Re: [Asterisk-Users] asteisk, sip & NAT
Hervé Thibaud wrote: Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit : ... 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. It's like i thought, the sound is nasty with many blanks when i try a connection on internet and ISDN bandwith on one channel 64kb/s is not enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and use a bandwith of 64kb/s so. Is anybody that has a free or a very cheap solution (it's to try asterisk) to have an IP phone hardware or software with G.723.1 codec Other thing, I would like to try X-Lite but i have pb with registration, i don't know how to write settings, i have an error (for example) : File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau " failed for '192,168,0,1' I try many form of settings but didn't succeed. X-Lite support speex (SPX on the X-Lite screen). This is a very impressive codec that will even allow you to talk with someone over a _modem_ with a little bandwidth to spare for other stuff. Obviously the modem adds a degree of latency but it's still impressive. So with the latest technology, etc we've managed to get a _voice_ conversation to travel over a standard phone line. ;-) But seriously, it is impressive. Regards, Andrew Radke ,-_|\ [EMAIL PROTECTED] mobile: +61 412 798593 / \ Member, System Administrators Guild of Australia \_,-._* o "I didn't know it was impossible when I did it." ___ 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 16:18, Hervé Thibaud a écrit : ... > 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. It's like i thought, the sound is nasty with many blanks when i try a connection on internet and ISDN bandwith on one channel 64kb/s is not enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and use a bandwith of 64kb/s so. Is anybody that has a free or a very cheap solution (it's to try asterisk) to have an IP phone hardware or software with G.723.1 codec Other thing, I would like to try X-Lite but i have pb with registration, i don't know how to write settings, i have an error (for example) : File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau " failed for '192,168,0,1' I try many form of settings but didn't succeed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip & NAT
--On Sunday, June 22, 2003 14:38:20 +0200 Hervé Thibaud <[EMAIL PROTECTED]> wrote: 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 Probably means some other program has already locked the sound output on your * box. You can put: noload => chan_oss.so in /etc/asterisk/modules.conf if you don't need OSS sound for *. Iain ___ 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
Re: [Asterisk-Users] asteisk, sip & NAT
>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) HTH Andy ___ 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 12:18, Dan a écrit :
> exten => _8X,1,SetCallerID(${FWDUSERID})
> exten => _8X,2,SetCIDName(${FWDUSERNAME})
> exten => _8X,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
> exten => _8X,4,Playback(invalid)
> exten => _8X,5,Hangup
It 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
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Re: [Asterisk-Users] asteisk, sip & NAT
>- Original Message - >From: "Hervé Thibaud" <[EMAIL PROTECTED]> >To: "asterisk-users" <[EMAIL PROTECTED]> >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 ___ 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 10:16, Dan a écrit : > 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. yes i have opened 5060 & 5082 and it's OK if i call directly without asterisk > What do you exactly want to do? > To call a FWD user when you are connected to > your Asterisk box? yes & it isn't ok > To be called by an FWD user? it's OK (i have mince sound but isn't matter) thanks > > --- > > pensée du jour : > > ... c'est pas tout, mais va falloir s'y mettre ... > > > > maître h thibaud > > ___ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asteisk, sip & NAT
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: "Hervé Thibaud" <[EMAIL PROTECTED]> To: "asterisk-users" <[EMAIL PROTECTED]> 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 > > --- > pensée du jour : > ... c'est pas tout, mais va falloir s'y mettre ... > > maître h thibaud > > ___ > 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] 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 --- pensée du jour : ... c'est pas tout, mais va falloir s'y mettre ... maître h thibaud ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
