[asterisk-users] Asterisk 13 very high trasnlation time between codecs
Hi all. I've installaed a PIAF with Asterisk 13.0 on a ESX virtual machine on a DELL node. As usual I loaded g729 codec in modules folder but I notice very high translation time. Over several PIAF with Asterisk 1.8.X (always in a virtual environment) I always had for example 601 between G729 and ALAW or ULAW while in this case I have 15000. Even recalculating with different values (eg: core show translation recalc 100) I always get same result. I downloaded codec, as usual, from http://asterisk.hosting.lv/ and chosen appropriate version for my distro / CPU. Did someone face the same problem? Is this something related to VM? Or Version? Thanks in advance for any advice. Davide Anzaldi -- _ -- 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] OpenVZ with asterisk 13
Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta, Indonesia -- _ -- 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] Fwd: OpenVZ with asterisk 13
Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta, Indonesia -- _ -- 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] OpenVZ with asterisk 13
Why not use just one single box and create 300 sip clients having 150 odd con calls. OpenVZ might not be a good idea for this sort of volume. Mitul On 07-Apr-2015 7:12 PM, "Ikka Tirtawidjaja" wrote: > Dear all, > > Is anyone has experience making Asterisk server with virtual server > OPEN-VZ (in proxmox 3.4 box) ? > > My boss want to build a production server with it, and it will have +/- > 300 sip user (concurrent call maybe < 150 call) > > Is it good to go, or not ? > > I really hope someone who have experience with it willing to share with > me... > > Thanks in advance... > > > Best Regards, > > > Ikka - Jakarta, Indonesia > > > > -- > _ > -- 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] OpenVZ with asterisk 13
Dear Mitul, I already told my boss about it, I really want a single box, no virtual, but my boss insist. He said that openvz use less resource then KVM (or other virtual for cloud). I really need a solid analysis to argue with him. On the other hand, dahdi cannot be installed in openvz virtual server. I dont have any experience with openvz at all. Thx, On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja wrote: > Dear all, > > Is anyone has experience making Asterisk server with virtual server > OPEN-VZ (in proxmox 3.4 box) ? > > My boss want to build a production server with it, and it will have +/- > 300 sip user (concurrent call maybe < 150 call) > > Is it good to go, or not ? > > I really hope someone who have experience with it willing to share with > me... > > Thanks in advance... > > > Best Regards, > > > Ikka - Jakarta, Indonesia > > > > -- _ -- 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] OpenVZ with asterisk 13
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/07/2015 09:41 PM, Ikka Tirtawidjaja wrote: > Dear all, > > Is anyone has experience making Asterisk server with virtual > server OPEN-VZ (in proxmox 3.4 box) ? > > My boss want to build a production server with it, and it will have > +/- 300 sip user (concurrent call maybe < 150 call) > > Is it good to go, or not ? > > I really hope someone who have experience with it willing to share > with me... > > Thanks in advance... > > > Best Regards, > > > Ikka - Jakarta, Indonesia > > > > Guess we are talking about a callcenter, right? If, then use a single machine for it, not a VM. And if you can't do it, use docker instead OpenVZ for a much better performance. Regards Guenther Davao City, Philippines, Planet Earth - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVI+SAAAoJENexF5oIz3BC+I0QAPWnfgdFx0iJAl0QsUXYsgww i8KlaKNnAwgQLGjTi8y9Ijrm4eSgIv3zLxCLNTZOq/EG6S2MWb1vQ0KPTGKeydGP WqHW/FGIArb9QJ67752ANXGJWGX7SKX2zOGQGMi3nbkfOZZV7sL4+fAS7Grrj/xX eFvaYWcCcrngL7k614NAmEC80Zhj0uRdxscpSAmV0oR0YDRpFsAoSUQRT5/JroqJ 7eper9rjR7qPA3iLrP5QQIyGzLzem/c3hWRSnF++dRLV91UbfqchS5vqA191/t4D Rfb9b0OgJx5PrVxCBbj1vETpjcUfHCgBZQPi+03QHJ9lFtbeCU2XD+skemDUAin2 9l8GrDOyK2m+y6ybUBO/fY3pFz58Lb1C4zBVVjL1Qf+tzFj+F0SCyC94g7j674tf zW/6xkYzltO5I+uors6A8+jEr1T+Mom6uFJ9NVkL6jDBd/YmSEvgIrJYcJXbUML9 wr9OC+cLCwPGSvUhSm0T+jCo9p01vC+dZ3xNmzYrjoXn6uohuJe+L3LbjchZ3d0V Lwks3F7gObKgserwp23tyc/sy8Kcmrn2l6ZYibFqVbeIruPUBp9m/fmlDIX2gziA 0uuNFVUQFz7UnbXwHUzNK3B05zFPMaoHEMw1JygJyHAetl9Vz3ZYll8PPAtbxb27 DqIZs2yydHJmzJ3td0xM =FKrd -END PGP 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] OpenVZ with asterisk 13
