[asterisk-users] Asterisk crashes

this problem? Thank you! -- Sincerely, Elman Efendiyev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP-H323 calls without proxying RTP

asterisk. Same question about H323-H323 calls I'm using NuFone Network's H323 cahhel Thanks -- Sincerely, Elman Efendiyev PROTECH INC. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

RE: [asterisk-users] Digium TE110P

light -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of rivoli.durand Sent: Friday, 16 February, 2007 15:52 To: asterisk-users Subject: [asterisk-users] Digium TE110P Hi I am currently installing a TE110P. SUSE10

RE: [asterisk-users] Trixbox vs. Custom install

coise on your own. -- Sincerely, Elman Efendiyev PROTECH INC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefano Corsi Sent: Monday, 12 February, 2007 18:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users

[asterisk-users] TE110P working hardware configurations

) configuration here. I think this info could be very useful for many people and we can make detail compatibility list in wiki. There is some hw examples list in wiki already but I think not enough detailed. Thank you! -- Sincerely, Elman Efendiyev PROTECH INC

[asterisk-users] Digium TE110P

at e800 Memory at fe6fd000 (32-bit, non-prefetchable) Capabilities: [40] Power Management version 2 I appreciate any ideas. Thank you. -- Sincerely, Elman Efendiyev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)

. Are anybody currently working on T.38 support for * ? I don't mean T.38 support on zap interfaces, just passing T.38 packets trouth asterisk -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent

RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)

comparing to voice call, you can (and probaby will) have troubles with fax transmission (quality, line drops etc) but not with * complain about codecs. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer

[Asterisk-Users] H.323 call problemm (no sound)

channelsOpen = 0 -- Unknown [111.222.111.222] has cleared the call == Spawn extension (test, , 1) exited non-zero on 'SIP/234-d01b' == H.323 Connection deleted. Could anybody point me what I'm doing wrong Thanks. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED

RE: [Asterisk-Users] H.323 call problemm (no sound)

] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Monday, September 20, 2004 5:33 PM

RE: [Asterisk-Users] How would you handle a fax without T.38 orG.711uLaw?

Isn't it possible to use T.38 for interconnecting hardware gates supporting T.38 with asterisk using SIP REINVITE? I'm not shure but but think its's might be possible because after reinvite traffic goes directly from one gate to anotger, not over Asterisk -- Sincerely, Elman Efendiyev [EMAIL

RE: [Asterisk-Users] How would you handle a fax withoutT.38orG.711uLaw?

And what about using same codecs for asterisk and endpoints? Lets say G.729. Yes, it needs license but while G.729 is industry standart de-facto I thing most of us need to use it anyway -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto

[Asterisk-Users] Unknown RTP codec 72 received

insecure = no username = 332 secret = xxx host = dynamic nat = yes dtmfmode = rfc2833 callerid = 332 Could somebody tell me whay this Unknown RTP codec 72 received means and how to fix it? Thanks. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] ___ Asterisk

RE: [Asterisk-Users] Unknown RTP codec 72 received

Thanks for the hint Eric, but yes, before sending a message to the list I checked google and wiki and NO - I didn't find an answer/solution/any info on this subject. There was couple of the same questions on the list but none of them answered -- Sincerely, Elman Efendiyev [EMAIL PROTECTED

[Asterisk-Users] T.38 pass-thru

troubles with this setup (no faxing) while two gates conneted directly with same network path without Asterisk able to faxing without problemms Where I'm wrong? -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL

RE: [Asterisk-Users] T.38 pass-thru

Jusc couldn'n transmit faxes trouth asterisk. It just hangs up when starting a fax transfer If U will do some experiments with it I would be happy to hear any reslts/info Thanks -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL

[Asterisk-Users] FXO/FXS with T.38 over SIP

Hello, Could anybody suggest cheap FXO/FXS devices with full T.38 support over SIP? I found a number of devives with declared H323/SIP and T.38 support but some of them supports T.38 only with H323, others have buggy T.38 -- Sincerely, Elman Efendiyev [EMAIL PROTECTED

[Asterisk-Users] Phone numbers for testing

Hi, Is there any phone numbers with answering machine wich can record my voice and play it back to me? It would be very helpful for asterisk testing, but im not shure such service exsists at all. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED

RE: [Asterisk-Users] Phone numbers for testing

Yes, something like this. I would like ability to call such system via PSTN to test my * setup and my ITSP termination Something like this: My* -- MyITSP -- PSTN -- System with extensions you tell about -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL

[Asterisk-Users] D-Link DG-104SH H323 problemm

and netmeeting (without asterisk) its ok in both directions Could anybody explain what I'm doing wrong? Thanks -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk

RE: [Asterisk-Users] Busydetect problems

In my case this was because PSTN line didn't reverse polarity for disconnect notification and busy tone level was to low for asterisk to detect it. I solved this problemm by increasing rxgain (up to 4 in my case, your may de different) -- Sincerely, Elman Efendiyev [EMAIL PROTECTED

[Asterisk-Users] PBX functions and different channels grouping

! -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [Asterisk-Users] Please help I fear I have missed something very important! but what?

Looks like you missed 's' extension for incoming calls You need something like this in extensions.conf exten = s,1,Answer exten = s,2,Dial(SIP/1001,20,t) See sample of extensions.conf in asterisk distribution (make samples if you didn't install samples) -- Sincerely, Elman Efendiyev [EMAIL

[Asterisk-Users] IP Soft Phone with FAX

a solution for my problemm? Thanks. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com