Asterisk In The News

That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. >Asterisk 1.4 -- though I could probably upgrade. >Suggestions on how to make this happen? This "might" work - Exten => 1234,1,Dial( DAHDI/1/w#1&#2&#3,30,p) The ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

That way, if one of the phones is > off, or out of range, it doesn't go straight to that phone's voicemail. The problem is that, if one of the destination phones is diverting to voicemail, you won't know it's voicemail until it's answered - - by which time it's ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

But using asterisk-1.8 between revisions 281912 and 281982 it > > > breaks -- after a few seconds of the call, I lose audio from the > > > asterisk box to my soft phone, but not the other way around. ... Maybe 282891 (same change, but to the 1.8 branch)? Or did you fat finger the revision? -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- ...

i had some problems like this, but only when a snom phone transfered a > call. if you use asterisk 1.6.x this could also be an answer bug which > is allready been fixed. this bug cause some strange issues with moh and > wrong codec write formats. ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

This has the added advantage of allowing the web server to create sub directories on the monitor directory if you have more than one client using the same asterisk server -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

I had a quick look at this: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf And it looks like you might be better off creating your own macro for one touch recording and adding it to the features.conf as shown in this part of that web ... but using MixMonitor rather than monitor. Let me know how you got on with it as I think I'm going to be asked to do this in the next month or 2. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- ...

Asterisk Internet PBX: Re: Call Recording Questions. ... Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

But using asterisk-1.8 between revisions 281912 and 281982 it > > breaks -- after a few seconds of the call, I lose audio from the > > asterisk box to my soft phone, but not the other way around. This looks > > like one commit, but obviously I would like to know ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk- users ...

... (playback) Options: (num:333456,say) >AGI Tx >> 200 result=0 >Anybody have any ideas to work it out in agi playback ? Replace playback “num:334456,say” with “say number 334456”. Refer to. http://www.voip-info.org/wiki /view/say+number ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

Here are the description for asterisk and dahdi. Asterisk 1.6..2.9. Dahdi: 2.3.0.1. I have two issues with dahdi 1) I am not getting full callerid on my phones from sangoma card to asterisk users. if i am connecting analog phone directly then i am getting callerid ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

Asterisk Internet PBX: IAX2 calls getting rejected without a CAUSE CODE. How to debug this? ... Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http:// www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

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DragonWave CEO Peter Allen Presents Keynote Address at 4GWE in Los Angeles
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In addition, TMC produces ITEXPO; 4GWE Conference and M2M Evolution (in conjunction with Crossfire Media); Digium Asterisk World (in conjunction with ...

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Read http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf. But I'm guessing you knew that and are just after getting someone else to do the work... Just create a catch-all pattern to match anything your specific dialplan doesn't (what it ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

configuration of Asterisk by dan091 2010-07-29 09:18:29. Hello! The task is to configure the software ASTERISK to 5 internal lines inside the server through remote access. The software must support all functions. ... Asterisk PBX installation and configuration 2009-07-14 04:57:36. We need somoene with strong experience to install an Asterisk PBX and bridge to PSTN, set up VOIP providers, conferencing etc. ...

Asterisk Internet PBX: Re: Call Recording Questions. ... Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...

i had some problems like this, but only when a snom phone transfered a call. if you use asterisk 1.6.x this could also be an answer bug which is allready been fixed. this bug cause some strange issues with moh and wrong codec write formats. but without further ... Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ...