search for: narkov

Displaying 6 results from an estimated 6 matches for "narkov".

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2006 Oct 26
2
"Cheapest" way to determine channels in a group from outside asterisk?
I need to determine the number of active calls in a group from outside of Asterisk. Currently I poll the manager API and parse the channel status list but this is becoming too expensive on CPU. What are my options? What is considered "standard practice" ? Update a DB field? Poll the manager api? Use an asterisk -rv 'some command' call?
2007 May 12
2
zonedata.c
Hi, Could anyone tell me how to read the values in the "zonedata.c" file? I am looking at the "zt_tone_ringtone" field mainly. Thank you. Jad Wauthier -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070512/4c0387be/attachment.htm
2006 Dec 02
0
Answering Machine detection in Australia
Hello, Can anyone comment on the success of AMD/NVMachineDetect in an Australian setting? What kind of hit/miss ratio can we expect on a good quality g711 IAX tunk? Does the region even matter? I'm really not sure if these applications are tailored to a US/UK machines and VM services. Regards, Nick.
2007 Jan 06
0
Hint and call-limit issue
Hello, I have a Sipura SPA-3000 connected to my PSTN line and forwarding calls to my Asterisk box. It is a SIP peer "pstn-spa3k". I have setup "call-limit=1" in the peer config. When a call comes into Asterisk I get the correct "inuse" values but the hint isn't updated: sprite*CLI> sip show inuse * User name In use Limit * Peer name
2007 Apr 12
1
hanguponpolarityswitch - where did it go??
There are a few mentions in the wiki [1] about a zapata.conf flag "hanguponpolarityswitch". It is meant to cause Asterisk to detect a hangup when the line polarity switches at the end of the call. The wiki mentions using the flag in zapata.conf but when I do Asterisk "ignores" it: Apr 12 17:59:38 WARNING[12804]: chan_zap.c:10875 setup_zap: Ignoring hanguponpolarityswitch
2007 Nov 14
1
Asterisk ignoring manager events when busy
Hello, I currently have a pretty standard 1.2.21 Asterisk system running purely SIP termination (no zap/IAX/H323..etc). We have an auto-dialing system that generates calls via the manager API. The system runs beautifully until it gets to about 200 calls. I can generate these calls in quite literally seconds if desired (or minutes). The kicker: I can't seem to get past this 200 call point