similar to: Dialplane slip

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 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 Anyone doing asterisk in New-Zealand?

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 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 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 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 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 I'm in South-Africa,currently we are using openline4 cards for our pbx systems.Now we first need approval on the cards form icasa(a government standards) before we can use the card.The market here is very big for a system like asterisk.The only problem is to get a card approved(for a small company like us) its just about impossible. Now what I'm looking for is a company that

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've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

I'm trying to install spandsp But when I try to patch the Makefile it gives this error [root@pbxmilkshake apps]# patch < apps_makefile.patch patching file Makefile Reversed (or previously applied) patch detected! Assume -R? [n] y Hunk #1 succeeded at 41 (offset -6 lines). Hunk #2 FAILED at 67. is it ok to go on

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'm trying to configure 2 asterisk servers running sip to connect with each other with iax so both sip extensions can dial each other I'm using this webpage but I'm a bit stuck each time I try to dial the other server's sip extension it says trying and then just gives a busy tone.In asterisk it says it could not create aix channel and that all is busy at the moment Can

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'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 <-->