search for: asrerisk

When a connection is carried between two IP end points, does the Asterisk server incur CPU usage to pass the voice bearing circuit between the two end points? Is it possible to have Asterisk setup the call but hand off the voice traffic to be handled directly between the two end points? Thanks! Chip __________________________________ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month!

...? It's really hard for new users to keep the pace with CVS. So can you recommend more stable Asterisk versions, which are suitable for production environments? My needs is simple: standard switch (Call transfer, Parking...), Voicemail, Voice menu (Autoattendand). If someone have running Asrerisk on production system, please share the yours version info. best regards, Nguyen At 03:49 PM 8/12/2003 -0500, you wrote: >As far as Asterisk's stability goes: new features tend to be less >stable than older features, just like any software. If your user >base isn't requesti...

...to keep the >pace with CVS. > >So can you recommend more stable Asterisk versions, which are suitable for >production environments? > >My needs is simple: standard switch (Call transfer, Parking...), >Voicemail, Voice menu (Autoattendand). > >If someone have running Asrerisk on production system, please share the >yours version info. > >best regards, >Nguyen > > > > > >At 03:49 PM 8/12/2003 -0500, you wrote: >>As far as Asterisk's stability goes: new features tend to be less >>stable than older features, just like any sof...

with a h323 client over my gatekepper a call comes over asrerisk to my sip endpoint: == Spawn extension (sip-phones, 01634255122, 1) exited non-zero on 'SIP/0699073201-528d' -- Executing Dial("H323/ip$10.0.0.124:49638/18690", "SIP/0699073201") in new stack -- Called 0699073201 -- SIP/0699073201-dc61 is ringing...

Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

Using svn trunk, I was trying to see what the astdb entry in the sip.conf file does. Nothing :) I presume that it's meant to create an entry in the astdb. so, I have astdb=chan2ext/SIP/grandstream1=1234 in sip.conf But database show only gives *CLI> database show /SIP/Registry/706 : 192.168.0.200:5060:3600:706:sip:706@192.168.0.200:5060