Show him this freaking thread, or else ask him to prove it otherwise. We all here have decades of exp dealing with asterisk. Mitul On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" wrote: > Dear Mitul, > > I already told my boss about it, I really want a single box, no virtual, > but my boss insist. > He said that openvz use less resource then KVM (or other virtual for > cloud). > I really need a solid analysis to argue with him. > > On the other hand, dahdi cannot be installed in openvz virtual server. > > I dont have any experience with openvz at all. > > Thx, > > On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja > wrote: > >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >> >> Is it good to go, or not ? >> >> I really hope someone who have experience with it willing to share with >> me... >> >> Thanks in advance... >> >> >> Best Regards, >> >> >> Ikka - Jakarta, Indonesia >> >> >> >> > > -- > _ > -- 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] OpenVZ with asterisk 13
I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani : > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" wrote: > >> Dear Mitul, >> >> I already told my boss about it, I really want a single box, no virtual, >> but my boss insist. >> He said that openvz use less resource then KVM (or other virtual for >> cloud). >> I really need a solid analysis to argue with him. >> >> On the other hand, dahdi cannot be installed in openvz virtual server. >> >> I dont have any experience with openvz at all. >> >> Thx, >> >> On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja >> wrote: >> >>> Dear all, >>> >>> Is anyone has experience making Asterisk server with virtual server >>> OPEN-VZ (in proxmox 3.4 box) ? >>> >>> My boss want to build a production server with it, and it will have +/- >>> 300 sip user (concurrent call maybe < 150 call) >>> >>> Is it good to go, or not ? >>> >>> I really hope someone who have experience with it willing to share with >>> me... >>> >>> Thanks in advance... >>> >>> >>> Best Regards, >>> >>> >>> Ikka - Jakarta, Indonesia >>> >>> >>> >>> >> >> -- >> _ >> -- 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 > -- _ -- 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] OpenVZ with asterisk 13
PBX =! CC my friend. 150 Conc Calls for CC agent is going to be far more expensive then running 200 extn PBX doing hardly 20 Conc Calls. Load is way too diff. On 07-Apr-2015 8:18 PM, "Vinicius Fontes" wrote: > I have several large customers (200+ extensions) running on vSphere > without issue. Not sure about OpenVZ, thought. > > 2015-04-07 11:36 GMT-03:00 Mitul Limbani : > >> Show him this freaking thread, or else ask him to prove it otherwise. >> >> We all here have decades of exp dealing with asterisk. >> >> Mitul >> On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" wrote: >> >>> Dear Mitul, >>> >>> I already told my boss about it, I really want a single box, no virtual, >>> but my boss insist. >>> He said that openvz use less resource then KVM (or other virtual for >>> cloud). >>> I really need a solid analysis to argue with him. >>> >>> On the other hand, dahdi cannot be installed in openvz virtual server. >>> >>> I dont have any experience with openvz at all. >>> >>> Thx, >>> >>> On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja >>> wrote: >>> Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) Is it good to go, or not ? I really hope someone who have experience with it willing to share with me... Thanks in advance... Best Regards, Ikka - Jakarta, Indonesia >>> >>> -- >>> _ >>> -- 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 >> > > > -- > _ > -- 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] OpenVZ with asterisk 13
I guess best way for your boss to learn is to deploy a box once and get bombed and then follow what ppl said here. Both modes u should be the happy guy u see, u will get paid twice for same work !!! Mitul On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" wrote: > Dear Mitul, > > I already told my boss about it, I really want a single box, no virtual, > but my boss insist. > He said that openvz use less resource then KVM (or other virtual for > cloud). > I really need a solid analysis to argue with him. > > On the other hand, dahdi cannot be installed in openvz virtual server. > > I dont have any experience with openvz at all. > > Thx, > > On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja > wrote: > >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >> >> Is it good to go, or not ? >> >> I really hope someone who have experience with it willing to share with >> me... >> >> Thanks in advance... >> >> >> Best Regards, >> >> >> Ikka - Jakarta, Indonesia >> >> >> >> > > -- > _ > -- 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] OpenVZ with asterisk 13
