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 the option proxy 192.246.69.223 port 5060 but i think rapidely that i have to use proxy adress 192.246.69.247 port 5082 and i succeed to call me (and have rings) i try to do the same thing i sip.conf but i don't succeed where i have to write 192.246.69.247 port 5082 ? thanks --------------------------------------- pens?e du jour : ... c'est pas tout, mais va falloir s'y mettre ... ma?tre h thibaud
Hi, Have you opened the port 5060 on your firewall? Then you need to open ports used for RTP, in order to have audio too. What do you exactly want to do? To call a FWD user when you are connected to your Asterisk box? To be called by an FWD user? BR, Dan ----- Original Message ----- From: "Herv? Thibaud" <ht_asterisk@beltegeuse.org> To: "asterisk-users" <asterisk-users@lists.digium.com> Sent: Sunday, June 22, 2003 10:13 AM Subject: [Asterisk-Users] asteisk, sip & NAT> 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 the option proxy 192.246.69.223 port 5060 > but i think rapidely that i have to use proxy adress 192.246.69.247 port > 5082 and i succeed to call me (and have rings) > i try to do the same thing i sip.conf but i don't succeed > where i have to write 192.246.69.247 port 5082 ? > thanks > > --------------------------------------- > pens?e du jour : > ... c'est pas tout, mais va falloir s'y mettre ... > > ma?tre h thibaud > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > >
>----- Original Message ----- >From: "Herv? Thibaud" <ht_asterisk@beltegeuse.org> >To: "asterisk-users" <asterisk-users@lists.digium.com> >Sent: Sunday, June 22, 2003 8:13 AM >Subject: [Asterisk-Users] asteisk, sip & NAT>hi >My stations are behinds a firewall, the system is windows 2000 & 98, i >use sjphone >aterisk 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)This sound like the problem that I've been having this weekend. My setup is a Snom100 and X-Lite connected to my * box, and the same box is the NAT gateway for the devices. I could have external users call in no problem at all, but when I tried to call out I got about 1/1.5 seconds of audio and then all incoming audio died. the other end could hear me, however. It turned out to be the fact that * sending reinvite requests to fwd, which was then trying to connect directly to the snom100, and, obviously, failing because it's behind NAT. After much hair-pulling from myself and Andy, I stumbled across an unrelated post that pointed me to the 'canreinvite=no' option. I stuck this in the [fwd.pulver.com] section of the sip.conf file and magically, it all worked! Maybe, just maybe, it'll work for you too :) Jon
Le dim 22/06/2003 ? 12:18, Dan a ?crit :> exten => _8XXXXX,1,SetCallerID(${FWDUSERID}) > exten => _8XXXXX,2,SetCIDName(${FWDUSERNAME}) > exten => _8XXXXX,3,Dial(SIP/${EXTEN:1}@fwd.pulver.com) > exten => _8XXXXX,4,Playback(invalid) > exten => _8XXXXX,5,HangupIt is better now, i try to call an other sip user (an other station but with sjphone directly registered to fwd) the call ring but when i accept i have no sound i try the other way it is not better, i have a sound with many blank i have tested my sound card so that i registered radio on internet and registers are ok (on both) it suppose that demonstrate my sound cards are full-duplex. My connexion is an ISDN 64k/b and i suppose it's enough Andy, your update is http://www.automated.it/guidetoasterisk.htm isn't it ? i have an error when i start asterisk in : chan_modem.so (Generic Voice Modem Driver) -- Parsing "/etc/asterisk/modem.conf': Found -- Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulates Modem Driver) Warning(32771): File chan_oss.c Line 228 (sound_thread): Read error on sound device; Ressource temporarily unavilable --------------- I suppose this pb has matter with PSTN phone (tests was OK for me ) thanks