Re: [asterisk-users] Syncronizing files on different Asterisk servers
Thanks Alot @ Jeff LaCoursiere,@Arjan Kroon,@Robin,@Joseph @ Jeff LaCoursiere Well you already suggested that you would send all files to server A, so A is your server Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . Can you give me some clues or links @ Arjan Kroon As i dont have good idea about Mounting what about the script actually i need some thing that dont needs human hand after development. And if script can do this then it will be fine. @Robin Which Application do use for that ?? Please elaborate Hell, you could even abuse dropbox for this purpose. What does this means? @ Joseph No Joseph its not some thing voice mail its recording of suggestions etc Actually operators are located at different locations and if a user leave a suggestion at one operator then the file will be on that particular server. But if the user of another operator want to listen that file then this file must be present on that server also ..Thats why I am considering these options On Wed, Oct 21, 2009 at 10:08 AM, Joseph syscon...@gmail.com wrote: On 10/20/09 17:24, ABBAS SHAKEEL wrote: Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C users come to listen the recorded voices (I am using agi ). any user records his profile on server B , NOW a user who make a call to server C cannot listen to profiles recorded at server B. Because these profiles reside on Server B ... Similar in case of server C. By ...listen to profile... do you mean retrieve their voice-mail on a different server? -- Joseph ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] Syncronizing files on different Asterisk servers
With dropbox i mean a service (http://getdropbox.com). I've been thinking about using dropbox for stuff at my asterisk servers, but haven't done so yet. It was just an idea that came to mind when reading your question. You could check out the site though, maybe it is the right solution for you. On Wed, Oct 21, 2009 at 08:59, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Thanks Alot @ Jeff LaCoursiere,@Arjan Kroon,@Robin,@Joseph @ Jeff LaCoursiere Well you already suggested that you would send all files to server A, so A is your server Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . Can you give me some clues or links @ Arjan Kroon As i dont have good idea about Mounting what about the script actually i need some thing that dont needs human hand after development. And if script can do this then it will be fine. @Robin Which Application do use for that ?? Please elaborate Hell, you could even abuse dropbox for this purpose. What does this means? @ Joseph No Joseph its not some thing voice mail its recording of suggestions etc Actually operators are located at different locations and if a user leave a suggestion at one operator then the file will be on that particular server. But if the user of another operator want to listen that file then this file must be present on that server also ..Thats why I am considering these options On Wed, Oct 21, 2009 at 10:08 AM, Joseph syscon...@gmail.com wrote: On 10/20/09 17:24, ABBAS SHAKEEL wrote: Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C users come to listen the recorded voices (I am using agi ). any user records his profile on server B , NOW a user who make a call to server C cannot listen to profiles recorded at server B. Because these profiles reside on Server B ... Similar in case of server C. By ...listen to profile... do you mean retrieve their voice-mail on a different server? -- Joseph ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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 ___ -- 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] Syncronizing files on different Asteriskservers
I don't know if you server is running under Unix. If so, here is a wiki link about mounting http://en.wikipedia.org/wiki/Mount_%28Unix%29 Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens ABBAS SHAKEEL Verzonden: 21-10-2009 08:59 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [asterisk-users] Syncronizing files on different Asteriskservers Thanks Alot @ Jeff LaCoursiere,@Arjan Kroon,@Robin,@Joseph @ Jeff LaCoursiere Well you already suggested that you would send all files to server A, so A is your server Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . Can you give me some clues or links @ Arjan Kroon As i dont have good idea about Mounting what about the script actually i need some thing that dont needs human hand after development. And if script can do this then it will be fine. @Robin Which Application do use for that ?? Please elaborate Hell, you could even abuse dropbox for this purpose. What does this means? @ Joseph No Joseph its not some thing voice mail its recording of suggestions etc Actually operators are located at different locations and if a user leave a suggestion at one operator then the file will be on that particular server. But if the user of another operator want to listen that file then this file must be present on that server also ..Thats why I am considering these options On Wed, Oct 21, 2009 at 10:08 AM, Joseph syscon...@gmail.com wrote: On 10/20/09 17:24, ABBAS SHAKEEL wrote: Hello I need some advice regarding the Asterisk server that are located at different locations. Three asterisk servers are here each at different location. Suppose A,B,C be the three servers respectively. Server A is connected to server B and server C through a VPN. I have a developed an IVR service on server B and server C where users come and record their voice. On the same servers B and C users come to listen the recorded voices (I am using agi ). any user records his profile on server B , NOW a user who make a call to server C cannot listen to profiles recorded at server B. Because these profiles reside on Server B ... Similar in case of server C. By ...listen to profile... do you mean retrieve their voice-mail on a different server? -- Joseph ___ -- 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 -- Best Regards Shakeel Abbas ___ -- 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] error - sources for the 2.6.18-92.1.22.el5xen kernel
while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error; make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Entering directory `/usr/src/zaptel-1.4.12' echo You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12' make: *** [all] Error 2 i understand i have to install 2.6.18-92.1.22.el5xen kernel installed. How do i do this? Any help or guide will be highly appreciated. ___ -- 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] wrond DTMF detection on Zap channel
After a lot of debugging i have reproduced the error and the behaviour look me very strage: i've tried to change dtmfthereresold, vpmdtmfsupport and other kernel module settings without noting any significative change. But what i've notice (recording all the IVR calls and then listening the registration of the call) is that DTMF tones are not recognized by the system when the DTMF tone is clearly listenable in the audio recording!! Riassuming: good quality in voice and very low quality in the audio DTMF detected: the DTMF tone is recognized, is logged in che console (i've enabled dtmf log in full and console) and correctly detected by the AGI script good quality in voice and good quality in the audio DTMF detected: the DTMF tone is NOT recognized anything is logged in the console and the AGI script goes in timeout I've also upgraded asterisk to asterisk-1.4.26.2 dahdi-linux-complete-2.2.0.2 libpri-1.4.10.1 Any idea? On Tue, Oct 13, 2009 at 11:51 PM, nik600 nik...@gmail.com wrote: for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? -- /*/ nik600 http://www.kumbe.it ___ -- 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] Linksys 962
hi jeff, we use much of this phones, but i don't have seen such a symbol. The only thing i know is when you have an unregistered account (failed or not reachable) that the phone symbol has a red cross over it, which means its not online. Maybe on the phone a user pass has been set? best regards steve Jeff LaCoursiere schrieb: Working with a new client that has a ton of these phones, and in a new installation the phone is registered, can place and receive calls with no issues, but has a locked picture of a phone in the upper right corner. Any Linksys experts know what this means? I have searched the admin guide and googled to no results... really just an annoyance I suppose, but I would like to know what it means :) Cheers, j ___ -- 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 -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- 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] Searching on how to keep local calls... local
Hi list. Does anyone know how to keep calls between 2 local SIP-phones on the local private network when the 2 local IP-phones are registered to an online public Asterisk-server ?? What network-element / router do I need to install to prevent the RTP-traffic from flowing via the internet ? Config : Asterisk --internet-- router/firewall -- connected local IP-phones Internal call : IP-phone1 -- router/firewall --internet-- Asterisk --internet (back)-- router/firewall (back) -- IP-phone2 So I don't want an Asterisk server in my company (don't have appropriate place) and so I place the Asterisk-server in a datacentre. How about local calls going via the internet and back ?! Greetingz, Jonas. ___ -- 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] Searching on how to keep local calls... local
Your best option without a local asterisk server is to set up the remote server to do reinvites when calls are going local-local The calls will end up routed through your internet router, but not beyond that. Downside: might have to make each ip phone available via port forwards If you're really set against a local asterisk server, maybe try some other sip proxy software running on a small embedded computer (wrt54gl nslu2 ) On Wed, Oct 21, 2009 at 2:47 AM, jonas kellens jonas.kell...@telenet.bewrote: Hi list. Does anyone know how to keep calls between 2 local SIP-phones on the local private network when the 2 local IP-phones are registered to an online public Asterisk-server ?? What network-element / router do I need to install to prevent the RTP-traffic from flowing via the internet ? *Config :* Asterisk --internet-- router/firewall -- connected local IP-phones *Internal call :* *IP-phone1* -- router/firewall --internet-- *Asterisk* --internet (back)-- router/firewall (back) -- *IP-phone2* So I don't want an Asterisk server in my company (don't have appropriate place) and so I place the Asterisk-server in a datacentre. How about local calls going via the internet and back ?! Greetingz, Jonas. ___ -- 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 ___ -- 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] RAMDisk vs Extarnal server for recording
I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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] error - sources for the 2.6.18-92.1.22.el5xen kernel
Hi Just download tar.gz of your kernel version and extract into /usr/src/kernels/ directory ! -- Regards, Chandrakant Solanki On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE patricemb...@yahoo.comwrote: while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error; make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Entering directory `/usr/src/zaptel-1.4.12' echo You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12' make: *** [all] Error 2 i understand i have to install 2.6.18-92.1.22.el5xen kernel installed. How do i do this? Any help or guide will be highly appreciated. ___ -- 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 ___ -- 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] Calls hang up after 20 seconds
- Kevin P. Fleming kpflem...@digium.com ha scritto: | Da: Kevin P. Fleming kpflem...@digium.com | A: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com | Inviato: Lunedì, 19 ottobre 2009 14:03:53 | Oggetto: Re: [asterisk-users] Calls hang up after 20 seconds | | SIP wrote: | | In an ideal world, when Asterisk sent an ACK, whatever server/client | it | was connected to would respond accordingly. It is, however, not an | ideal | world, so this doesn't always happen. | | This is not correct; there are no responses to SIP ACK messages. In | addition. ACK messages are *required* for proper SIP operation; lack | of | an ACK to a response from Asterisk absolutely requires that Asterisk | assume that either the response was never delivered to the requester, | or | that that requester has stopped responding. In either case, the SIP | dialog/transaction in question must be terminated, because it is no | longer in a determinate state. | | If the SIP network does not route ACK responses properly, it is | broken. The SIP network from SIP server (ie EuteliaVoIP) to Asterisk? Internal network works correctly, internal calls are ok. Can I do something to favour the route of ACK responses with my firewall? Maybe opening, or forwarding something? Now port 5060 is opened in TCP and UDP, and ports from 1 to 2 are opened in UDP only. Another Asterisksm: if I restart Asterisk, initially calls works... after 1 or 2 hours every call hangs up after 20 seconds. Any suggestion would be appreciated. | | -- | Kevin P. Fleming | Digium, Inc. | Director of Software Technologies | 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA | skype: kpfleming | jabber: kpflem...@digium.com | Check us out at www.digium.com www.asterisk.org | | ___ | -- Bandwidth and Colocation Provided by http://www.api-digital.com -- | | AstriCon 2009 - October 13 - 15 Phoenix, Arizona | Register Now: http://www.astricon.net | | asterisk-users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users -- Gianni Fioretta - gianni.fiore...@yetopen.it YetOpen S.r.l. - http://www.yetopen.it/ Via Previati 72 - 23900 Lecco - ITALY - Tel 0341 220 205 - Fax 178 607 8199 D.Lgs. 196/2003 Si avverte che tutte le informazioni contenute in questo messaggio sono riservate ed a uso esclusivo del destinatario. Nel caso in cui questo messaggio Le fosse pervenuto per errore, La invitiamo ad eliminarlo senza copiarlo, a non inoltrarlo a terzi e ad avvertirci non appena possibile. Grazie. ___ -- 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] Searching on how to keep local calls... local
Your best option without a local asterisk server is to set up the remote server to do reinvites when calls are going local-local The calls will end up routed through your internet router, but not beyond that. So by placing canreinvite=yes in sip.conf, the RTP-traffic would flow between the 2 IP-phones and through the router. Do I loose music on hold ? I guess I do... Downside: might have to make each ip phone available via port forwards And if I place nat=yes in sip.conf ?? Or will IP-phone 1 not know the local IP-address of IP-phone 2 for sending a re-invite ?? Jonas. ___ -- 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] Need Help
Hi list, I am new to asterisk. I need help for installing and configure Asterisk IVR,OBD,IBD Server. We have a PRI line,I need to know what are the system requirements and hardware requirement for Asterisk *IVR*,*OBD*(Outbound dialer),*IBD*(Inbound dialer). Thanks and Regards, Kiran Reddy ___ -- 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] Need Help
Hi list, I am new to asterisk. I need help for installing and configure Asterisk IVR,OBD,IBD Server. We have a PRI line,I need to know what are the system requirements and hardware requirement for Asterisk *IVR*,*OBD*(Outbound dialer),*IBD*(Inbound dialer). Thanks and Regards, Kiran Reddy ___ -- 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] Need Help
Hi list, I am new to asterisk. I need help for installing and configure Asterisk IVR,OBD,IBD Server. We have a PRI line,I need to know what are the system requirements and hardware requirement for Asterisk *IVR*,*OBD*(Outbound dialer),*IBD*(Inbound dialer). Thanks and Regards, Kiran Reddy ___ -- 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] error - sources for the 2.6.18-92.1.22.el5xen kernel
Thanks solanki it worked fine. From: Chandrakant Solanki solanki.chandrak...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, October 21, 2009 1:45:42 PM Subject: Re: [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen kernel Hi Just download tar.gz of your kernel version and extract into /usr/src/kernels/ directory ! -- Regards, Chandrakant Solanki On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE patricemb...@yahoo.com wrote: while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error; make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Entering directory `/usr/src/zaptel-1.4.12' echo You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12' make: *** [all] Error 2 i understand i have to install 2.6.18-92.1.22.el5xen kernel installed. How do i do this? Any help or guide will be highly appreciated. ___ -- 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 ___ -- 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] RAMDisk vs Extarnal server for recording