Dear Guenther B. This server is not for call center. Its for office and appartment with +/- 900 sip users. The asterisk server will be split to 3 OpenVZ Virtual server in 1 proxmox server (but they will have clustering server in another proxmox server) and 1 database server (mysql), and it also use OpenVZ - Proxmox server. If its for call center with 300 concurrent call, I already have my experience with that situation, that I use 3 server, 1 for asterisk, 1 for database, and 1 for web application. Dear Mitul L. Yes, I will do that. I dont know why my boss love openvz so much. But he is a stubborn man... Thx & best regards... On Tue, Apr 7, 2015 at 9:36 PM, Mitul Limbani wrote: > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" wrote: > >> Dear Mitul, >> >> I already told my boss about it, I really want a single box, no virtual, >> but my boss insist. >> He said that openvz use less resource then KVM (or other virtual for >> cloud). >> I really need a solid analysis to argue with him. >> >> On the other hand, dahdi cannot be installed in openvz virtual server. >> >> I dont have any experience with openvz at all. >> >> Thx, >> >> On Tue, Apr 7, 2015 at 8:47 PM, Ikka Tirtawidjaja >> wrote: >> >>> Dear all, >>> >>> Is anyone has experience making Asterisk server with virtual server >>> OPEN-VZ (in proxmox 3.4 box) ? >>> >>> My boss want to build a production server with it, and it will have +/- >>> 300 sip user (concurrent call maybe < 150 call) >>> >>> Is it good to go, or not ? >>> >>> I really hope someone who have experience with it willing to share with >>> me... >>> >>> Thanks in advance... >>> >>> >>> Best Regards, >>> >>> >>> Ikka - Jakarta, Indonesia >>> >>> >>> >>> >> >> -- >> _ >> -- 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 > -- _ -- 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] OpenVZ with asterisk 13
Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) As long as you don't overload the server it works great. I've used OpenVZ to separate Asterisk instances from each other. For my application (mostly conferencing) I can put ~ 350 concurrent calls on a single HP Xeon server. OpenVZ is not really like KVM but more like Solaris containers or BSD jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but OpenVZ also has some cusom stuff. If you need Dahdi you will need to give the VE's access to these devices, there are articles out there that explain how this is done. Good luck! -- Johan Wilfer -- _ -- 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] OpenVZ with asterisk 13
On 04/07/2015 10:48 AM, Johan Wilfer wrote: Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe < 150 call) As long as you don't overload the server it works great. I've used OpenVZ to separate Asterisk instances from each other. For my application (mostly conferencing) I can put ~ 350 concurrent calls on a single HP Xeon server. OpenVZ is not really like KVM but more like Solaris containers or BSD jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but OpenVZ also has some cusom stuff. If you need Dahdi you will need to give the VE's access to these devices, there are articles out there that explain how this is done. Good luck! We use LXC (what is under Docker) instead of OpenVZ to separate asterisk instances, and when Dahdi is needed I typically run an asterisk instance "on the host" and have SIP trunks between the container and the host instances. Cheers, j -- _ -- 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] OpenVZ with asterisk 13
With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mi...@enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Tue, Apr 7, 2015 at 9:57 PM, Jeff LaCoursiere wrote: > On 04/07/2015 10:48 AM, Johan Wilfer wrote: > >> Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> >>> Dear all, >>> >>> Is anyone has experience making Asterisk server with virtual server >>> OPEN-VZ (in proxmox 3.4 box) ? >>> >>> My boss want to build a production server with it, and it will have +/- >>> 300 sip user (concurrent call maybe < 150 call) >>> >>> >> As long as you don't overload the server it works great. I've used OpenVZ >> to separate Asterisk instances from each other. For my application (mostly >> conferencing) I can put ~ 350 concurrent calls on a single HP Xeon server. >> >> OpenVZ is not really like KVM but more like Solaris containers or BSD >> jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, but >> OpenVZ also has some cusom stuff. >> >> If you need Dahdi you will need to give the VE's access to these devices, >> there are articles out there that explain how this is done. >> >> Good luck! >> >> > We use LXC (what is under Docker) instead of OpenVZ to separate asterisk > instances, and when Dahdi is needed I typically run an asterisk instance > "on the host" and have SIP trunks between the container and the host > instances. > > Cheers, > > > j > > -- > _ > -- 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] OpenVZ with asterisk 13
