Re: [Asterisk-Users] How can I know if a user is busy or not connected?

You're right. Just some practical examples which really neeeds this feature. - using ATA, when the phone is off-hook with dialtone, if someone call that line it sounds busy, even the call waiting is activated. - when a phone is not registered, iy appears to be busy, which is not the right

[Asterisk-Users] Chan_h323.so native?

Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? Thanks. Regards, Steven Thomas ___

[Asterisk-Users] Questions regarding CDR's

Hi- Two questions: (a) Does the CDR SQL module log both incoming and outgoing calls, or just outgoing only? (b) If I enable the CDR SQL module (to use mysql), does it disable the text logging at the same time? Thanks Scott S ___

Re: [Asterisk-Users] Questions regarding CDR's

Hi- Hello, Two questions: (a) Does the CDR SQL module log both incoming and outgoing calls, or just outgoing only? Asterisk is a PBX, so this question have no sense, because for asterisk you don't have outgoing and incoming calls. What is coming for a port is outgoing for another. (b)

Re: [Asterisk-Users] Chan_h323.so native?

Hi - does chan_h323.so come standard in the cvs checkout of Asterisk? or do you have to patch or add it in to the source directory structure before compiling? Can / and maybe how can this be added after? H.323 is coming into asterisk cvs, and i think is trying to find if you have openh323,

[Asterisk-Users] Great concept but a few issues unresolved

The past week or so I have been experimenting with Asterisk and overall find it to be a nice software suite, although I have encountered some problems, and have found almost no documentation (For example in sip.conf I needed the commands fromuser= and fromdomain= and only figured out this

[Asterisk-Users] music on hold help

I've only gotten to hear Music On Hold once. I am running the right version of mpg123 .059r as downloaded from mpg123.de and compiled locally So I'm looking for any help on getting this up and running. I can see on the console that the SIP phone is placing the call on hold, but there is no

[Asterisk-Users] Voicemail cliping digits via sip

Hi list, I've got a testbed running with the following config: 1. RH 7.3 linux machine 2. 2 Grandstream phones 3. 2 XTen soft clients When I dial voice mail I have a problem. here is the flow. 1. dial 8500 (the exten for voice mail) 2. enter 2600 via touch pad on the grandstreem 3. entire

Re: [Asterisk-Users] music on hold help

An update: Having the grandstreem place the call on hold yields no audio on the other end (a xten soft phone) having the (xten softphone) place the call on hold causes hold audio to come out of the grandstreem. the console (ast) shows the calls being place on hold by either phone On Sat, Aug

Re: [Asterisk-Users] Voicemail cliping digits via sip

Have you tried to CVS recently? I had some similar issues and mark fixed those.. but I wonder if they could be related to this. bkw On Sat, 16 Aug 2003, John Brown wrote: Hi list, I've got a testbed running with the following config: 1. RH 7.3 linux machine 2. 2 Grandstream phones 3. 2

Re: [Asterisk-Users] Voicemail cliping digits via sip

installed about a week ago. I'll go pull to see if there is more recent code thanks On Sat, Aug 16, 2003 at 09:53:54PM -0500, Brian West wrote: Have you tried to CVS recently? I had some similar issues and mark fixed those.. but I wonder if they could be related to this. bkw On Sat,

[Asterisk-Users] H323/SIP gatekeeper

Hello List, Does asterisk H323/SIP allowes me to conditionally use diff gatekeeper to route the call ? e.g. for the call to germany, I want to use gatekeeper1, and for the call to UK, I want to use gatekeeper2. if yes, where and how to specify in these .confs file ? Thanks, George Lin