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...e 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From: "Julian J. M." <julianjm@gmail.com> Subject: Re: [Asterisk-Users] Problem: ringtones stop unexpectedly when multiple channels are dialed To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <a60dba290604011059k242fd5c6lf9398b4228c624e3@mail.gma...

Good Morning, I have a weird situation, I'm testing with Xlite as SIP phone (is it any good ) and dialing an extension (also Xlite on same LAN) and I'm getting a real bad echo on the dialer's side and a not so bad one on the receivers side. Has anyone had something like this ? Aparently one should only get echo when you break out onto a telco network ? -------------- next part

...February 2005 10:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] SIP echo on LAN > > Check your soundcard controls... maybe it's recording "what you hear" > or PCM, thus sending it again to the other party. > > Julianjm. > > > On Mon, 21 Feb 2005 09:47:55 +0200, Nic le Roux <nicl@i-procc.za.net> > wrote: > > > > Good Morning, > > > > I have a weird situation, > > I'm testing with Xlite as SIP phone (is it any good ) and > dialing an > > extension...

Hi my name is Will Velez. Does Asterisk support E164? Thanks

There is a proble to put an H323 Asterisk server behind an iptables firewall? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/e0627e85/attachment.htm

Hi, I've finally given up on trying to fax over my Digium TDM400 card. I've found that fax over VoIP is quite more reliable (at least I can receive the faxes). My ITSP supports G729 and alaw/ulaw. As I won't be receiving faxes everyday (just ocasionally), i pretend on using g729, unless a fax is detected. Is there any way to force asterisk to make a reinvite, and swap the codec on

Hello there, I am playing around with Asterisk the first time and it really looks great. ;-) However, I have one problem: Any SIP device can connect to my PBX. How can I requre logon for SIP users and deny access in the case of wrong or missing credentials? Thanks Florian

I was just wondering if there are any problems using the latest FreePBX with SATA Raid 1 using hardware assisted software Raid like most modern chipsets support? I know that Digium and FreePBX were not recommending it awhile back but I think that was based on 2.4 Kernel and Digium hardware issues. I am assuming it is not a problem with the latest FreePBX using 2.6 Kernel and Sangoma cards? Any

> Still trying to get NAT working. Try adding a canreinvite=no. Nabeel

I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually. Both cards works fine Any idea ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M. Inviato: luned? 27 giugno 2005 12.04 A:

Hello, This is a little off-topic. I have an Ericsson FCT f251m, according to the specs it supports call signalling through polarity reversals and loop break, but it's currently disabled. On my PSTN line, my TelCo does send polarity switchs to signal answer and hangup (answeronpolarityswitch=yes and hanguponpolarityswitch=yes in zapata.conf), and asterisk detects it alright. I've been

Hello, I've been fighting one-way-audio issues with asterisk and SIP extensions for some time..., and I want to share with you my findings ;) My setup: * 1 ADSL router (Zyxel) * 1 Asterisk box with private IP, and interesting ports forwarded to it. * Several extensions, some local some remote The problem: * External extensions behind double nat don't get audio when they