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2006 May 28
3
doubts about asterisk configuration from database
hi, I want to complete asterisk configuration from database(MYSQL),now I come across some doubts: 1. http://voip-info.org/tiki-pagehistory.php?page=Asterisk+configuration+from+database&diff2=3 says that "Dynamic 'friends' (Asterisk v1.0.*) and the number of options supported by this 'MySQL_Friends' system is currently very limited",at the same time I find
2005 Oct 12
8
parameters documentation
Another trivial question: Is there a "place" where all the parameters are documented ? In example (my example!) I would like to know the meaning of a lot of parameter that can be used in sip.conf, A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; context=xxxxx) but other are not (at least for me) i.e.: type = peer, friend insecure=very host=dynamic and so on.
2006 Feb 10
3
Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg0000.gsm
2004 Feb 03
2
busy tones
Hi When I call a phone with CAPI if the phone available I hear ringing ok but if the phone is busy I don't hear anything at all. Also, when I call a mobile phone and it is turned off I don't hear the operator voice answer me telling me that the request phone is turned off or unavailable. Any ideas? m
2006 Feb 08
0
ARI - Voicemail not showing - Problem solved!
Hi, Just wanted to pass on a fix that I found with the ARI recordings interface (www.littlejohnconsulting.com) for using a browser to access voice mail. It turned out to be a rights issue and group membership issue. I was planning on moving Asterisk to a non-root (http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-root&diff2=25) user but I had not done this prior to
2006 Feb 22
0
problem with SU100
Hi all, I am trying to add a TigerJet usb adapter to my asterisk installation I have the 1.2.4 asterisk bristuffed version, holding two zaphfc (ISDN) cards If I connect the TigerJet adapter to my linux box (Suse 10) i see: Bus 005 Device 001: ID 0000:0000 Bus 004 Device 003: ID 06e6:831c Tiger Jet Network, Inc. Bus 004 Device 001: ID 0000:0000 Bus 003 Device 001: ID 0000:0000 Bus 002 Device
2006 May 25
1
Way to disable codec in dialingplan
in a dialing plan.. extensions.conf can we enable or force a codec on specified npa.. EX: 514NNNNNNN,1,force(gsm) 514NNNNNNN.2. dial(sip/blah) ??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060525/e50cf3bb/attachment.htm
2005 Sep 15
3
MusicOnHold not working
Hi On my FC3 box with asterisk 1.0.9....MusicOnHold is not working. It starts and stops immediately... An unknow option mono comes...from where it is originating.?? As there is nothing written in .conf file. Console output is below: I am using mpg123 version 0.59r. Although I am able to play music with mpg123 but why it is on No-cooperation movement against asterisk ? Need help..any
2005 Sep 20
4
how to distinguish the "ringing" and "connected" for zap channel
I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.
2005 Jun 09
12
VOIP-INFO
Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com
2006 Mar 27
0
Question about Polycom 601 and expansion module.
Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"
2005 Feb 19
16
Snom phone hint exten question
Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with