similar to: shared voicemail

Displaying 20 results from an estimated 3000 matches similar to: "shared voicemail"

2004 Aug 05
2
personal voicemail
Good day all IS there a way to personalise the voicemail message when you leave a message? Thanks Altus
2005 Jul 14
5
asterisk number of calls
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
2004 Sep 08
3
sendmail&hostname
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
2004 Sep 13
5
music on hold not strting
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
2005 Sep 15
2
cdr server
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
2005 Feb 08
2
bri dropping calls
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
2005 Jan 12
6
snom220
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
2004 Apr 05
1
sip no sound?
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
2005 Feb 10
1
Bri problem
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
2005 May 18
1
eicon fdc3
Good day all Did anyone get the eicon 4 bri working with asterisk and fedora core 3 Please Thanks Altus
2005 May 16
1
2 servers via PRI
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 <-->
2006 Nov 09
1
wip5000 roaming
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
2004 Aug 04
2
2 sip servers
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
2004 Sep 02
1
BRI&DDI
Good day all Is there anyone who has experience with ISDN BRI&DDI? I want to know if asterisk can distinguish between the different numbers? I want each number to play a different intro/answering message? Please Help Thanks Altus
2005 Feb 15
1
asterisk qualified
Good day all Is there any time of VOIP/SIP/asterisk qualifications or certificates? Thanks Altus
2003 Apr 29
28
H323
Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm
2005 Sep 01
3
Snom 360 and hints
PH> I am setting up a snom 360, and the lights come on OK when the mapped PH> user makes an outgoing call, but when the user takes an incoming call PH> the light does not come on. PH> I do not want to install the bristuff patch if possible. PH> (although I can see that with the devstate command I can make the lights PH> do whatever I want) First, ensure that the 360 has
2004 Apr 30
2
South-Africa
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
2004 Aug 13
2
not hangup
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
2004 Apr 29
2
conference & sip
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