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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 a call and play a local file - Make a dial from one channel to another. I know asterisk can be stripped to exactly fit my needs. I would like to know if there

Hola Hace poco comence a utilizar R y tengo dudas como utilizar el paquete de tree saludos ruddi --------------------------------- Yahoo! Sorteos ¡Ya puedes comprar Lotería de Navidad! [[alternative HTML version deleted]]

...aring about exciting IPO and private placement offerings from life sciences companies interest you?</span></p> <p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><span style="font-family:Arial">The exclusive <b>Ruddy-Carlisle Biotech Infoline</b> service keeps you abreast of investment opportunities in the life sciences space. Just sign up for it once and get important information instantly delivered to study at your leisure. Our service is <b><u>100% FREE</u></b>! <b><spa...

...er("MySQL") dbcon <- dbConnect(drv, dbname="mydbName") dbGetInfo(db) } When the function is called, the dbGetInfo(dbcon) does return connection info but no connection object is present. I would be ver grateful for any guidance on where I have gone wrong. Thank you. Mark Ruddy PhD student Geography Dept., Royal Holloway (University of London) Egham, UK. [[alternative HTML version deleted]]

Hi, I setup an MeetMe conference. So, the admin user calls and enter the conference in talk/listen mode. (Options : dAaxs) Then other users call the same conference and enters in muted mode (options: dlmx) How can the admin user decide, when he is ready to let everybody speaks ? I didn't find such option in the admin menu. Thanks -------------- next part -------------- An HTML

Hi, First, thanks in advance. Some useful info: >version platform x86_64-unknown-linux-gnu arch x86_64 os linux-gnu system x86_64, linux-gnu version.string R version 2.15.1 (2012-06-22) I'm trying to use the table() function on a 2 column matrix that has 711 million rows (see below). However, it freezes. If I subset the matrix to be less than or equal

...that some time, the src field is the same as the dst field which is the extension. When does it happens. Here, we have 4 dgits extensions and most of the time the dst field is the extension and the src field is the 10 digit customer phone number. Do you know when does this happens ?? Thanks Ruddy Gbaguidi http://www.astblog.com

Hi All I want to be able to read some sip informations (from a database) like username, password, host and extension number and place a Dial from asterisk. So basicly, I want to dial sip extensions without modifying sip.conf each time. I don't know, in the dialplan, what the dial string should look like. I tried SIP/<username>:<password>@<host>/<exten>

Hi, Here is my scenario. I have a SIP call between two SIP endpoints. A calls B. During the ringing, B disconnects (network cable is unplugged). But A continue ringing forever (until the dial timeout) even if asterisk detects that B is disconnected with the qualify. Is there any setup or asterisk configuration I need to enable to have A close its call ? Note: when A is already talking with B,

...cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi:...

...ip: Deadlock on fax detection (Reported by Joshua C. Colp) * ASTERISK-28536 - Asterisk release candidates fail to build on FreeBSD (Reported by Guido Falsi) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28499 - translate: Cr...

...cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi:...

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

...Falsi) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28499 - translate: Cr...

...Falsi) * ASTERISK-23756 - setvar directive when used in template and a child of said template, results in duplicate variable names (Reported by Michael Goryainov) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28527 - ChanIsAvail() creates a CDR if unanswered=yes is set in cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28499 - translate: Cr...

...a","sans-serif"'>From:</span>= </b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] <b>On Behalf Of = </b>Ruddy Gbaguidi<br> <b>Sent:</b> Thursday, April 23, 2009 11:57 AM<br> <b>To:</b> 'Asterisk Users Mailing List - Non-Commercial Discussion'<br> <b>Subject:</b> Re: [asterisk-users] AGI PHP = script<o:p></o:p></span></p>...

...cdr.conf (Reported by Frederic LE FOLL) * ASTERISK-28525 - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up (Reported by Frederic LE FOLL) * ASTERISK-28511 - codec_resample: Bad sound quality when up sampling from SLIN16 to SLIN32 (Reported by Ruddy G) * ASTERISK-28499 - translate: Crash when frame does not have a "src" field set (Reported by Gregory Massel) * ASTERISK-25592 - chan_unistim: Clang Warning: variable sized type not at end of a struct (Reported by Alexander Traud) * ASTERISK-28488 - pjsip mwi:...

Hi all I have been looking at my asterisk CDR in the mysql database and some channel names are set to "**Unknown**" string. When I look at the code, everybody when calling ast_channel_alloc set a channel format like SIP/%s or Zap/%s Only app_voicemail.c doesn't when sending emails and I don't use voicemail. Why app_voicemail needs to allocate a channel to send emails ? And in