Den 2015-04-07 20:47, Mitul Limbani skrev: With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. That's another issue thought and is not different on a dedicated server. With proper monitoring of the resources, you can troubleshoot. I've found voipmonitor.org to be invaluable in that regard. -- Johan Wilfer -- _ -- 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] res_fax.c: allowed rates for V27 modems
Hi all, We are running a fax2email service based on asterisk 1.8.18.0, and we are currently trying out asterisk 1.8.32.2 in our labs. We get the following error when sending faxes out: [Apr 7 14:34:20] ERROR[16653]: res_fax.c:2121 sendfax_exec: 'modems' setting 'V17,V27,V29' is incompatible with 'minrate' setting 2400 It looks like function check_modem_rate in res_fax.c has been updated and rate 2400 is not allowed for V27 any more. Found the following issue explaining the change: https://issues.asterisk.org/jira/browse/ASTERISK-23231. However ITU-T specifications for V27ter (which should supersede the one for V27) specify both 2400 and 4800. We are currently receiving faxes at rate 2400 on our production servers so we can't upgrade asterisk as is. Does anybody have some insights on this? 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] Fidelio protocol and Mitel protocol
Does anyone know anything about the Fidelio and Mitel protocol for hotel / motel? Are these industry standards or proprietary formats? Are there open standards for communication with Hotel management software's that could be used in conjunction with a custom asterisk deployment? Thanks Bryant -- _ -- 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] Help debugging a possible SIP channel leak in 11.17.0, possible race condition
I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below: 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s talktime), W:0, C:581, A:260, SL:82.6% within 60s Members: SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last was 800 secs ago) SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last was 708 secs ago) SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last was 656 secs ago) SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last was 789 secs ago) No Callers The program runs "queue show" through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are "Not In Use". If at least one is found, it reads that many phone numbers from the database and uses the AMI Originate command on each one, as follows: Action: Originate Channel: Local/NN@from-internal Exten: Context: from-internal Priority: 1 Async: true ActionID: xxx Here, NN is the number read from the database and is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX. The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the "queue show" command shows them to be "In Use" on all queues. Furthermore, in normal operation, the "core show channels" command shows at most one channel for each configured SIP client in the "Up" state, but when calls stop being placed, the same command reports multiple channels in the "Up" state, as follows (after sorting): Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192 Local/9759315789@from-internal-a456;1!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing Line)!9759315789!!!3!500!(None)!1428426084.169326 Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!110!!!3!590!(None)!1428425994.169124 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493 SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,"",trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510 SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!121!!!3!596!(None)!1428425988.169111 SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing Line)!122!!!3!677!(None)!1428425906.168693 SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499 SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735 SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844 SIP/125-00014bc7!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740435/0445549261961,30
Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition
