similar to: no hangup

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 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'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

Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus

Good day all I have figured out most/basics of asterisk.I went with sip and made my own sip.conf and extensions.conf No I have 2 servers running sip in different towns.Both is connected with static ip so thats fine,but now. Lets say someone want to call someone else in the other town.How do I get asterisk to know,for instance sip extension 101 is on another sip server on a different ip. And I

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 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

Good day all We have Grand Stream BT-100 phones The transfer button work well, for blind transfer What the users want to do is, when a call comes in and asked to be transferred to another extension,for example 100,they 1ste want to speak to exten 100,then have the option transfer or not to transfer the call to this extension Currently they must pus "flash" for a new line speak to the

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 I want to know how to configure asterisk so that for instance if you press *5 it will pickup any ringing(unanswered) calls. My problem is this,at lunch time a bunch of people go out for lunch and when a call comes in it just ring and go threw the whole step. I want someone,whoever is left in the office to press something and answer any ringing phone? Can this be done? 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 want to tell asterisk that it should hangup a channel in a certain step For example: exten => s,5,Dial(SIP/302,25) exten => s,6,Hangup exten => s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help

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 I got my snom 220 phone so that it displays on the buttons if someone is calling that extension I just added "exten => 403,hint,SIP/403" in my dialplan But These lights only comes on if someone calls that extension,not if that extension is busy are a call is made from that extension Can this be done? Please Help 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

Good day all Did anyone get the eicon 4 bri working with asterisk and fedora core 3 Please 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 I cant get my WIP 5000 to roam 100% I have 2 access points, different SSI's I make a config1 and config2 on the phone, each for the different SSID's(A & B) Im standing next to A and I walk to B, but.the phone does not want to change its signal to B, it still keeps the bad signal from A If I power A down, it will switch to B, if I switch A back on and go stand next to