There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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 ___ -- 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] RAMDisk vs Extarnal server for recording
Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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 ___ -- 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 ___ -- 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] TxFax works only with one of 2 PRI
Hi there,
I'm Using TxFAX to send faxes via Zaptel PRI. I have 2 PSTN PRI Providers,
with the first provider, all faxes are trasmited fine. With the second
provider, faxes can't be sent, we suspect about the setting of this PRI
provider, perhaps is doing some compression somewhere. Any suggestion
welcome.
I've tried these txfax calls with these results:
txfax(${FAXFILE}): Fax receive not successful - result (50) Disconnected
after permitted retries
txfax(${FAXFILE}|||ecm): Fax send not successful - result (51) The call
dropped prematurely.
txfax(${FAXFILE}|caller|debug|): Fax receive not successful - result (13)
Unexpected message received.
For detailed logs please take a look of http://www.pastebin.ca/1634790
Cordialmente,
Martin Cabrera
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Re: [asterisk-users] IMAP voicemail using subfolders fails.
Kevin P. Fleming wrote: It's not present in the current 1.4 doc/imapstorage.txt file, or any later version. I don't even know why the storage format would matter, since that would be very specific to the IMAP server that is managing that folder. Hmmm http://markmail.org/message/up3rfmdk2kjf6r7y is a link that contains the contents of a README file that looks like it came from Digium. About half-way down is: -- Mailbox Format -- Mailboxes should use the mbx mailbox format. The mbox format does not support concurrent access to mailboxes, which can cause deadlock or strange behaviors. You can convert mailboxes from mbox to mbx using mailutil: Perhaps that came from a different product? I think that I'm going to just go ahead and implement IMAP VM and see what happens. Thanks very much Kevin! Regards, Barry ___ -- 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] ChannelStateDesc: Ring ?
Martin a écrit : Ring is the state when the device sent 100 Trying after INVITE When it actually sends 180 Ringing or gets the progress or so message from another channel (when used with Dial) then the status changes to Ringing Humm. OK. So basically, it's Intended to ring... Thanks for the info. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- 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] ringing... or lack thereof
Want to make sure I understand why a caller might not hear ringing when outbound calling. A SIP phone is behind a firewall and is registered to an asterisk server on a public network. Sometimes (but not always) when placing an outbound call there is no ringing before the remote party answers. Its not that the remote party picks up very quickly - the delay may be as long as twenty seconds before it is answered (and I must assume that the remote phone had rung at least a few times). I vaguely understand that part of the SIP call setup includes a ringing message, sent from asterisk to the originating phone. If this is correct and the firewall for whatever reason isn't passing this message, will there be no ringing sound on the originating phone? This is confusing to me, as I kind of assumed that the ringing sound was in audio, and would be part of the RTP stream. But perhaps that isn't even flowing yet. Guess I am showing my SIP ignorance. Please enlighten ;) Cheers, j ___ -- 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] Linksys 962
On Tue, 20 Oct 2009, Jimmy Godbout wrote: Can you send a picture of this ? Thanks -Original Message- From: j...@jeff.net Sent: Tue, 20 Oct 2009 23:34:13 + (UTC) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Linksys 962 Working with a new client that has a ton of these phones, and in a new installation the phone is registered, can place and receive calls with no issues, but has a locked picture of a phone in the upper right corner. Any Linksys experts know what this means? I have searched the admin guide and googled to no results... really just an annoyance I suppose, but I would like to know what it means :) Cheers, j I will try to get one - the client is actually at a remote site and this is all so far hearsay :) j ___ -- 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] Linksys 962
On Wed, 21 Oct 2009, Stefan Schmidt wrote: hi jeff, we use much of this phones, but i don't have seen such a symbol. The only thing i know is when you have an unregistered account (failed or not reachable) that the phone symbol has a red cross over it, which means its not online. Maybe on the phone a user pass has been set? best regards steve Jeff LaCoursiere schrieb: Working with a new client that has a ton of these phones, and in a new installation the phone is registered, can place and receive calls with no issues, but has a locked picture of a phone in the upper right corner. Any Linksys experts know what this means? I have searched the admin guide and googled to no results... really just an annoyance I suppose, but I would like to know what it means :) Cheers, j I am going to try to get a picture taken of this odd icon, since I haven't actually seen it myself yet. It may become obvious once I have... Its not that the phone isn't registered - in fact it doesn't seem to stop them from using the phone at all... Cheers, j ___ -- 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] AMI 1.0 - 1.1 with originate.
