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...d on various flavours of 1.8). Queue static and realtime (same outcome). Cheers Andy -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

Hello folks, I have a 20 channel fractional PRI and I would like to dedicate one of the lines for a Fax service (in and outbound). Is this possible with Asterisk and what conf would I need for that? Thanks, -JE

...ow to send * a semi-colon from a realtime database. I know that * uses the semi-colon as a 'seperator' - but I need to be able to use one in a command. I know I can use \; in the non-realtime configs, but this doesn't work in realtime. Cheers, Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended...

...ON STILL OPEN - IN CASE CALLER HUNG UP AFTER TIME-OUT)?MYSQL(Disconnect ${connid})) Any ideas? Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

...is added by the server you see - and it's now annoying me - let alone the rest of the list. Ta If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

...ber 2008 17:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [SPAM] - Re: [asterisk-users] [SPAM] - Re: [SPAM] - Dahdi,b410p and looping from 1 port to another - Email found in subject -Email found in subject - Email found in subject 2008/12/1 Andrew Thomas <andy at datavox.co.uk> Apart from you were dialling out on your inbound context and vice-versa. The best advice I can give now is to change to mISDN - as this is proven to work with v1.4 and v1.6. I wanted to try the Digium B410P card with Dahdi as those are now supported in 1.6.X. Hopefully, it would...

I have just the thing in PHP. Drop me a personal e-mail and I'll whiz it over. Andrew Thomas Technical Services Manager andy at datavox.co.uk DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Trevor Hammonds Sent: 16 July 2009 12:50 To: 'Asterisk Us...

Hello I haven't found an example on how to compare the value of a string variable with spaces in it, and the While loop below never exits: ========== extensions.conf exten => start,n,Set(MYVAR="Dummy value") exten => start,n,NoOp(${MYVAR}) ;BAD TOO ;exten => start,n,While(!$[${MYVAR} : "Some string"]) exten => start,n,While($[${MYVAR} != "Some

Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the

...com/pipermail/asterisk-dev/2008-May/033153.html]. But what does that mean in English ;)? Thanks If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

Sorry about this - testing this disclaimer problem :) -- If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK,

Is there a way for asterisk's voicemail to send an email (including voicemail attachment) to multiple email addresses? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/8908db5f/attachment.htm>

Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use

...first came in to the queue. So, it looks like it's trying to use the same unique ID again. If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is...

Hello, I'm using SIP realtime with MySQL DB. Is it possible to get the status of the SIP peer (free / calling) from this realtime DB ? If not, is there another way to obtain the call state of a SIP peer ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...where that the misdn debug message: -->> -->> -->> P[ 1] --> TON: Unknown -->> -->> may mean that the carrier did not recognize the caller id I set. Is -->> this true? -->> -->> -->> --- On Fri, 2/6/09, Andrew Thomas <andy at datavox.co.uk> wrote: -->> -->> > Use Set(CALLERID(num)=9999999999) instead of using -->> > CALLERID(all). -->> > -->> > Remember to set this BEFORE you Dial. -->> > -->> > -->> > -->> -----Original Message----- -->&...