similar to: Linux SIP/IX clients

Greetings everyone, I'm still a newbie, so please indulge me. I have set up an asterisk server on my RH9 boxes for testing. Instead of immediately launching into the hardware aspect of it I decided to go with 2 IAX clients (gnophone) which I placed on 2 other strategic machines within my LAN. Here's my problem, on one of the IAX clients (gnophone) I am able to do the asterisk demo

Hi, is there another IAX-CLient for Linux beside Gnophone? The problem with Gnophone is, when there is an incoming call. There is no pop-up-window or any sound that caughts my attention. So calls easily remain unanswerd. Or can I configure Gnophone in the desired way? Thx, Holger

Hey all...I'm trying to use gnophone to connect to my asterisk box behind my firewall..I thought I could just setup a tunnel with something like ssh host.com -L5036:asteriskserver:5036 and just change my gnophone to connect to localhost:5036 but I never see anything happen on the asterisk server. I'm even trying this on the same network just in case there is something funky with NAT.

hi all does anyone have the soundcore lib? I need it for a slackware installation of gnophone ... perhaps it's time to return it to the new cvs? -- Roy Sigurd Karlsbakk, Datavaktmester ProntoTV AS - http://www.pronto.tv/ Tel: +47 9801 3356 Computers are like air conditioners. They stop working when you open Windows.

We are in the prossess of ordering a ISDN (PRI) line from Verizon. There are a couple of questions on the application that I don't understand. I'm still a newbie here but we're taking the big plunge. If someone could give me some direction here I would be greatly appreciative. The questions on their app that I don't understand are: 1.) The Verizon Co will outpulse 10

For anybody who didn't know there is an article on asterisk in February's Linux Journal. AJ

I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes.

Can anyone give me presise instructions on how to compile cdr_mysql.so? When I initially installed asterisk on the system, I didn't have mysql installed. Since then I have installed mysql, created the database and table structure for cdr_mysql and placed the appropriate settings in the cdr_mysql.conf file. However when I do a show modules at the CLI I cannot find cdr_mysql.so.

Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ

Hi U can visit the http://iaxclient.sf.net for some opensource underdevelopment softphones. Take Care Obaid Amin Syed >From: Chris Albertson <chrisalbertson90278@yahoo.com> >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: [Asterisk-Users] Good W2K softphone >Date: Fri, 3 Oct 2003 23:00:13 -0700 (PDT) > > >I haven't

Is there a way to configure voicemail to do reminder paging? I would like to configure some voicemail boxes to send an e-mail message to a pager every 10 minutes until the message is retrieved.

Greetings all, As most of you probably know from my previous questions on the list, I'm still in the newbie category. My question today is pretty brief, as I told you all a few weeks ago I ordered a PRI from Verizon. I understand that there is a "B" channel that comes with this. The question is just what can I use this "B" channel for and how??? Thanks AJ

In taking a cursory browse at Vonage's site today, I noticed they are now offering a soft phone. Has anyone had any experience using this? And does this possibly open new opportunities for using Vonage with Asterisk? Just thinking outloud on the list, soliciting thoughts and experiences from others. AJ

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

Hello all I just checked out the latest zaptel/zapata/libpri/asterisk/asterisk-addons from the cvs and ran through the entire make procedures. Everything seemed to go fine however now when I attempt to start asterisk, it says ok but it seems to be immediately crashing. The following messages are displayed in my /var/log/asterisk/messages file for the time right around the crash: Dec 29

In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be

Hi! I was trying to use gnophone with asterisk, but I can't make a call (It just get the a answer of "REJET"), but I can register an everything. Anyway, I decided to move to the cvs version of gnophone, so I checked out EVERYTHING from cvs.digium.com (yes, a cvs -z7 co .). I installed libiax2, gsm (the one that was inside gnophone), and got gnophone to start compiling. But

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Hi guys, Having a bit of a problem trying to get conference bridges working. In my meetme.conf file I have the following line [rooms] conf => 6000 In my extensions.conf file I have: exten => 1000,1,MeetMe,6000 My problem is that when I dial into extension 1000 it is telling me "this is not a valid conference number". Can anybody telling me what I'm doing wrong here?

Can anyone explain to me the difference in zaptel.conf and zapata.conf? I'm trying to get a real clear understanding of them but its getting a little murky in places. I will be setting up a PBX running asterisk with 2 T100P cards. I will be bringing a 23 channel PRI into one card and connecting the other card to a Nortell 24 channel FXS channel bank. As I understand it zapata.conf is