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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 asterisk-1.2.* don't provide this function...

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.

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 rwxr-x--- asterisk msg0000.txt rw-r----- asterisk msg0000.wav rwxr-x---...

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

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 installing ARI. Once I setup Asterisk to use a non root user and added 'apache' to the 'asterisk' user group everything worked perfectly. I also want to thank Dan for his patience and help in solving...

...fy to the /etc/zaptel.conf file: # hfc-s pci a span definition # most of the values should be bogus because we are not really zaptel loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 span=2,1,3,ccs,ami bchan=4-5 dchan=6 # fxsks=7 fxoks=7 according to http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+config+zaptel.conf&diff=13 I think that I should consider this card as a Wildcard S100U (hence the fxoks line) but if I type ztcfg -vvv i see : SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear...

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

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

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.

Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com

...ommercial Discussion" > <asterisk-users@lists.digium.com> > Message-ID: > <d75be1ca0603271204s4c6f9cat5d41624c1d0c7635@mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You can use dialout file > > http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+auto-dial+out&prev > iew=20 > > 2006/3/27, Steve Totaro <stotaro@asteriskhelpdesk.com>: >> >> SIPPS is one, I would like to hear of others. >> >> >> >> Of course you could create a dialplan that loops calls in and out. &...

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