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My system is: Redhat 9.0 + Asterisk + ISDN4Linux + Teles 16.3 ISA Passive card I haven't sound card. Comunication between two SIP Clients is OK Comunication between PSTN and SIP Client is OneWay (i cant recive dtmf and voice from pstn) is it needed sound card ?

The best place for Astri Con 2006 would definatly be Omaha, Nebraska! ;) very central ...ah one could hope. __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

...4144 at mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 by 3rd call do you mean over the internet? if the answer is yes, then I wouldn't be surprised. another thing what codec are you using? Date: Fri, 7 Dec 2007 17:02:31 +0000 From: "Giovanni Miano" <giomiano at gmail.com> Subject: Re: [asterisk-users] asterisk performance To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <d75be1ca0712070902u6d25ee49w368eda405a32bce8 at mail.gmail.com> Content-Type: text/pl...

I'm curious (ok, well I admit it - it's for perosnal gain) what methods people are using to manage asterisk config files when they have multiple asterisk systems? Some sort of revision control such as cvs,rcs or subversion? A central 'config server' where you edit the files and then rsync them out? I have 5 systems to manage, and it seems that about the only common file is

...> Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ------------------------------ > > Message: 13 > Date: Mon, 27 Mar 2006 22:00:29 +0200 > From: "Giovanni Miano" <giomiano@gmail.com> > Subject: Re: [Asterisk-Users] Config File Management > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271200l3536e7cbp9ffe8a269c51b8ca@mail.gmail.com> > Content-Ty...

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if

...ry grateful, as it is turning into a nightmare! Cheers Tony -- Tony Mountifield Work: tony@softins.co.uk - http://www.softins.co.uk Play: tony@mountifield.org - http://tony.mountifield.org ------------------------------ Message: 17 Date: Fri, 11 Mar 2005 12:02:21 +0100 From: Giovanni Miano <giomiano@gmail.com> Subject: [Asterisk-Users] Asterisk + Call hangup To: Asterisk-Users@lists.digium.com Message-ID: <d75be1ca0503110302165f7e7c@mail.gmail.com> Content-Type: text/plain; charset=US-ASCII Scenario PSTN <-> ZAP CHANNEL <-> ASTERISK <-> SIP When i recive call i f...

...n/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > ------------------------------ > > Message: 22 > Date: Fri, 11 Mar 2005 14:16:17 +0100 > From: Giovanni Miano <giomiano@gmail.com> > Subject: [Asterisk-Users] Asterisk@home 0.6 + bristuff > To: Asterisk-Users@lists.digium.com > Message-ID: <d75be1ca05031105162038a069@mail.gmail.com> > Content-Type: text/plain; charset=US-ASCII > > How install bristuff in Asterisk@home ? > > i tried v...

...update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> -- Best regards, Alessio mailto:afoc@interconnessioni.it ------------------------------ Message: 26 Date: Fri, 18 Mar 2005 17:28:36 +0100 From: Giovanni Miano <giomiano@gmail.com> Subject: [Asterisk-Users] I4l + HiSax To: Asterisk-Users@lists.digium.com Message-ID: <d75be1ca050318082837548380@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 I need HELP pls! BRISTUFF: Bad Sound quality CAPI : PTP Mode dont supported mISDN : kernel...

God save this mailling list :) Which is the best settings for the best quality of Audio ? I'use isdn4Linux driver and SIP client but bad quality PSTN <-> Asterisk and SIP <-> PSTN

I use codec g711u or g711a but comuncation between two sip client (XTen lite) have bastard dalay of 0,5 - 1 second Is it normal ? Are there any configuration to solve problem ? Thanks all

Hi, I've a billion isdn0 card but i suppose than i cant patch kernel to use mISDN support because i've kernel 2.4 and patch on CVS i4l is for >= 2.6.8 How do it ? help me please Thanks

How install bristuff in Asterisk@home ? i tried version 0.2.Rca to last RC7k and when try to compile zaptel (after patched it) i've this error: make: *** [zaptel.o] Error 1

I need HELP pls! BRISTUFF: Bad Sound quality CAPI : PTP Mode dont supported mISDN : kernel is 2.4.x and not 2.6.x HISAX : PTMP ok, PTP incoming ok but in outgoing asterisk dont compose number(i listen dial tone and than i can compose number via dtmf) Asterisk CLI (g3 is group of Modem[i4l]/ttyI0 and ttyI1): Called g3: 3453444444