[asterisk-users] Call Presence for Offhook/Onhook Only
We have a customer with a dozen phones and they want nearly all of them to ring.Unfortunately this causes a firestorm of call presence notifications that overwhelm something on their network. Any existing calls get gappy audio for a few milliseconds when a new call comes in and when someone picks it up due to all the state changes between ringing and not ringing. They have a T-1 dedicated to voice so it isn't a bandwidth issue per se. We've been through a handful of routers and QOS settings but nothing has worked. Turning off the busy lamps fixes the problem but of course that isn't really a long term solution. Really I don't think anyone cares about the busy lamps for ringing. They just want to know when someone is on the phone. Is there any way short of hacking code that we can make notifications ignore changes involving ringing and just report inuse/notinuse? We are using 1.8.x if that matters. Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:25 PM, Danny Nicholas da...@debsinc.com wrote: You could do a simple PHP/Perl script to query hints and ring only the not-in-use phones. Or more simply that that do a ChanIsAvail() against the list and ring the returned array. If I do ChanIsAvail(line1/line2/line3/line4/line5) and 1 and 3 are in use, it returns an array with 2/4/5 and I can Dial the array. I don't think the problem isn't the phones that are in using getting the notifications so much as just the shear number of phones getting them. If we only ring a few phones the problem goes away even if the phone you are on is one of the ones getting the notifications. Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote: Check the notifyringing option in sip.conf Interesting. Looks like exactly what I want other than it looks like it is a global only setting? I'll play with it tonight but any idea if this is still global only? Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Presence for Offhook/Onhook Only
On Oct 22, 2012, at 4:11 PM, isr...@gmail.com wrote: Check the notifyringing option in sip.conf Looks like this really doesn't do what I had hoped: ;notifyringing = no ; Control whether subscriptions already INUSE get sent ; RINGING when another call is sent (default: yes) Chris -- - Chris Owen- Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - smime.p7s Description: S/MIME cryptographic signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] benefits of asterisk 1.8
On Jun 2, 2011, at 11:24 PM, Satish Barot wrote: With due respect to Digium work, are there no issues with Asterisk 1.8? https://issues.asterisk.org/view_all_bug_page.php And the first of those is a real show stopper at least for us. We've got to have multiple parking lots and that has been broken since the end of last year at least. We opened that ticket on 12/29/10. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
On Apr 28, 2011, at 9:53 AM, Russell Bryant wrote: I don't think it's a separate issue at all. I would like to see discussion of exactly which issues are preventing users from using Asterisk 1.8. We're trying to shift focus to those issues and get them resolved as quickly and as efficiently as we can so that we can all move forward. For us the biggest issue is multi-tenant parking not working. We've really given up testing anything beyond that point because without that feature there really isn't any way we could use it. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
On Apr 28, 2011, at 10:21 AM, Russell Bryant wrote: For us the biggest issue is multi-tenant parking not working. We've really given up testing anything beyond that point because without that feature there really isn't any way we could use it. Broken as compared to 1.6.2? I ask since that feature wasn't in 1.4. As compared to 1.6.1.x. We were using it precisely because we had to have multi-tenant parking. Can you point to a bug report? I'd like to understand better what's not working. https://issues.asterisk.org/view.php?id=18553 Basically for several versions of 1.6.2.x and all 1.8.x that we've tested, when you park a call it gets parked in the first parking lot regardless of what context the call is in when it is parked. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3
Best I can tell, multi-tenant parking also hasn't worked in any of the 1.8.x releases. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Jan 18, 2011, at 6:42 PM, Chad Wallace wrote: We need to ban all versions of outlook until microsoft decides to fix it. Amen. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Purpose of qualify=yes
On Sep 16, 2010, at 10:44 AM, Steve Totaro wrote: The other purpose is for DCHP and the IP address of a particular phone may change. If you hard code the phone and the corresponding entry in sip.conf, you don't need to register or use qualify. If the phone is reachable then it will reply and the call will go normally. If it doesn't reply, then on with the dialplan. Now I'm not sure that makes sense to me. If the IP address of the phone changes and the phone doesn't reregister then yes calls can't get to it but neither can the qualify packets. I'm not sure how sending a qualify helps here. Chris -- - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Purpose of qualify=yes
On Sep 16, 2010, at 10:45 AM, Zeeshan Zakaria wrote: I prefer to keep qualify=on for all the extensions, as it gives you an idea which extensions are going to give you trouble. For extensions with qualify value greater than 300 ms you should definitely worry. For extensions at 2000ms delay or more, turning qualify off simply means to ignore the obvious problem. Such extensions have communication or network issues which require serious attention. You can set this parameter to, e.g. 3000 ms or more if dealing with 2000 ms delay is unavoidable, but don't turn it off. Afterall even at 2000 ms conversation is not truly real time and not easy. In our case the problem isn't that the phones are experiencing high latency per se but rather than a full pipe plus all these SIP messages is playing hell with the QOS stuff. 20 phones in one location times say 4 SIP packets every 2 seconds equals 40 SIP packets a second. That normally isn't a problem but when the pipe gets congested then we start seeing issues when a call comes in and 400 BLF notices go out etc. Obviously we can increase the amount of bandwidth reserved for SIP traffic but I'm just not sure why we're sending all those packets in the first place. In other words, the qualify traffic is actually causing the problem, not revealing it. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Purpose of qualify=yes
We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what I'm unclear on is exactly what the purpose of having asterisk qualify the phones is. I know that in a NAT situation, qualifications can help keep UDP sessions open in the firewall but in our case most phones are not behind NAT. I realize qualifying phones is also how asterisk keeps track of who is available for things like BLF but surely it doesn't need to do that every 2 seconds to keep the BLFs reasonably current. So I guess my question is what is the real purpose of the qualify setting in a non-NAT situation and can one safely set the qualification as something higher. I'd think something like 15 seconds would be more than enough for BLFs and the like. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Apr 20, 2010, at 5:18 PM, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. This is an incredibly lame post on their part. They go out of their way to point out there was nothing unique about this attack that made it require that it come from EC2. However, that isn't true. Had this attack come from anywhere else it would have been shut down _days_ before it was on EC2. Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On Apr 12, 2010, at 8:17 AM, Fred Posner wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. I would contribute to this as well. Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phones won't stop ringing
We're having an issue that isn't easily googleable so I thought I might might try here. We have several customers who want all their extensions to ring on incoming calls. Frankly I think it is craziness to ring 11 extensions all at once but that is how they want it. We're doing this by creating an incoming route that goes to a hunt list containing all the extensions. This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. I'm pretty sure the underlying cause of this problem is funkiness in their network but it just seems to happen too easily and then once it stops it won't stop.Even if this is caused by network issues is there anything I can do to mitigate the problem. Just seems wrong that the phones would continue to ring forever. Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones won't stop ringing
On Mar 10, 2010, at 9:05 PM, Warren Selby wrote: On Wed, Mar 10, 2010 at 8:27 PM, Chris Owen ow...@hubris.net wrote: This normally works fine but occasionally when someone picks up the call other phones don't seem to realize the call has been answered and will continue to ring. On at least once occasion I saw a call that went to voicemail and all the phones continued to ring. When this happens the phones will continue to ring forever. The only way to stop them from ringing is to pickup the handset at which time they realize there is no call and reset. What kind of phones? All Aastra 6755i Chris - Chris Owen - Garden City (620) 275-1900 - Lottery (noun): President - Wichita (316) 858-3000 -A stupidity tax Hubris Communications Inc www.hubris.net - -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