El 07/04/15 a las 17:38, Alex Villacís Lasso escribió: I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below: 9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s talktime), W:0, C:581, A:260, SL:82.6% within 60s Members: SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last was 800 secs ago) SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last was 708 secs ago) SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last was 656 secs ago) SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last was 789 secs ago) No Callers The program runs "queue show" through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are "Not In Use". If at least one is found, it reads that many phone numbers from the database and uses the AMI Originate command on each one, as follows: Action: Originate Channel: Local/NN@from-internal Exten: Context: from-internal Priority: 1 Async: true ActionID: xxx Here, NN is the number read from the database and is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NN through one of a series of SIP trunks using the outbound routes as configured by FreePBX. The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the "queue show" command shows them to be "In Use" on all queues. Furthermore, in normal operation, the "core show channels" command shows at most one channel for each configured SIP client in the "Up" state, but when calls stop being placed, the same command reports multiple channels in the "Up" state, as follows (after sorting): Local/9757007441@from-internal-a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192 Local/9759315789@from-internal-a456;1!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing Line)!9759315789!!!3!500!(None)!1428426084.169326 Local/9759315789@from-internal-a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323 SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224 SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!110!!!3!590!(None)!1428425994.169124 SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760 SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273 SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384 SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850 SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113 SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436 SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493 SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,"",trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510 SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!121!!!3!596!(None)!1428425988.169111 SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing Line)!122!!!3!677!(None)!1428425906.168693 SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499 SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735 SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844 SIP/125-00014bc7!macro-d
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on. For example, the receptionist answers calls
for 8 different companies and would like the phone to display the company
name that she should announce to the caller.
Here is a more complete output of an incoming call. I've changed the SIP
numbers to "Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
> Verbosity is at least 12
> asterisk*CLI>
> asterisk*CLI>
> asterisk*CLI>
> == Using SIP RTP CoS mark 5
> -- Executing [s@incoming:1] *Set*("*SIP/Company1-0797*",
> "*thedid=""NodePhone" >"*") in new stack
> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid="NodePhone" >*") in new stack
> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid="NodePhone" -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-0797*", "
> *pseudodid=** sip:Company2*") in new stack
> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,33,1:6*") in new stack
> -- Goto (incoming,s,6)
> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,88,1:7*") in new stack
> -- Goto (incoming,s,7)
> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-0797*", "
> *0?internal,36,1:8*") in new stack
> -- Goto (incoming,s,8)
> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-0797*", "
> *1?internal,36,1:9*") in new stack
> -- Goto (internal,36,1)
> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-0797*", "
> *CALLERID(name)=SIP/**Company1**-0797*") in new stack
> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-0797*", "
> *SIP/36,20*") in new stack
> == Using SIP RTP CoS mark 5
> -- Called SIP/36
> -- SIP/36-0798 is ringing
> == Spawn extension (internal, 36, 2) exited non-zero on
> 'SIP/Company1-0797'
> asterisk*CLI> exit
And here is the "sip.conf":
[general]
> match_auth_username=yes
> register=081...:...@sip.internode.on.net/s
> register=082...:...@sip.internode.on.net/s
> register=083...:...@sip.internode.on.net:/s
> register=084...:...@sip.internode.on.net:/s
> register=085...:...@sip.internode.on.net/s
> register=086...:...@sip.internode.on.net/s
> register=087...:...@sip.internode.on.net/s
> register=088...:...@sip.internode.on.net/s
>
> [Company1]
> username=081...
