[Asterisk-Users] macro-record-cleanup in extensions.conf
hello Newbie, i try examples to understand asterisk. I have a pb with your macro macro-record-cleanup. the progress of the macro stops if the macro is execute on a hangup. I try many other configure with exchange of rules but it seems me that there is no execute after the first (or second) instruction after an hangup. i use two sip phones x-lite, one with a direct fwd config, the other is a phone on asterisk. The only situation where there is no stop is when the call is from the extern sip phone and this caller hang up so i have to use soxmix and gsm monitor to have correct records. xmix give me bad gsm records than i can't use. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] macro-record-cleanup in extensions.conf
hello Newbie, i try examples to understand asterisk. I have a pb with your macro macro-record-cleanup. the progress of the macro stops if the macro is execute on a hangup. I try many other configure with exchange of rules but it seems me that there is no execute after the first (or second) instruction after an hangup. i use two sip phones x-lite, one with a direct fwd config, the other is a phone on asterisk. The only situation where there is no stop is when the call is from the extern sip phone and this caller hang up so i have to use soxmix and gsm monitor to have correct records. xmix give me bad gsm records than i can't use. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] macro-record-cleanup in extensions.conf
Le sam 05/07/2003 à 14:59, Hervé Thibaud a écrit : I have a pb with your macro macro-record-cleanup. Sorry, i didn't look at bugs and now i have seen remark 5 in your extensions.conf I hope update soon regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [Newbie] SIP via fwd
Le ven 04/07/2003 à 10:21, Hervé Thibaud a écrit :
hello
to asterisk start
WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum
retries exceeded on call [EMAIL PROTECTED] for seqno 102
(Request)
with a call from x-lite [EMAIL PROTECTED]
WARNING {98311] : File chan_sip.c, line 2002 (__transmit_response):
Unable to determine sequence number from ''
and x-lite hang up
the second warning is new since morning and i have not made changes from
yesterday
thanks
it seems good now but i did renter inits in my x-lite phone directly
connected to fwd.pulver.com (like yesterday)
is x-lite or fwd crazy ?
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Re: [Asterisk-Users] record a conversation
and what the way to play records in the spool Le mer 02/07/2003 à 09:28, Matteo Brancaleoni a écrit : show application monitor in the cli Matteo. Il mer, 2003-07-02 alle 09:22, Hervé Thibaud ha scritto: hi is there a simple way to record a conversation with asterisk ? thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x-lite and audio
My bandwith is 64kb/s (ISDN BRI) and so i try to use x-lite which has many codecs. But i have no audio and i don't see where is the problem. the calls ring, the connexions are good x-lite - x-lite, x-lite - phone, there is no drop on the firewall (gateway+firewall+asterisk) and if i call with an external phone and exten default, i hear default messages from asterisk but not with x-lite. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x-lite and audio
i do't see what you ask, i go to proprties audio but there is no what you say. since i see there is no pb with x-lite on a win 98 but i have problem with x-lite on win 2000. in sip.conf i have put dtmfmode= info for there is pdb with GSM and inband on x-lite codecs are automatically place on GSM but i try to force on other codec and it is the same. So on WIn 2000 i have suppress x-lite do a new download and reinstall-it. Now the sound is there but there is some blanks if i try with default demo. But if i phone there i hear but nobody hears me. I go and try to reinstall it again. Le ven 27/06/2003 à 11:40, Angelo Sampietro a écrit : see if your pc has the auto turned on in the main audio control panel... ;) i use x-lite and work very well, which codec are you using? send the trace of sip debug command... regards, Angelo Friday, June 27, 2003, 11:14:18 AM, you wrote: HT My bandwith is 64kb/s (ISDN BRI) and so i try to use x-lite which has HT many codecs. But i have no audio and i don't see where is the problem. HT the calls ring, the connexions are good x-lite - x-lite, x-lite - HT phone, there is no drop on the firewall (gateway+firewall+asterisk) and HT if i call with an external phone and exten default, i hear default HT messages from asterisk but not with x-lite. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip NAT
