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hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

hi ---------modem.conf :---------- msn=240862922 incomingmsn=240866365,6365 device => /dev/ttyI2 group=1 device => /dev/ttyI1 ; ttyI3, ttyI4 ---------extensions.conf ;------- [sip] exten => _XXXXXXXXXX,1,Dial,Modem/g1:BYEXTENSION (Sjphpone) Call to : 024076xxxx result : --Executing Dial(Sip/roseau-6163","Modem/g1:BYEXTENION") in new stack -- Called g1:024076xxxx --

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

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

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

hello to asterisk start WARNING {98311] : File chan_sip.c, line 388 (retrans_pkt) : Maximum retries exceeded on call xxxxxxx...xxxxxxxxxx@192.168.0.1 for seqno 102 (Request) with a call from x-lite 38113@fwd.pulver.com 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

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

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