> fromuser=081...
> secret=...
> canreinvite=no
> qualify=yes
> context=incoming
> type=friend
> insecure=invite,port
> fromdomain=sip.internode.on.net
> host=sip.internode.on.net
> dtmfmode=rfc2833
> disallow=all
> allow=alaw
> allow=ulaw
> allow=g729
> bindport=5060
> bindaddr=0.0.0.0
> nat=yes
> registertimeout=5
> allowoverlap=no
> srvlookup=no
> ubscribecontext=from-sip
> callcounter=yes
[Company2]
> ...
> [Company3]
> ...
> [Company4]
> ...
And here is some of the "extensions.conf" file:
[incoming]
> ; Get the DID number from the TO header.
> exten => s,1,Set(thedid="${SIP_HEADER(TO)}")
> exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
> exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
> exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
> ; Direct the DID accordingly.
> exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
> exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
> exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
> exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
> exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
> exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
> exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
> exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov wrote:
>
> This is one of the chronic problems. Try this option in sip.conf:
> match_auth_username=yes
>
> Carefully read the description, it is better to test in "after hours".
>
> 02.04.2015 2:50, Andrew Galdes пишет:
>
> Hello all,
>
> I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
> with the same service provides. We have 8 phone numbers in total.
>
> Incoming calls from the public are all correctly directed to appropriate
> office handsets. However, the display on the reception phone (the only one
> i care about) is always showing the same "SIP/Account1_0843214321" rather
> than the account representing the number dialed.
>
> For-instance, if Sam on her mobile calls "*08*", Asterisk will
> show a log entry like the following:
>
> -- Executing [s@incoming:1] Set("SIP/*Acc
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.
Here is my "extensions.conf" file:
exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
exten => s,5,Set(callersname=${IF($[ ${pseudodid} =
081...]?Company1:${callersname})})
exten => s,6,Set(callersname=${IF($[ ${pseudodid}
= 082...]?Company2:${callersname})})
exten => s,7,Set(callersname=${IF($[ ${pseudodid}
= 083...]?Company3:${callersname})})
exten => s,8,Set(callersname=${IF($[ ${pseudodid}
= 084...]?Company4:${callersname})})
exten => s,9,Set(callersname=${IF($[ ${pseudodid}
= 085...]?Company5:${callersname})})
exten => s,10,Set(callersname=${IF($[ ${pseudodid}
= 086...]?Company6:${callersname})})
exten => s,11,Set(callersname=${IF($[ ${pseudodid}
= 087...]?Company7:${callersname})})
exten => s,12,Set(callersname=${IF($[ ${pseudodid}
= 088...]?Company8:${callersname})})
exten => s,13,GotoIf($["${callersname}" = "Company1"]?internal,36,1:14); to
reception
exten => s,14,GotoIf($["${callersname}" = "Company2"]?internal,88,1:15); to
department1
exten => s,15,GotoIf($["${callersname}" = "Company3"]?internal,36,1:16); to
reception
exten => s,16,GotoIf($["${callersname}" = "Company4"]?internal,36,1:17); to
reception
exten => s,17,GotoIf($["${callersname}" = "Company5"]?internal,36,1:18); to
reception
exten => s,18,GotoIf($["${callersname}" = "Company6"]?internal,89,1:19); to
department2
exten => s,19,GotoIf($["${callersname}" = "Company7"]?internal,36,1:20); to
reception
exten => s,20,GotoIf($["${callersname}" = "Company8"]?internal,13,1:21); to
department3
And later in same file:
; Phone 36 reception
> *exten => 36,1,Set(CALLERID(name)=${callersname})*
> exten => 36,n,Dial(SIP/36,20)
> exten => 36,n,VoiceMail(36,u)
> exten => 36,n,Hangup
Ta,
-Andrew Galdes
Managing Director
RHCE, LPI, CCENT
AGIX Linux
Ph: 08 7324 4429
Mb: 0422 927 598
Find us: Website | LinkedIn
| Blog |
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*Platform Architects for High Demand Web Applications.*
On Wed, Apr 8, 2015 at 9:18 AM, Andrew Galdes
wrote:
> Hi Dmitriy and others and thanks for your help so far.
