[Asterisk-Users] codec translation problem???

hallo, i experienced codec translation problems between my sip adapter and different sip providers i use a grandstream ht286 sip adapter and an asterisk server, the asterisk server is registered with two sip providers, sipgate and voipbuster, i would like to use ilbc o. g726 because of bandwith,

[Asterisk-Users] iax2 native transfer question.

hallo, maybe somebody could help me, i try to bring my asterisk server to native bridge two iax2 channels, on my old asterisk server (Asterisk CVS-v1-0-03/23/05-10:07:13) it is working, since i have installed the latest cvs code, asterisk stayes always in the middle of my iax clients? i

[Asterisk-Users] meetme problem

hallo, i just experienced that all meetme rooms share the same voice data, if i connect to 499, it could be heard in all other rooms (498,500, 501) could sombody help me, why does asterisk send the voice out of all rooms if i only connect to one? thanks for help, alex meetme.conf conf =

[Asterisk-Users] nativ bridging problem with ilbc!!

hallo all, could sombody please help me, i dont know why nativ bridging is not working when i choose the ilbc codec, with speex it is working,?? iaxcomm (ilbc) ---asterisk -- ( asterisk2 -- sip grandstream (alaw) ) \-native bridge--/ 1. if i use on

[Asterisk-Users] voicemail, busy does not work?

hallo, i tried to setup my extentions,conf like this but it never jumps to the busy part (102) asterisk always plays the unavail msg, also when i am connected to another iax channel (conferece room) and no more channel on my client is available. could sombody give me a hint what could be

[Asterisk-Users] dial script, send variable problem??

hallo, i trying to dial with a python script via the manager interface, it works ok but i would like to send a soud file name as a variable to the dialplan, so that i can call a number and send it a different soundfile i choose in my pyton script. the problem is, that the * dials correct and

[Asterisk-Users] manager interface, get callerid number??

hallo, does sombody know how to get the callerid from iax.conf ( callerid=name 1234) via the manager interface? Action: IAXpeers gives only the Name/Username but not the call number? any ideas how to do this? thanks, alex ___ Asterisk-Users mailing

[Asterisk-Users] LineJACK dial problem

hallo all, i have a quicknet LineJACK card and it seems to work ok, the only problem is, that when i use this in extentions.conf, exten = _[1-9]., 1, Dial(IAX2/krath:[EMAIL PROTECTED]/${EXTEN},50,Ttr) exten = _[1-9]., 2, Congestion it dials only 2 digits, e.g when i dial 1234 it dials only 12,

[Asterisk-Users] iax2 nativ bridge question?

is what my asterisk server shows during connection (unable to transfer): tahnks, alex Connected to Asterisk CVS-v1-0-12/02/04-14:33:02 currently running on snd (pid = 3792) Verbosity is atleast 5 -- Registered 'atuc' (AUTHENTICATED) at 82.82.238.221:30512 == Parsing '/etc/asterisk/manager.conf

[Asterisk-Users] iax2 nativ bridge question?

is what my asterisk server shows during connection (unable to transfer): tahnks, alex Connected to Asterisk CVS-v1-0-12/02/04-14:33:02 currently running on snd (pid = 3792) Verbosity is atleast 5 -- Registered 'atuc' (AUTHENTICATED) at 82.82.238.221:30512 == Parsing '/etc/asterisk/manager.conf

[Asterisk-Users] iax preferred codec question?

hallo, could somebody help me, i would like to select ilbc as preferred codec but dont disable gsm totally, i can only make a call with ilbc if i disable the gsm codec in iax.conf, if i enable gsm and make call to the same enpoint, always the gsm codec is choosen as audio codec. any idea

[Asterisk-Users] Unable to find a path from GSM to SPEEX ??

hallo, does anybody know, how to enable the the new iaxcomm client (with speex codec!!) to work with asterisk? i get a Unable to find a path error?? i have enabled speex in iax.conf, thanks for help, alex Nov 2 15:37:19 NOTICE[281616]: channel.c:1698 ast_set_write_format: Unable to find a

RE: [Asterisk-Users] Looking for a commercial version of an IAX2Softphone

At 19:19 26.09.2004, you wrote: Yes We have one... Its going to be released in October... We are in middle of Moving so once we move and get things settled we will be going online Unlike the diax and iaxcom and iaxphone ours is using DirectSound and has many more abilities. We are releasing

[Asterisk-Users] ./zaprtc.o: unresolved symbol ??

hallo, i tried to install the zaprtc.o module but get errors when i try to insmod it? i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see below lsmod), does anybody has an idea what is wrong? many thanks for help, alex snd:/usr/src/zaptelrtc# make cc -c zaprtc.c -D__KERNEL__

Re: [Asterisk-Users] ./zaprtc.o: unresolved symbol ??

At 10:05 21.09.2004, you wrote: hallo, i tried to install the zaprtc.o module but get errors when i try to insmod it? i use a th kernel 2.4.27, rtc is compiled as module but not loaded (see below lsmod), does anybody has an idea what is wrong? many thanks for help, alex sorry, i found teh

[Asterisk-Users] sip is logged out every morning?

hallo, my asterisk server is working fine, the only problem i have is, that every morning when i look at my sipgate page i am logged out, when i do a cli reload, everything is working until the next morning. so my question, how can i force my asterisk server to keep logged in at my sip

Re: [Asterisk-Users] sip is logged out every morning?

At 19:43 15.09.2004, you wrote: if you're using ppp, you can add an asterisk reload command in your ip-up.local how does it look like? do you meen over the manager server? any infos on the net? thanks, alex ___ Asterisk-Users mailing list [EMAIL

[Asterisk-Users] iax2 - peer 2 peer - asterisk?

hallo all, i am new to asterisk, just started my first tests and tried to build up a phoneserver with serveral iaxcomm clients running over iax2. my question is: now, it is possible to reach all phones over the asterisk server, but the media stream is always routed through the server. is it