[asterisk-users] 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 even though the system is/seems very idle. Low CPU utilisation (~40%), plenty of RAM, close to zero disk usage (I/O Wait ~2%), bandwidth is plentiful (1Gbit) and the current calls are crystal clear. I get the feeling Asterisk is ignoring or dumping manager events. We watch for the Success from the originate command and it comes if there is less than roughly 200 calls but is unreliable after that. I could understand if this was happening at 100% cpu but not at the under-utilised state the box is at. Any hints? Thanks, Nick. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ignoring manager events when busy
Doug Lytle wrote: Nick Adams wrote: The kicker: I can't seem to get past this 200 call point even though the What does your console show at this time? When testing, I've noted the 200 call limit was because I had too many open files. I had to increase this by typing ulimit -n 4096 before starting Asterisk. The default is 1024. Doug Thanks for the reply Doug. The console doesn't reveal anything too suspicious. I do get some RTP Read too short but I don't think that has anything to do with the problem. I'll adjust the open files limit and see how it goes. Thanks for your advice. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: DELL Platforms
Lacy Moore - Aspendora wrote: On 9/1/07, *Dovid B* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Why work with two separate devices when you can have one ? And yes the DC is staffed 24/7 but do you want to call them every time you need a new CD/DVD inserted in to the box when you are working on it ? IMHO A rac card + a better server is worth it than going with the SC series. The RAC card will swap CD/DVDs for you? Wow! That's pretty cool! No but you can remotely mount an ISO image and boot it. Super handy feature! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Difference between making a call and Originate
Christopher Robinson wrote: When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=SIP/[EMAIL PROTECTED], 'Context'='mycontext', 'Exten'='899', 'Priority'=1, 'Callerid'='whatever')); It creates a screech sound when the first audio file is played. Doesn't seem to happen with another VSP I tried, but still, why would a regular outbound call work just fine and Originate create this strange sound. I know for sure that it isn't the audio file that I'm playing by the way. I too have noticed this. I'm taking a stab in the dark here but is it possibly voice packets of a different codec being decoded as garbage/static? Our issue mysteriously went away. Don't know exactly what caused it though unfortunately. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Snom 320 voicemail key MWI
Stephen Bosch wrote: Ariel Monaco wrote: Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI notifications. But the light is still blinking and there are no voicemail messages, any ideas about how to address this issue will be welcome. You've mentioned how your Asterisk server is configured, but how is the *phone* configured? If the MWI light on the phone is set to use the wrong mailbox, you would see a blinking light, even if you've erased all the messages in the mailbox that is accessed from the voicemail button. Two things are happening here: 1. You've got a button that you configure for retrieving messages 2. You've got a Message Waiting Indicator light that blinks when there are messages in the specified mailbox. Those are separate things -- you can have a button that retrieves from one box and a light that indicates messages in another box. Check your phone configuration again. By default the Snom phones also use that light for missed calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: hanguponpolarityswitch - where did it go??
yusuf wrote: Hi, as far as I know, it only says ignoring when you do a reload, as Asterisk is telling you its not reconfiguring this variable, to change it you might need a restart. So hanguponpolarityswitch only gets looked at on startup, not reloads. Thanks for that. I restarted and I don't get any kind of output on either the CLI or in dmesg. Am I meant to see something? I'm not sure if it's really helping. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 Does anyone have any ideas how to enable or use this feature? Many thanks, Nick. [1] http://www.voip-info.org/wiki/view/Australia+Asterisk+Details ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Grandstream GXP2000 and Interception of call ?
Noc Phibee wrote: Hi i use a lot of Grandstream GXP2000 with BLF How to set up on the same key BLF blinking call interception? So that someone is able to take a call that is destinated to another user phone It's called Call Pickup. http://www.voip-info.org/wiki-PBX+Call+Pickup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: command like break ore exit in the dialpan
exten = 99,4,Hangup ? nik600 wrote: Hi i have a similar dialplan: exten = 99,1,Gotoif(?2:3) exten = 99,2,Meetme(100) exten = 99,3,Meetme(100|options) i'd like to do something like: exten = 99,1,Gotoif(?2:4) exten = 99,2,Meetme(100) exten = 99,4, ... exit ... exten = 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 In use Limit pstn-spa3k1 1 sprite*CLI show hints -= Registered Asterisk Dial Plan Hints =- 205 : SIP/pstn-spa3kState:Unavailable Watchers 1 When I call OUT (Asterisk - pstn-spa3k) the hint works as expected. It's just that incoming calls from the spa to Asterisk don't update the hint. Any ideas? Cheers, Nick. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AMI - Originate Action and Busy, NoAnswer calls - CDR
Michael Collins wrote:
Has anyone found a workaround or a best practice that allows CDR records
to contain the dialed phone number for every Dial() or Originate that
Asterisk processes?
I got around this by generating a call to a Local channel which is
always (well...nearly always) successful. The Local channel then issues
the Dial command and the dialplan captures the ${DIALSTATUS} via AGI.
Messy but works.
___
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[asterisk-users] 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. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: 3xx redirect from asterisk?
Benjamin Jacob wrote: Hello ppl, Is it possible to send a REDIRECT from an Asterisk box, to an incoming call?? e.g. A calling B, via Asterisk, Asterisk sends redirect to A to contact C. REINVITE? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: DTMF Tones
Jason Walker wrote: I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks What phones and codec are you using? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Cheapest way to determine channels in a group from outside asterisk?
They aren't zap interfaces unfortunately. They are SIP/IAX channels started from originate and the manager API. Lenz wrote: why not using a zap show command and parse the results externally? l. On Thu, 26 Oct 2006 13:12:46 +0200, Nick Adams [EMAIL PROTECTED] wrote: 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? --Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