Miguel Molina a écrit : Guillaume Yziquel escribió: So what is this permission issue? Where are the changes from 1.0 to 1.1 documented? When I was testing asterisk 1.6.0.X with the AMI Originate action, I fell into the same issue as you. I found that it was that the permissions now are more fine-grained, and to have the ability to originate a call you need to set additional write permissions compared to the 1.4.X AMI. When I put the originate permission on the write settings of my AMI user, everything went fine. To find more documentarion about the changes from AMI 1.0 to 1.1 take a look of these files on your asterisk source code: UPGRADE-1.6.txt doc/manager_1_1.txt Hope it solves your issue. It pretty well did. Thanks a lot. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- 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] Need Help
Hi list, I am new to asterisk. I need help for installing and configure Asterisk IVR,OBD,IBD Server. We have a PRI line,I need to know what are the system requirements and hardware requirement for Asterisk *IVR*,*OBD*(Outbound dialer),*IBD*(Inbound dialer). Thanks and Regards, Kiran Reddy ___ -- 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] Need Help
On Wed, 21 Oct 2009, kiran.re...@mpowerglobal.in wrote: I am new to asterisk. I need help for installing and configure Asterisk IVR,OBD,IBD Server. 4 posts in 3 hours? 1) Don't repost, you just annoy people that may have helped you. 2) Ask specific questions, not I know nothing, please tell me everything. 3) Use a meaningful question. You may attract the interest of someone who can help you. 4) Help yourself. Use google. Read TFOT.pdf. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Astricon
Or charge for full access! Leave a few teasers, and charge some amount to see them all. I would pay - even close to attendance price... could only help you get past break even ;) I agree, I would be quite willing to pay for full access to all the videos from the Conference. Bob ___ -- 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] Linksys 962
I am going to try to get a picture taken of this odd icon, since I haven't actually seen it myself yet. It may become obvious once I have... Its not that the phone isn't registered - in fact it doesn't seem to stop them from using the phone at all... Just because they can use the phone doesn't mean the other 5 lines are registered. I bet one of those lines is the one with the actual odd icon. It is probably a phone with a red cross. I have a SPA962 and its exactly what shows under a specific line that cannot register. Andres http://www.neuroredes.com Cheers, j ___ -- 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 ___ -- 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] troubleshooting NAT
Date: Tue, 20 Oct 2009 21:02:29 -0500 From: asteriskl...@callthem.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT if you're using SIP then you look at SIP headers ... SDP part from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP Here is the SIP header that you see when you run the asterisk -r command. Reliably Transmitting (NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport From: Unknown sip:unkn...@ourpublicip;tag=as0186791c To: sip:216.82.224.202 Contact: sip:unkn...@ourpublicip Call-ID: 52019c8970f8727a04fd79f0083cc...@ourpublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 13:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Here is a debug from one of our phones calling an external number SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.46:5060;branch=z9hG4bKb033ebc8ff0de35dc.650da238f5237c4a1;received=10.0.0.46 From: me sip:1...@10.1.0.8;tag=aa5daa3277 To: 95457878 sip:95457...@10.1.0.8;tag=as0b5e19fc Call-ID: 2edce254de2a77ab CSeq: 32330 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:95457...@10.1.0.8 Content-Length: 0 == Spawn extension (from-internal, 95457878, 4) exited non-zero on 'SIP/117-09c4fc20' -- Executing [...@from-internal:1] Macro(SIP/117-09c4fc20, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] GotoIf(SIP/117-09c4fc20, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [...@macro-hangupcall:4] GotoIf(SIP/117-09c4fc20, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [...@macro-hangupcall:7] GotoIf(SIP/117-09c4fc20, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] Hangup(SIP/117-09c4fc20, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/117-09c4fc20' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/117-09c4fc20' and then you can try to get some packet dump with tcpdump/wireshark if am ssh into the server and run tcpdump not port 22. i get normal LAN traffic until i make a call. then i get a ton of this. .8 is the phoneserver and .46 is one of the phones. i haven't done wireshark because I haven't looked up how to take the tcpdump and import it into wireshark. 