>
> The option "match_auth_username=yes" seems to have had no effect. From my
> reading, this option will try to match the username of the incoming SIP
> account to a section heading. If that is how it must work then i can see a
> big problem. I'm trying to present the receptionist with a nice display of
> which line the call came in on. For example, the receptionist answers calls
> for 8 different companies and would like the phone to display the company
> name that she should announce to the caller.
>
> Here is a more complete output of an incoming call. I've changed the SIP
> numbers to "Company1', etc, to hide the numbers.
>
> Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
>> Verbosity is at least 12
>> asterisk*CLI>
>> asterisk*CLI>
>> asterisk*CLI>
>> == Using SIP RTP CoS mark 5
>> -- Executing [s@incoming:1] *Set*("*SIP/Company1-0797*",
>> "*thedid=""NodePhone"> >"*") in new stack
>> -- Executing [s@incoming:2] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> >*") in new stack
>> -- Executing [s@incoming:3] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid="NodePhone"> -- Executing [s@incoming:4] *Set*("*SIP/**Company1**-0797*", "
>> *pseudodid=** sip:Company2*") in new stack
>> -- Executing [s@incoming:5] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,33,1:6*") in new stack
>> -- Goto (incoming,s,6)
>> -- Executing [s@incoming:6] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,88,1:7*") in new stack
>> -- Goto (incoming,s,7)
>> -- Executing [s@incoming:7] *GotoIf*("*SIP/**Company1**-0797*", "
>> *0?internal,36,1:8*") in new stack
>> -- Goto (incoming,s,8)
>> -- Executing [s@incoming:8] *GotoIf*("*SIP/**Company1**-0797*", "
>> *1?internal,36,1:9*") in new stack
>> -- Goto (internal,36,1)
>> -- Executing [36@internal:1] *Set*("*SIP/**Company1**-0797*", "
>> *CALLERID(name)=SIP/**Company1**-0797*") in new stack
>> -- Executing [36@internal:2] *Dial*("*SIP/**Company1**-0797*", "
>> *SIP/36,20*") in new stack
>> == Using SIP RTP CoS mark 5
>> -- Called SIP/36
>> -- SIP/36-0798 is ringing
>> == Spawn extension (internal, 36, 2) exited non-zero on
>> 'SIP/Company1-0797'
>> asterisk*CLI> exit
>
>
> And here is the "sip.conf":
>
> [general]
>> match_auth_username=yes
>> register=081...:...@sip.internode.on.net/s
>> register=082...:...@sip.internode.on.net/s
>> register=083...:...@sip.in
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
On 4/7/15 7:48 PM, Andrew Galdes wrote:
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From
my reading, this option will try to match the username of the incoming
SIP account to a section heading. If that is how it must work then i
can see a big problem. I'm trying to present the receptionist with a
nice display of which line the call came in on. For example, the
receptionist answers calls for 8 different companies and would like
the phone to display the company name that she should announce to the
caller.
Here is a more complete output of an incoming call. I've changed the
SIP numbers to "Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid =
32267)
Verbosity is at least 12
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
== Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*("*SIP/Company1-0797*",
"*thedid=""NodePhone"mailto:sip%3acompa...@sip.internode.on.net>>"*") in new stack
-- Executing [s@incoming:2]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone"http://sip.internode.on.net>>*") in new stack
-- Executing [s@incoming:3]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone" exit
And here is the "sip.conf":
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net
host=sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes
[Company2]
...
[Company3]
...
[Company4]
...
