[asterisk-users] Problem with reINVITE on BYE
Hello all. I am currently using Asterisk 11.7.0 (also tried Asterisk 12, but same behavior) and is having an issue when it comes to reINVITE on BYEs. Apparently one of the SIP providers that I am using does not always process reINVITEs correctly, and would return a 500 Internal Server Error message on some (but not all) of these transactions. To get around this issue, I have been using directrtpsetup = yes in my sip.conf, and it worked quite well. However, even with this option set, Asterisk would reINVITE itself back into the audio path as soon as the caller hangs up. The behavior I am seeing is that if the SIP provider sends back a 500 Internal Server Error on the reINVITE, Asterisk will not hang up the call until the called party hangs up. The transaction goes something like this: 1. Caller calls a number using a target SIP server. 2. -- Early Media -- 3. Answer, -- Answering Machine -- 4. Caller hangs up 5. Asterisk sends reINVITE to target SIP server back to itself --- If SIP server returns 500 Internal Server Error, step 6 is never reached and the call stalls. 6. Asterisk sends a BYE to the target SIP server This is what happens when the reINVITE proceeds normally, but if 500 Internal Server Error is returned on step 5, then Asterisk will only acknowledge the 500 Internal Server Error and never send back a BYE. As a result, the other parties are getting minutes of empty voicemail (due to timeout) and I am getting charged for these minutes on my provider. With this in mind, is there something I can do so that a BYE is sent immediately to the SIP provider when the client initiates a hang up? I don't believe this can be done via some kind of setting, but maybe changing the source may help. I don't plan to have any dialplan rules execute after hangup so making this a global option would be okay in my case. Anyone has any pointers? Thanks! -- _ -- 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 11, CEL and ConfBridge recordings
Thank you very much Rusty. It really works. Even if ${MyCustomFileName} gets a different value when the second participant enters the conference, the filename remains the name defined when the first participant enters (because he started the conference). Another thing, if I need to know the time a conference ended should I use CEL or is there another better approach? Best. 2014-03-06 21:28 GMT-03:00 Rusty Newton rnew...@digium.com: On Wed, Mar 5, 2014 at 1:30 PM, Jairo jairomolin...@gmail.com wrote: Dear friends, Need to know filenames of conference recordings in Asterisk 11. Besides directory scanning the recordings could use CEL: Filter MySQL rows with eventtype equal CHAN_START and channame like ConfBridgeRecorder and then get the eventtime field and convert to timestamp to complete filename(s). Would you suggest any other approaches? You might set the record file path yourself through the CONFBRIDGE function, for example, in dialplan: ...stuff up here to build a unique file name into MyCustomFileName... exten = 1,n,Set(CONFBRIDGE(user,record_file)=${MyCustomFileName}.wav) Then of course you now know the file name so you could do whatever you wanted with it afterwards. https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CONFBRIDGE -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 -- _ -- 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] High Availability with Asterisk
On Thu, Mar 6, 2014 at 10:57 AM, Mitul Limbani mi...@enterux.in wrote: Hello, Using Single Server with multiple VMs essentially kills the purpose, coz it doesnt protect against physical hardware failures. To save costs, use low end box as failover, to keep u in business, till primary box goes live. Correct, in this case para-virt is not the way to go. You'll want to use a virtualization platform that does support multi-hardware with live migration support. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] High Availability with Asterisk
On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. If you want to further develop the high availability thought, then you could use CephFS which will give you self-healing, 100% available storage over multiple physical storage servers. There you could store your Asterisk config files, or your MySQL database used by all the Asterisk servers, for CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for Asterisk / MySQL. :) And, to scale and to make the Asterisk nodes redundant (redundancy is not really needed anymore, since Kamailio takes care of that, but basically then you get also VM/physical redundancy), you could look into OpenNebula which provides a nice auto-scaling feature already out of the box. If there's load on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs (probably on different physical servers, otherwise it doesn't make that much sense performance-wise) which will automagically receive requests/calls from Kamailio. If the load goes down, the VM can be automagically stopped again to free resources for other VMs/applications. OpenNebula is less popular than OpenStack, which seems to be the first choice for Cloud-stuff today, but what I liked about OpenNebula is that it provides the auto-scaling feature already in the customer-facing web-frontend out-of-the-box, unlike OpenStack. So you could offer your customers a self-managed, redundant Asterisk cloud or something like that. :) In theory, this combination should give you a 100% redundant, auto-healing, auto-scaling VoIP setup. :) +1 to this post. A lot of good information here. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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] High Availability with Asterisk
Good post. Actually this is the architecture we have. On Fri, Mar 7, 2014 at 11:31 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. If you want to further develop the high availability thought, then you could use CephFS which will give you self-healing, 100% available storage over multiple physical storage servers. There you could store your Asterisk config files, or your MySQL database used by all the Asterisk servers, for CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for Asterisk / MySQL. :) And, to scale and to make the Asterisk nodes redundant (redundancy is not really needed anymore, since Kamailio takes care of that, but basically then you get also VM/physical redundancy), you could look into OpenNebula which provides a nice auto-scaling feature already out of the box. If there's load on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs (probably on different physical servers, otherwise it doesn't make that much sense performance-wise) which will automagically receive requests/calls from Kamailio. If the load goes down, the VM can be automagically stopped again to free resources for other VMs/applications. OpenNebula is less popular than OpenStack, which seems to be the first choice for Cloud-stuff today, but what I liked about OpenNebula is that it provides the auto-scaling feature already in the customer-facing web-frontend out-of-the-box, unlike OpenStack. So you could offer your customers a self-managed, redundant Asterisk cloud or something like that. :) In theory, this combination should give you a 100% redundant, auto-healing, auto-scaling VoIP setup. :) +1 to this post. A lot of good information here. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- 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 -- _ -- 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] High Availability with Asterisk
On 2014-03-07 17:31, Paul Belanger wrote: On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote: Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. Sorry, for the stupid question, but what happens if Kamailio fails ? Thanks. regards Hans -- _ -- 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] High Availability with Asterisk
On 07/03/14 16:52, Johann Steinwendtner wrote: Sorry, for the stupid question, but what happens if Kamailio fails ? We have two copies on different servers which make use of keepalived to provide a virtual IP address between them. We also have them connected to two databases with active-active replication. -- _ -- 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 11, CEL and ConfBridge recordings
On Fri, Mar 7, 2014 at 7:21 AM, Jairo jairomolin...@gmail.com wrote: Thank you very much Rusty. It really works. Even if ${MyCustomFileName} gets a different value when the second participant enters the conference, the filename remains the name defined when the first participant enters (because he started the conference). Another thing, if I need to know the time a conference ended should I use CEL or is there another better approach? If you can get it from CEL, there is that, otherwise you can track when you receive the AMI event ConfbridgeEnd https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeEnd That is all I got from poking around the docs. :) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 11, CEL and ConfBridge recordings
Great :) Thank you very much. Best. 2014-03-07 18:07 GMT-03:00 Rusty Newton rnew...@digium.com: On Fri, Mar 7, 2014 at 7:21 AM, Jairo jairomolin...@gmail.com wrote: Thank you very much Rusty. It really works. Even if ${MyCustomFileName} gets a different value when the second participant enters the conference, the filename remains the name defined when the first participant enters (because he started the conference). Another thing, if I need to know the time a conference ended should I use CEL or is there another better approach? If you can get it from CEL, there is that, otherwise you can track when you receive the AMI event ConfbridgeEnd https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeEnd That is all I got from poking around the docs. :) -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- 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 -- _ -- 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