[Asterisk-Dev] Can I play announcement to caller and callee when I answer the ca ll?
I have put in the following line in my extension.conf: exten=1234,1,dial(SIP/1234, rtA(hello_i_am_tony)) I do hear "hello_i_am_tony" wave file in my 1234 phone. Anyway to modify it to hear on both my phone and the one who is calling in? Many thanks, Tony ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Interesting fix on CVS
On Fri, 2004-08-20 at 13:40, [EMAIL PROTECTED] wrote: > Update of /usr/cvsroot/asterisk/channels > In directory localhost.localdomain:/tmp/cvs-serv27553/channels > > Modified Files: > chan_vpb.c > Log Message: > / check so as not to enable loo-drop on FXS > > > ??? A loo drop on FXS? Doesn't sound too nice! I dropped the p ;) it was ment to say loop-drop :) > > Matt Riddell > ___ > Asterisk-Dev mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-dev -- Ben Kramer <[EMAIL PROTECTED]> ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Interesting fix on CVS
Update of /usr/cvsroot/asterisk/channels In directory localhost.localdomain:/tmp/cvs-serv27553/channels Modified Files: chan_vpb.c Log Message: / check so as not to enable loo-drop on FXS ??? A loo drop on FXS? Doesn't sound too nice! Matt Riddell ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Andre Bierwirth's ring state patches for SNOM 200 programmable buttons
I have the programable button led's working properly on my snom 200 except they don't flash during a ring event. I found a post by Andre Bierwirth saying he had a patch that he submitted but didn't make it into CVS. I would like to get a copy of that as a starting point to implement button flashing on ring. I have read through all the code and it looks like it should be pretty easy. Does anybody have a copy of this patch? Or have Andre's e-mail address? David Hinkle Sr Linux Engineer DerbyTech Inc. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Preventing Asterisk from codec conversion when reinviting
Hi, I'm new to Asterisk and SIP, but have experience with developing VoIP apps in general and H.323 apps in particular. I would like to fix a problem we're having with Asterisk performing codec conversion in the following situation: EP1, preferred codec order aLaw, G.729 EP2, preferred codec order G.729 EP1 places call to EP2, we see two call legs: EP1 to * is aLaw * to EP2 is G.729 (so Asterisk is performing codec conversion) In this scenario we'd like to use G.729 for both call legs with the media stream bypassing Asterisk. Thus reducing the CPU load on the * machine and the need for additional G.729 licenses. We would like Asterisk to use the codec settings of EP2 while sending a reinvite to EP1. Currently Asterisk uses it's own codec settings as defined in sip.conf. Since we are using Asterisk together with SER, our endpoints are not defined in sip.conf so we cannot put information about them in it. Can someone tell me where to start developing? Which files, functions? TIA. -- Andreas SikkemaRits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] Asterisk channel variable access: a proposal
Holger, I'm not suggestion that we plug only one scripting language in. Quite the contrary, I'm suggesting that there should be a robust structure to plug an arbitary programming/scripting language into Asterisk. The goal of the working group would be to regularize the information so it's available to all APIs, regularize the APIs so they are the same no matter how one hooks into Asterisk, and to create hooks into integrate an arbitrary language into Asterisk. Sound reasonable? Thanks, David Holger Schurig wrote: The end goal would be the ability to plug a Python or Perl interpreter into Asterisk and have the Python or Perl script run in parallel with a Dial Plan rather than being called out from the Dial Plan. As much as I love Python: it would probably not be useful, because in Python there is the equivalent of the Big-Kernel-Lock in Linux 2.4. So, do it in Perl or Lua (much more lightweight). ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] Banned from the channel?
[12:40:32] NickServ [EMAIL PROTECTED]: Password accepted - you are now recognized [12:40:33] You have set user mode +e [12:40:33] [NOTICE MemoServ]: You have no new memos [12:40:42] #asterisk: You're banned from that channel I go on holidays for 3 weeks and look what happens!! I hear rumours that all Turkish IPs are banned?!? How do I go about getting the ban lifted for my office and home (Both have static IP addresses)? Cheers -- Peter Nixon http://www.peternixon.net/ PGP Key: http://www.peternixon.net/public.asc pgpUWLJ2gVWuz.pgp Description: PGP signature
