search for: 20realtime

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

...ip.conf register command to the RealTime format. It seems that sip_friends table setup doesn't allow for such thing to happen. So far the only way I see to do this is dumping the sip_friends table setup in favor of Asterisk RealTime Static ( http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static) which seems to be quite an ugly solution. Am I missing anything? ScriptHead -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/a5f33d17/attachment.htm

...=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found here:<http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static> * NOTE:* You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. This would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP client is added. Do I hav...

...romuser=... fromdomain=... secret=... host=... port=5060 nat=yes canreinvite=yes qualify=no disallow=all allow=ulaw dtmfmode=rfc2833 insecure=port,invite context=incoming-sip Is this correct? What's throwing me off is this statment found @ http://www.voip-info.org/wiki/index.php?page=Asterisk%20RealTime%20Static: NOTE: You can only store a static config OR a RealTime config. You cannot, for example, store sip.conf and use sipfriends via RealTime. If I am correct, it would suggest that I'll have to do a reload when I add a DiD, but a reload won't be necessary if a new SIP...

...#39; == Everyone is busy/congested at this time (1:0/1/0) As you can see none of them go through. I have another Asterisk server connected with IAX2 that does work. To that server I can dial any extension without problems. I used http://www.voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20IAX to configure my * server. Any ideas? All three providers were working before I changed to Realtime IAX and I made sure to put all the necessary information into the Database. -- Carlos Chavez Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext...

I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the