And here is some of the "extensions.conf" file:
[incoming]
; Get the DID number from the TO header.
exten => s,1,Set(thedid="${SIP_HEADER(TO)}")
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
Since your objective is to have the receptionist identify the company
she should be answering to then might I suggest a simple workaround to
your problem. Since right here you are already sending the call to the
expected internal context and extension, you could simply alter the
Caller Name and put in the Company Name so she could see it on the
screen. Something like:
[internal]
exten => 33,1,Set(CALLERID(name)=Company1:${CALLERID})
...
exten => 88,1,Set(CALLERID(name)=Company2:${CALLERID})
...
exten => 36,1,Set(CALLERID(name)=Company3:${CALLERID})
...
etc...
That will display the Company Name you want to see followed by the
caller ID #
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov > wrote:
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in "after hours".
02.04.2015 2:50, Andrew Galdes пишет:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the same service provides. We have 8 phone numbers
in total.
Incoming calls from the public are all correctly directed to
appropriate office handsets. However, the display on the
reception phone (the only one i care about) is always showing the
same "SIP/Account1_0843214321" rather than the account
representing the nu
Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete "register" lines.
b) add option "callbackextension=Company1" to Company1 friend section..
And in others with their names too.
or you can change "/s" to "/Company1" in register line.
2. beautiful display of this information
a) add option "setvar=fromCompany=Company1" to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})
Maybe this will help?
Dmitiy.
08.04.2015 2:48, Andrew Galdes пишет:
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From
my reading, this option will try to match the username of the incoming
SIP account to a section heading. If that is how it must work then i
can see a big problem. I'm trying to present the receptionist with a
nice display of which line the call came in on. For example, the
receptionist answers calls for 8 different companies and would like
the phone to display the company name that she should announce to the
caller.
Here is a more complete output of an incoming call. I've changed the
SIP numbers to "Company1', etc, to hide the numbers.
Connected to Asterisk 10.12.4 currently running on asterisk (pid =
32267)
Verbosity is at least 12
asterisk*CLI>
asterisk*CLI>
asterisk*CLI>
== Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*("*SIP/Company1-0797*",
"*thedid=""NodePhone"mailto:sip%3acompa...@sip.internode.on.net>>"*") in new stack
-- Executing [s@incoming:2]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone"http://sip.internode.on.net>>*") in new stack
-- Executing [s@incoming:3]
*Set*("*SIP/**Company1**-0797*",
"*pseudodid="NodePhone" exit
And here is the "sip.conf":
[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net
host=sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes
[Company2]
...
[Company3]
...
[Company4]
...
And here is some of the "extensions.conf" file:
[incoming]
; Get the DID number from the TO header.
exten => s,1,Set(thedid="${SIP_HEADER(TO)}")
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)})
; Direct the DID accordingly.
exten => s,5,GotoIf($["${pseudodid}" = "081"]?internal,33,1:6)
exten => s,6,GotoIf($["${pseudodid}" = "082"]?internal,88,1:7)
exten => s,7,GotoIf($["${pseudodid}" = "083"]?internal,36,1:8)
exten => s,8,GotoIf($["${pseudodid}" = "084"]?internal,36,1:9)
exten => s,9,GotoIf($["${pseudodid}" = "085"]?internal,36,1:10)
exten => s,10,GotoIf($["${pseudodid}" = "086"]?internal,89,1:11)
exten => s,11,GotoIf($["${pseudodid}" = "087"]?internal,36,1:12)
exten => s,12,GotoIf($["${pseudodid}" = "088"]?internal,13,1:13)
-Andrew Galdes
On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov > wrote:
This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes
Carefully read the description, it is better to test in "after hours".
02.04.2015 2:50, Andrew Galdes пишет:
Hello all,
I have an Asterisk server (Asterisk 10.12.4) with multiple sip
accounts with the same service provides. We have 8 phone numbers
in total.
Incoming calls from the public are all correctly directed to
appropriate office handsets. However, the display on the
reception phone (the only one i care about) is always showing the
same "SIP/Account1_0843214321" rath
