search for: asteriskstar

Displaying 8 results from an estimated 8 matches for "asteriskstar".

2006 Jan 09
1
how to adjust volume
how to adjust voice volume for sipura 2000 and cisco ata186?
2007 Jul 31
3
1and1 dedicated servers have been down for a few hours .
1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm
2006 Jun 14
7
open source sip softphone (Window OS version )
are there any open source sip softphone (Window OS version )?
2005 Jan 12
0
pass through mode
hi 1 does asterisk do full proxy to all calls, or it can be configured to do proxy signal only ? if yes, how to configure to proxy signal only? 2 G.729 codec license: do i have to buy license if * doesn't do codec conversion, just proxy the calls? Jeffrey
2005 May 13
0
Spawn extension -----what does this mean ?
for every call, * gives out : Spawn extension (default, 00xxxxxxxxx, 3) exited non-zero on 'SIP/201.50.117.161-081628e0' Spawn extension -----what does this mean ? how to avoid this ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050513/fd156a67/attachment.htm
2005 Sep 30
0
oh323 implementation 0.67 has call-id problem
I am trying oh323(version 0.67) , make call from sip UA to h323 gateway, can't get Call-id pass from sip UA to h323 gateway, h323 always gets call-ID sent from Asterisk as *. are there any configure to pass the correct call-id from sip UA to h323 gateway? or this is a bug in oh323 0.67? how about oh323 0.73 ? Mario On 9/29/05, Kanishka Somaratne <kani@technoportal.biz>
2007 Jul 02
1
DID providers in Toronto
hi Can anyone recommend a good DID provider offering numbers in Toronto ? ( 1 very stable 2 support porting numbers from Bell, primus, telus.. ) Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070702/cc39db9d/attachment.htm
2007 Jul 22
1
Wake-Up Call didn't work
I have setup wake up call in * ( 1.2crc1) following those instructions > http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP i can enter the time after dialing 77 , and i see there is wakeup files in /tmp but * nevers make the wakeup call when it is due , what can be the problem ? what shall i check? Mario -------------- next part -------------- An HTML attachment was