 12, 2016 at 1:00 PM, wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via

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. Wh

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

On Mon, 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 If you start your thread with a relevant subject you may get

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

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 Sent from my Samsung Galaxy smartphone. Original message From: asterisk-users-requ...

Re: [asterisk-users] asterisk-users Digest, Vol 138, Issue 10

Hello sir have a nice day and sory to distrb you again . I am unable to get solution of my problem Unable to get user extension no from cid Sent from my Samsung Galaxy smartphone. Original message From: asterisk-users-requ...@lists.digium.com Date:15/01/2016 11:00 PM (GMT+

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 cou

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

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 Sent from my Samsung Galaxy smartphone. Original message From: asterisk-users-requ...@lists.digium.com Date:13/01/2016

Re: [asterisk-users] asterisk-users Digest, Vol 130, Issue 14

- Original Message - > From: "Steve Davies" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, May 13, 2015 11:39:29 AM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in > dropped calls after 32 seconds > > Hi, > > In my experience,

Re: [asterisk-users] asterisk-users Digest, Vol 126, Issue 18 mtr

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-requ...@lists.digium.com wrote: > > Send asterisk-users mailing list submissions to >asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visi

Re: [asterisk-users] asterisk-users Digest, Vol 125, Issue 33

On 30/12/14 06:22, Stefan Viljoen wrote: ... > > I think you're overcomplicating your problem. (if I understand you > correctly!) This is probably right ;) in both parts > > Your scenario is almost exactly ours, except we use ATCOM-820P's (with LCD > displays) and no softphones. So incoming CID

Re: [asterisk-users] asterisk-users Digest, Vol 125, Issue 33

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is qu

Re: [asterisk-users] asterisk-users Digest, Vol 123, Issue 38

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to t

Re: [asterisk-users] asterisk-users Digest, Vol 119, Issue 7

Hello, You could use Timeout(digit) in dialplan,like this: . . . same => n,Set(Timeout(digit)=10);10 sec -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introduct

Re: [asterisk-users] asterisk-users Digest, Vol 119, Issue 7

alf Of CDR Sent: Saturday, June 07, 2014 1:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk-users Digest, Vol 119, Issue 7 I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtm

Re: [asterisk-users] asterisk-users Digest, Vol 119, Issue 7

I changed in asterisk.conf mindtmfduration = 50 The inter-digit duration is for this function SendDTMF when we dial out dtmf The question is, how do I change it without changing the source code? On Sat, Jun 7, 2014 at 1:00 PM, wrote: > Send asterisk-users mailing list submissions to > as

Re: [asterisk-users] asterisk-users Digest, Vol 117, Issue 7

Hi Patrick, Thanks a lot for your quick help. Yes, I configured the NAT options in sip.conf. BTW, I am using 12.1.1, will try 11.8.1 and see if I can make it work. Thanks again, William === Date: Sat, 05 Apr 2014 23:38:32 +0200 From: Patrick Laimboc

Re: [asterisk-users] asterisk-users Digest, Vol 109, Issue 30

On Fri, Aug 30, 2013 at 11:44 AM, CDR wrote: > > I am stumped > In features.conf,I programmed this > > [applicationmap] > Answer0 => 0,self/both,Macro,nway_start > > But do I pass an argument or parameter to my macro? I tried > Answer0 => 0,self/both,Macro,nway_start^0 > Answer0 => 0,self/both,Ma

Re: [asterisk-users] asterisk-users Digest, Vol 109, Issue 30

I am stumped In features.conf,I programmed this [applicationmap] Answer0 => 0,self/both,Macro,nway_start But do I pass an argument or parameter to my macro? I tried Answer0 => 0,self/both,Macro,nway_start^0 Answer0 => 0,self/both,Macro,nway_start,0 but the usuar variable ${ARG1} is empty in my d

Re: [asterisk-users] asterisk-users Digest, Vol 108, Issue 14

Unsubscribe Elvin G. Nodalo -Original Message- From: asterisk-users-requ...@lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.c

Re: [asterisk-users] asterisk-users Digest, Vol 108, Issue 14

Unsubscribe Elvin G. Nodalo -Original Message- From: asterisk-users-requ...@lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.c

Re: [asterisk-users] asterisk-users Digest, Vol 106, Issue 41

hi, try exten = .,n,System(wget -P /var/log/asterisk/wgets 'http://theUrlYouWantToCall' &) kind regards, andre Am 30.05.2013 19:00, schrieb asterisk-users-requ...@lists.digium.com: > Message: 9 > Date: Thu, 30 May 2013 15:06:59 + > From: Salaheddine Elharit > Subject: [asterisk-users] h