Le dim 22/06/2003 à 15:02, Andy Powell a écrit : Andy, your update is http://www.automated.it/guidetoasterisk.htm isn't it ? yes, same place, just added some extra notes in there (they should be obvious) Yes my asterisk is on the internet gateway with sorewall (firewall) on it and my stations his behind the firewall. I have open ports (5060,5082) and others like my DROP LOGS from the firewall was writing and now thereis no drop with links sip and asterisk sessions it seems if i try with context=nocontext, nothing is right then when i have with context=sip my call rings the other side (one station on asterisk and the other directly to fwd.pulver.com proxy 192,246,69,247 port 5082) but the sound has many blanks. I try to connect directly the both to fwd.pulver.com and now i have a perfect sound but the question is perhaps links after opening session is only on the local networks with 10Mb/s. Once i can (when i'll have an external user to call) i'll try. -- pensée du jour : ... Que calor . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [HS] results testing asterisk with ISDN BRI look for help tounderstand configuring SIP with asterisk
configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP SJPHONE on 2 local stations windows (i don't succeed to use telephon SIP X-lite with asterisk) testing résults with asterisk SJPHONE local - IVR asterisk : OK extern telephon (analogic) - SJPhone : OK SJphone - extern telephon : OK extern telephon - SJPHONE : OK local network SJPhone -local network SJPhone (with asterisk) OK configuration sjphone : Use Local OuntBound Proxy (selected) Proxy IP address 192,168,0,1 port 5060 caller ID : SIP station@domain.dyndns.org (stations défined dans /etc/asterisk/sip.conf) I don't understand what i have to make and set to communicate with external telephons SIP (Sjphone, X-lite, MS messenger ...) Must i have a SIP provider subscription, how to integrate this subscription with asterisk The purpose i have is to keep control with asterisk to tape, redirect, establish conference ... with communicates I am swimming with (english) documentation anglaise and i understand very badly asterisk system, my knowledge in system software an linux is too low But with your patient help, i am sure i'll reach thanks to help me ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN BRI
Le mer 18/06/2003 à 20:00, Iain McWilliams a écrit : - Original Message - From: Hervé THIBAUD [EMAIL PROTECTED] result : --Executing Dial(Sip/roseau-6163,Modem/g1:BYEXTENION) in new stack -- Called g1:024076 -- Modem[i4l]/ttyI1 is busy --Everyone is busy at this time -- Hungup 'Modem[i4l]/ttyI1' -- Executing Congestion(SIP/roseau-aa5b, ) in new stack ( no audio, i don't hear the message) I had that problem, it was caused by stripmsd=1 in modem.conf. Commenting it out caused the correct number to be dialed and everything then worked correctly. Thanks, it's good now ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN BRI
hi -modem.conf :-- msn=240862922 incomingmsn=240866365,6365 device = /dev/ttyI2 group=1 device = /dev/ttyI1 ; ttyI3, ttyI4 -extensions.conf ;--- [sip] exten = _XX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076 result : --Executing Dial(Sip/roseau-6163,Modem/g1:BYEXTENION) in new stack -- Called g1:024076 -- Modem[i4l]/ttyI1 is busy --Everyone is busy at this time -- Hungup 'Modem[i4l]/ttyI1' -- Executing Congestion(SIP/roseau-aa5b, ) in new stack ( no audio, i don't hear the message) it is the same thing with the console exept i hear the message about congestion neverless an incoming call on 0240866365 is OK thanks for help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie: isdn4linux and BRI (FRANCE)
hi i would like samples examples to configure with isdn4linux i have hisax card : gazel and an ISDN(BRI) line (2 channels B and 1D) In fist time i'll use sjphone only Perhaps there is french people on this list who can help me to do first steps with Asterisk thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie : i try and test to use asterisk
I try to use X-lite with asterisk on intranet In sip.conf i have [general] port = 5060 bindaddr = 0.0.0.0 context = default [roseau] type=friend host=dynamic dtmfmode=inband context=sip [bambou] type=friend host=dynamic dtmfmode=inband context=sip and in extensions.conf [sip] exten = 1000,1,Dial,SIP/roseau exten = 2000,1,Dial,SIP/bambou i use X-Lite on windows in setup ; Display name : roseau user name : 1000 authorization user : same as user name Password : Domain/Realme : 192.168.0.2 SIP Proxy : 192.168.0.2:5060 ; i can't have this field empty i obtain in /var/log/messages when i try to call [handle_request]: Registration from 'roseau 'sip:[EMAIL PROTECTED]' failed for '192.168.0.4' Is anybody help me to start please regards (and very sorry for my english) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
