Re: [asterisk-users] Asterisk on Android?
I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. On Sat, Sep 3, 2011 at 9:45 AM, Daniel Tryba dan...@tryba.nl wrote: On Sat, Sep 03, 2011 at 01:53:54PM +0200, Gilles wrote: Do you want to run the entire PBX on the Android client or are you just looking for a IAX programm to be installed for receiving calls?! The entire PBX so I can have an IVR in the phone. I don't think you can access the radio of the phone (RIL) at this moment. So if you want to use the GSM itself you are out of luck. -- Daniel Tryba -- _ -- 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 -- _ -- 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
Re: [asterisk-users] 1.8 issues with Local Bridging
On Mon, Aug 8, 2011 at 3:57 PM, CDR vene...@gmail.com wrote: I encourage the developers to check this out http://forums.asterisk.org/viewtopic.php?f=1t=77692p=161590#p161590 I am calling from behind a NAT, and there is no way to force Asterisk to stay in the path. If the codec is the same as the outbound leg, it always does Remote bridging, but of course, creates a 1 way audio. I tried everything in the book directrtpsetup=no directmedia=nonat canreinvite=nonat and directrtpsetup=no directmedia=no canreinvite=no But it just behaves different than in 1.6.2 Any ideas how to make sure that the NAT works? -- _ -- 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 All That I had to do was to set: nat = yes directmedia=no directrtpsetup=no -- cobra2 -- _ -- 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
[asterisk-users] SMS within asterisk users
I'm trying to setup SMS among users of a single asterisk box.
I've set up asterisk10-beta to send SMS messages using MessageSend(). If I
manually set the 'from' variable. I can two way messages only between those
two extensions.
i.e.
[sms]
exten = 12000,1,MessageSend(sip:12000,12001)
exten = 12001,1,MessageSend(sip:12001,12000)
This works fine, But limits the users to only be able to text each other
back and forth. When I add a third extension matters are complicated.
So I tried to set up something that was a little more flexible. I thought
that I would be able to use ${CALLERID(num)} for the 'from' variable.
[sms]
exten = _X.,1,MessageSend(${EXTEN},${CALLERID(num)})
However, the CALLERID(num) variable is an empty string.
Is there another way to identify the extension that originated the message?
--cobra2
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Re: [asterisk-users] make calls from DID
Look, I don't mean to be a jerk. Most of the time you will need to add the info in sip.conf. You need all your info about your DID. Some of them don't allow you to dial out. So it all just depends on what is allowed. On May 26, 2011 1:56 PM, virendra bhati virbh...@gmail.com wrote: How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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
Re: [asterisk-users] make calls from DID
I was trying really hard to not say RTFM. cobra2 http://linuxindixie.info On May 26, 2011 8:24 PM, Matt Riddell li...@venturevoip.com wrote: On 27/05/11 5:27 AM, virendra bhati wrote: Hi, Thanks for replay ... How to get incoming call from DID to server ? You might want to read: http://downloads.oreilly.com/books/9780596510480.pdf Or you can support it and future books by buying a copy: http://oreilly.com/catalog/9780596009625 -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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 -- _ -- 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