Re: [asterisk-users] asterisk-users Digest, Vol 106, Issue 23

Hi, I am looking for the easiest and fastest way to send or pull callerID and extension# from asterisk to a web server for sales data lookup and display. User would be logged in with known extension. I have been looking at several options but was hoping someone here would have the best one. I w

Re: [asterisk-users] asterisk-users Digest, Vol 105, Issue 39

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was

Re: [asterisk-users] asterisk-users Digest, Vol 104, Issue 53

Roberto estoy en uruguay en estoos momentos. Recien lllego el miercoles El mar 31, 2013 1:59 p.m., escribió: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/

Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 37

Chris, Thanks for answering my message. I'm currently using version 10.5.1. I included the error message on the dial plan to show what errors I was displaying. The call goes through after that error message is displayed. As soon as I hear the phone ring, it drops my call on the calling phone y

Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 9

> > Here is my IP-PBX setupmy setup is : sips softphone <-> asterisk <-> xorcom > > PSTN gateway <-> pstn line to telcoi'm using xlite for windows > > > when I make a phone call (sip - outgoing channel),I can hear my own voice > > so clear. it's very annoying mewhen talking a little loud... any

Re: [asterisk-users] asterisk-users Digest, Vol 98, Issue 38

Hi Danny, Thank you for your prompt response. The way you are suggesting is great . Infect asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not fulfill my need. There is and is

Re: [asterisk-users] asterisk-users list testing - hopefu...@hotmail.com

Okay Sent from my iPod On Jul 14, 2012, at 9:35 PM, rnew...@digium.com wrote: > This message was sent to test a problem with the mailing list. > Please ignore it, and we apologize for any inconvenience. > -- _ -- Bandwidth and

Re: [asterisk-users] asterisk-users list testing - msegovia....@gmail.com

Gracias 2012/7/14 > This message was sent to test a problem with the mailing list. > Please ignore it, and we apologize for any inconvenience. > -- Maria Segovia -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 20

Anybody, can you please share your thoughts to overcome this issue? > Hi, > > I'm getting error: ' FAX session '9' is complete, result: 'FAILED' > (FAX_FAILURE_PARTIAL), error: '3RD_T2_TIMEOUT', pages: 1, resolution: > '204x196', transfer rate: '9600', remoteSID: '' ' when I tried send fax > more

Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 14

Thank you John. This is a much more elegant solution since I have already defined 'mailbox' for my SIP device. I'm now using this in my dial plan. Chet Stevens asterisk-users@lists.digium.com writes: >> Also, related to this question; is there a "best" or recommended method to >> determine the d

Re: [asterisk-users] asterisk-users Digest, Vol 95, Issue 6

I figured out the problem. Actually the sending fax machine speed was set as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully set to use T.38 protocol. After that I was able to receive fax. Thanks Tim for assisting me out :). > - Original Message - > > > Hi Tim, > >

Re: [asterisk-users] asterisk-users Digest, Vol 92, Issue 43

asterisk-users-requ...@lists.digium.com wrote: >Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with

Re: [asterisk-users] asterisk-users@lists.digium.com Nacha Alert ID10416

<< asterisk-users@lists.digium.com wrote: << Please click the link the NACHA site and update your user account:ID0664474 Interesting. Came from tipas...@gmail.com Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither

[asterisk-users] Asterisk-users caller ID

Hello, I have a server that connects to my Voice Server provider so far is working great! I have a second server that I want to set caller id to a different number second server I'm going to call it server B. And server B will go through server A which is connected to my Voice Server Provider. Thu

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 32

On 11-12-17 12:56 PM, Chet W. Stevens wrote: I am out of the office until 12/16 but I will still be checking my messages. For immediate assistance, please call Telecommunication Services at 799-6543. Thank you. This is the 3rd auto-notification we got from your mail client. Please disable the

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 32

I am out of the office until 12/16 but I will still be checking my messages. For immediate assistance, please call Telecommunication Services at 799-6543. Thank you. Chet Stevens Telecommunication Services Clark County School District -- _

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 29

I am out of the office until 12/16 but I will still be checking my messages. For immediate assistance, please call Telecommunication Services at 799-6543. Thank you. Chet Stevens Telecommunication Services Clark County School District -- ___

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 13

On Fri, Dec 09, 2011 at 10:55:07AM -0600, Danny Nicholas wrote: > You have to install the dahdi framework, but no actual device is > required (dahdi-dummy is loaded). Minor clarification: Since DAHDI-Linux v2.3.0 dahdi_dummy.ko is no longer loaded. You just need dahdi.ko loaded by itself and it wi

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 13

: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 13 Yes, DAHDI is a timing source and meetme depends on DAHDI for voice mixing. You can check details here http://www.asterisk.org/docs/asterisk/trunk/applications/meetme >Install DAHDI then !!? >

Re: [asterisk-users] asterisk-users Digest, Vol 89, Issue 13

Yes, DAHDI is a timing source and meetme depends on DAHDI for voice mixing. You can check details here http://www.asterisk.org/docs/asterisk/trunk/applications/meetme >Install DAHDI then !!? >On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra < >durgesh.mis...@rancoretech.com> wrote: >> In make me

Re: [asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

Hey, I don;t think asterisk-guru could've been wrong on this one - possibly different scenario than your's. Anyway I see what you did there ! There is no need for separate context for incoming or outgoing if you don't want. What you are doing is *exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401**) * * *

[asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

Sorry if this question already asked. I'm implementing Voip with asterisk and grandstream gxw4108, according from the manual, for connecting with PSTN I must configure one SIP account and assign that for dialing the PSTN so in my sip.conf I configure SIP account(extension) : [1401] type=friend use

Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23

Behalf Of Tahar .H Sent: Saturday, August 13, 2011 7:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23 hi folks, can some one please explain to me what this one stands for : Exten => 1234,1,read(numtodial,enternum,10,skip,1,10) t

Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23

On Sun, 14 Aug 2011, Tahar .H wrote: can some one please explain to me what this one stands for : Exten => 1234,1,read(numtodial,enternum,10,skip,1,10) [core] show application read -- Thanks in advance, - Steve Edwards

Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 23

hi folks, can some one please explain to me what this one stands for : Exten => 1234,1,read(numtodial,enternum,10,skip,1,10) that numtodial and enternum -- * HARAZ Tahar * *Engineering Student at the National Institute for Posts and Telecommunications (INPT) * * Phone: +212 6 78030050 E-

Re: [asterisk-users] asterisk-users Digest, Vol 85, Issue 10

Use service from sms providers like smsglobal.com, they have scripts to do that. On Fri, Aug 5, 2011 at 9:00 PM, wrote: > Send asterisk-users mailing list submissions to >asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >http://lists.di

Re: [asterisk-users] Asterisk users Calculation

It would be a great help to others(including me) if those using 1.8.X can provide some details on hardware configurations,features they have implemented on it and some sort of load testing results. Thanks, [SATISH] On Mon, Jun 6, 2011 at 6:28 AM, Sherwood McGowan wrote: > May I add...I still ha

Re: [asterisk-users] Asterisk users Calculation

May I add...I still have documented cases of asterisk 1.4.x running ulaw with no transcoding and running 2k+ concurrent calls on a CentOS 4(5?, fuzzy) machine with 2ghz CPU and 2gb ram Sent from my iPhone On Jun 5, 2011, at 3:02 PM, Steve Edwards wrote: > On Sun, 5 Jun 2011, Khaled W. Chehab

Re: [asterisk-users] Asterisk users Calculation

On Sun, 5 Jun 2011, Khaled W. Chehab wrote: 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding 3-And what is the number of simultaneous calls if I use d

[asterisk-users] Asterisk users Calculation

Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ

Re: [asterisk-users] asterisk-users Digest, Vol 83, Issue 3

> Letting a carrier use you as a carrier seems like quite a bad idea generally.. I think I would agree. :) > > _NXXNXX => Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here > get routed upstream > in the 'local' context instead of the other one? > So here is where the fin

Re: [asterisk-users] asterisk-users Digest, Vol 82, Issue 52

I listened to your Email using DriveCarefully and will respond as soon as possible. Download DriveCarefully for FREE from http://www.drivecarefully.com-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] asterisk-users Digest, Vol 82, Issue 27

Dear; In the extensions. conf, I have the following: exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}@Internal) So, I am writing the arguements of the Voicemail ( ) wrong? Regards Bilal > > Dear; > > > > Where I can find a new documentation for Asterisk > 1.8? > > > > Where is the wrong in th

Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27

On Sat, 9 Apr 2011, darin iv wrote: 0) Don't re-post the entire digest back to the list it came from. Posting 36k of cruft to ask 'How to change SIP port number?' seems somewhat 'newbish.' 1) Try Google. Try 'How to change SIP port number in Asterisk?' 2) Re-post with a new, relevant Subject

Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 27

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help On 4/8/11, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list s

Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 21

Thank you Paul. I have downloaded the code. How out-of-call messaging can be configured in the Dialplan? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 |

Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 12

On 4/5/2011 2:45 PM, Bill Michaelson wrote: > > > On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote: >> Message: 12 >> Date: Tue, 5 Apr 2011 13:36:21 -0500 >> From: Sherwood McGowan >> Subject: Re: [asterisk-users] Iptables configuration to handle brute, >> force registrat

Re: [asterisk-users] asterisk-users Digest, Vol 81, Issue 12

On 04/05/2011 03:06 PM, asterisk-users-requ...@lists.digium.com wrote: Message: 12 Date: Tue, 5 Apr 2011 13:36:21 -0500 From: Sherwood McGowan Subject: Re: [asterisk-users] Iptables configuration to handle brute, force registrations? To: Asterisk Users Mailing List - Non-Commercial Disc

Re: [asterisk-users] asterisk-users Digest, Vol 80, Issue 73

>> Back to the original question, for those of you using Fail2Ban, >> Does it take an unusually high amount of break-in attempts before > attackers are banned? >> I have it set to 5 attempts in fail2ban but usually, the attacker is able > to make over 100 attempts before fail2ban bans them. >> I've

Re: [asterisk-users] asterisk-users Digest, Vol 78, Issue 66

It may have gone to sleep. Chris Cooper Systems/Network Administrator EFC International 1940 Craigshire Blvd St. Louis, MO 63146 US Phone - 314-439-4325 Fax -314-439-4443 Mobile - 314-402-8912 - -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-use

Re: [asterisk-users] asterisk-users Digest, Vol 77, Issue 27

Le 13/12/10 19:00, asterisk-users-requ...@lists.digium.com a écrit : > Date: Mon, 13 Dec 2010 12:00:09 -0600 (CST) > From: "Jonathan C. Bailey" > Subject: [asterisk-users] Voice mail distribution - missing messages > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID

Re: [asterisk-users] asterisk-users Digest, Vol 76, Issue 58

We'll get this to you asap "asterisk-users-requ...@lists.digium.com" wrote: Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via

Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

Hi Paul, Thanks for reply. I have some mistake send compilation logs. i have written cdr_webservice.c module and its work on asterisk-1.6.2.6 version on production server. but i want to upgrade asterisk version. # make [CC] cdr_webservice.c -> cdr_webservice.o In file included from /usr/src/aste

Re: [asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

On 10-11-24 06:09 AM, RAJNIKANT VANZA wrote: > make[1]: *** [cdr_webservice.o] Error 1 > make: *** [cdr] Error 2 > What is cdr_webservice.o ? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.or

[asterisk-users] [Asterisk-users] asterisk-1.8.0 compilation error

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 7

On Thu, Oct 7, 2010 at 11:25 PM, Fazil Amaan wrote: > Hi, > > > I cannot get asterisk to start again after the g729 install failed. > > > kindly advise what's the problem. > > Thank's > > > -- > _ > -- Bandwidth and Colocation Pr

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 7

Hi, I cannot get asterisk to start again after the g729 install failed. kindly advise what's the problem. Thank's -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a liv

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 2

El 04/10/10 07:26, Dan Cropp escribió: > Date: Fri, 1 Oct 2010 18:40:40 -0300 > From: Rodrigo Lang > Subject: Re: [asterisk-users] AMI Originate > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain; charset="iso-8859-1" > > 3 milis

Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 2

Date: Fri, 1 Oct 2010 18:40:40 -0300 From: Rodrigo Lang Subject: Re: [asterisk-users] AMI Originate To: Asterisk Users Mailing List - Non-Commercial Discussion Message-ID: Content-Type: text/plain; charset="iso-8859-1" 3 miliseconds... 2010/10/1 Danny Nicholas [Dan Cropp] Th

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 63

On Sun, Aug 29, 2010 at 01:31:26PM -0400, David Cook (Asterisk List) wrote: > > I have 2 FXO channels from which I want to route incoming calls to > > different contexts in extensions.conf. I edited the context entries in > > dahdi-channels.conf and created matching entries in extensions.conf. >

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 63

> I have 2 FXO channels from which I want to route incoming calls to > different contexts in extensions.conf. I edited the context entries in > dahdi-channels.conf and created matching entries in extensions.conf. > One channel is routed to the new context as I want, but the other > channel is stu

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 58

On 8/27/10, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 58

On 8/27/10, asterisk-users-requ...@lists.digium.com wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 24

Le 10/08/10 19:00, asterisk-users-requ...@lists.digium.com a écrit : > Subject: [asterisk-users] How to determine which party hangup the > call? cause of Hang-up needed.? > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Message-ID: > > Content-Type: text/plain;

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