search for: severna

...What other Cisco routers can act as SIP gateways today? Thanks, Dave ===================================================================== David C. Troy [dave@toad.net] 410-384-2500 Sales ToadNet - Want to go fast? 410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net

I need to get your thoughts on something.. :) I am trying to create a system to process the CDR call logs for department accounting.. I think there are two ways of doing it.. Either I can create an AGI that will run on the "h" extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately..

...t;pick up the damn phone" push message would be cool too. :) Dave ===================================================================== David C. Troy [dave@toad.net] 410-384-2500 Sales ToadNet - Want to go fast? 410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net

...risk.toad.net/app_zapbarge.c-change-to-zapscan Comments welcome! Dave ===================================================================== David C. Troy [dave@toad.net] 410-384-2500 Sales ToadNet - Want to go fast? 410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net

I searched for "say number" in the * google archives and have not found reference to options for "say number". I would like to have * say digits instead of the hundreds and thousands. EG, "1234" would say one two three four.

Is there a way to clear the Queue stats? That is with out restarting *? I'd like to reset them daily and don't see a way to do that. Unless the only way is maybe a cron restart asterisk like every weekday @ 04:00?

Hi everyone! I was wondering... Does the musiconhold quality improve if the mpg123 processes run with a negative priority? If so, is there a way to make them start like that, so I don't have to renice them? Regards, Evert

Hello, I've search through all of the lists and cannot find any descriptions of live monitoring (monitoring a phone call going on between an extension and a zaptel channel live from another extension while the monitoring phone is muted). I am aware of the monitor function which is actually a call recorder, but I'm looking for live monitoring from a muted extension. is this easily

...spoof the White House, btw, but it makes a fun example. Thanks, Dave ===================================================================== David C. Troy [dave@toad.net] 410-384-2500 Sales ToadNet - Want to go fast? 410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net

...this contradict any sort of master plan on the subject? Regards, Dave ===================================================================== David C. Troy [dave@toad.net] 410-384-2500 Sales ToadNet - Want to go fast? 410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

I have a PRI link up and running between Asterisk and a Nortel Norstar MICS v4.1 . I'm having a problem getting the textual Caller Name across the link from Ast to Ns, however numeric Caller ID arrives and displays fine. From Ns to Ast both elements come through fine. I'm forcing dummy values for testing using: exten => s,1,SetCIDName(Test) exten => s,2,SetCallerID(1234561234)

...Thanks, > > Dave > > > > ===================================================================== > > David C. Troy [dave@toad.net] 410-384-2500 Sales > > ToadNet - Want to go fast? 410-544-1329 FAX > > 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________...

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client