similar to: caller id?

Good day all I added the music on hold entry in vpb.conf and commented out default line in musiconhold.conf. Asterisk starts up with the default mp3 but as soon as I remove it and add my mp3 it just doenst start up and gives a broken pipe error? Please Help or advice Thanks ALtus

Good day all I had a look at the extensions.conf sorting http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting What I'm trying to do is route all my cellphone number threw a channel and all other calls threw the other 3 channels Cellphone numbers are 10 number,i.o.w XXXXXXXXXX. This is what I tried but it doesn't seem to work,please help Thanks Altus extensions.conf

Good day all I got my Music on hold to work but can I/how do i get music on transfer? Please help Thanks

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

Good day all We have a quad bri card,installed on fedora core1,downloaded the latest bri-stuff that download asterisk 1.0.3 and zaptel 1.0.3 and libpri 1.0.3 All installed and working.BUT after 5min+ of talking it just drops the calls? Any reason why? Please help Thanks Altus

Hi all ! Let's say you have about 6 small different companies sharing the same E1 / Asterisk server, and every company needs its own fax number. Since they don't really need fax machines, what would be the most cost-effective way to handle this (keeping fax-privacy at its best) ? Is there a way to configure Hylafax or sth & one modem behind an ATA-186 to email faxes to different

Good day all I'm still struggling with getting asterisk to hangup. If I make a call out threw my vpb pstn and the person on the other line hangs up 1ste it still shows the line is busy!Only after I hanged up it will show its still open? Why? Please le me know Thanks Altus

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

Good day all What is the amount of calls that asterisk can handle,SIP and from/to PSTN -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301

Good day all I'm using sip protocol and a openline4 card.If I dial out of the pstn and hangup a answered call it does not disconnect the connection.It shows there is still a call on the external phone I called but on my side its says i'm not connected.We are using x-ten soft phones Can someone please help me Thanks Altus

Goo day all I'm going to put a asterisk server running sip in at a client.The server is going to have a public ip so that it can talk to another server. My question is how do I secure asterisk/sip. I got a firewall only allowing tcp/udp 5060? I got sip to work with md5 What more? Please Advice Thanks a million

Hi All, Just a quick question, but I could really use some help on this one. I've got the CVS-Head of * installed and running, and am using a VoiceTronix OpenSwitch12 to connect to 12 analouge lines. I've got callerid activated by the Telco, and can get callerid using a std phone. However, using *, I always get an error 'Cannot decode callerid'. Does anyone know if I need to

Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to someuser@myname.co.za it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the

Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus

Good day all Is it possable to set asterisk up as a cdr server for other voip units We got a quintum dx here and its got a option to log to a cdr server on port 9002 Thanks Altus

Good day all I got my voicemail message working,thanks but now,keep in mind I'm using SIP We have,for example 4 people in our admin department.Each user has its own voicemail so that when their extension is dialed directly and not answered it gos to voicemail. But there is also a option to dial 3 for admin with will dial all 4 number in sequence.This I got working 100% but now I want a

Good day all Is there a page where you can doanload legal,good music on hold mp3? Thanks Altus

Good day all How is asterisk version I'm having problems with 1.0.0 and 1.0.1 If I'm starting asterisk it give problems with the modules saying unresolved symbols Please give some input Thaks Altus

Good day all So I've installed asterisk with my openline4 card and I've setup sip and I can do sip on the local network,we are using soft clients,x-lite. But... When a call comes in from the outside(PSTN) and the you dial the extension it forwards the call the the client and you see incoming call on x-lite,you accept he call....BUT there is no sound.It shows there is a call and you are

Good day all I've installed a few systems with quad/octo bri cards On these systems incoming numbers are ether the full number,example 12345657 or ether the last 4 digits,example 7654 But for some reason the latest installation incoming numbers comes in as extension "s"?? Is this something to do with the telecoms provider or a asterisk config? Please Help ore advice Thanks Altus