similar to: Reminder paging for voicemail (?)

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.

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

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

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 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

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

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

I would like to know how to define the s extension when I have an incoming PRI line? Currently I have 5 incoming DID numbers. Four of these DID numbers I have going to specific extensions, the fifth number which is the main number I wish to go to a background sound where callers can hear message, get directory, dial extension, whatever. I see that the way to normally do this would be to

Hi guys As usual I am playing around with IAX soft clients. I was wondering with the various IAX clients, IAX client, DIAX, etc how's one park calls, transfer calls, etc since there is no flash key? Is there something I must do in the iax.conf or is it something I must do with the individual clients? Also, is it very difficult to use musiconhold with the IAX software clients? Thanks a

Hello guys Is there anyway for me to change the sounds that are presented in VoicemailMain? For instance, instead of it saying "mailbox", I would like it to say something like "please enter your mailbox number now". Is there a way for me to do this? I also noticed that when in some of the menus, even if I select one of the announced options it simply repeats the same menu

DOes anyone have any good suggestions as to good SIP or IAX clients for linux? I have set up and am currently testing asterisk in a controlled environment. I have gnophone running on one of my boxes but the gnophone site has been down. So I can't seem to fing the IAX and Ix-devel rpms or the gsm and gsm-devel rpms. So that prevents me from setting up gnophone on another box. I am

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?

Since the list community has done so much for me in my humble asterisk beginnings I have put together a simple little script written in php that serves as a paging reminder script. If anyone is interested in a copy of it contact me off list and I'll forward you a copy. The basics of the script are as follows: It queries an asterisk inbox of your choosing for the existence of a file that

Ok for those of you all up in a tizzy over my subject line, please don't take it literally because I'm certainly not saying that asterisk is the problem here. I just got a little nightmare problem that I need a bit of help figuring out. I installed an asterisk system a few months ago for a client, it has run almost flawlessly with the exception of a few small glitches. However, I got a

Hello All When I open up iaxcomm, it registers fine with the asterisk server. If I call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle for awhile (I haven't figured out exactly how long) it seems to miss calls. I can see the calls coming in on the asterisk server but they never ring through on iaxcomm. If I close it and reopen it, it takes calls again fine. I thought I

On my asterisk server I have placed valid email addresses in the voicemail.conf file as to allow mailbox users to receive message notification. My problem is it appears that the messages are attempting to be sent but instead they are bouncing with a fatal error message like the one below: (reason: 550 [PERMFAIL] yahoo.com requires valid sender) First of all this is not the whole message but

Hi guys I've set up a layered menu system on one of my asterisk servers where there is a main menu and several submenus; one for each department. Each menu plays a background intro message giving its various options. My problem is when I'm in the main menu and press the option to go to one of the submenus there seems to be a 5-8 second pause before it plays the background of the

Does anyone know of a way to transfer a call from the CLI? AJ