09:40:58.510750 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.530758 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.550762 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.570770 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.590775 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.610781 IP 10.1.0.8.12036 10.1.0.46.hbci: UDP, length 172 09:40:58.625026 IP 10.1.0.46.sip 10.1.0.8.sip: SIP, length: 348 09:40:58.625485 IP 10.1.0.8.sip 10.1.0.46.sip: SIP, length: 417 09:40:58.625608 IP 10.1.0.8.sip 10.1.0.46.sip: SIP, length: 435 09:40:58.679832 IP 10.1.0.46.sip 10.1.0.8.sip: SIP, length: 334 and maybe configure your router so it works it's the first thing to look for ... if the phone server can access the internet then shouldn't that mean the router has NAT setup correctly on it? you can also try to use the stun server ... asterisk has it built in ...never used it but saw it's there Martin On Tue, Oct 20, 2009 at 1:32 PM, Ott Rose sixfourimp...@hotmail.com wrote: Can anyone tell me how to troubleshoot NAT issues? We had Freepbx look at your install and they said we are having a NAT problem but didn'ttell me if it was with the asterisk conf or the Cisco ASA. Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ -- 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 ___ -- 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 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. http://clk.atdmt.com/GBL/go/171222985/direct/01/___ -- 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] Astricon
On Wed, Oct 21, 2009 at 4:01 PM, Bob Pierce pier...@westmancom.com wrote: Or charge for full access! Leave a few teasers, and charge some amount to see them all. I would pay - even close to attendance price... could only help you get past break even ;) I agree, I would be quite willing to pay for full access to all the videos from the Conference. I missed the first part of this, but has anyone said: not all the presentations were recorded. /r ___ -- 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] troubleshooting NAT
Here is what i think the is helpful from wireshark OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport From: Unknown sip:unkn...@mypublicip;tag=as7b5287b3 To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 311d57516ef9649b7dfab93720393...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as7b5287b3 To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340 Call-ID: 311d57516ef9649b7dfab93720393...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport From: Unknown sip:unkn...@mypublicip;tag=as20c07cef To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 09003fa1042464842df21c73339a1...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as20c07cef To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e Call-ID: 09003fa1042464842df21c73339a1...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport From: Unknown sip:unkn...@mypublicip;tag=as271c263c To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as271c263c To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4 Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport From: Unknown sip:unkn...@mypublicip;tag=as3913f8ae To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 05e50acb725d34f01bc78666422f5...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as3913f8ae To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790 Call-ID: 05e50acb725d34f01bc78666422f5...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 From: sixfourimp...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 21 Oct 2009 14:00:20 + Subject: Re: [asterisk-users] troubleshooting NAT Date: Tue, 20 Oct 2009 21:02:29 -0500 From: asteriskl...@callthem.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT if you're using SIP then you look at SIP headers ... SDP part from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP Here is the SIP header that you see when you run the asterisk -r command. Reliably Transmitting (NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport From: Unknown sip:unkn...@ourpublicip;tag=as0186791c To: sip:216.82.224.202 Contact: sip:unkn...@ourpublicip Call-ID: 52019c8970f8727a04fd79f0083cc...@ourpublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 13:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Here is a debug from one of our phones calling an external number SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.46:5060;branch=z9hG4bKb033ebc8ff0de35dc.650da238f5237c4a1;received=10.0.0.46 From: me sip:1...@10.1.0.8;tag=aa5daa3277 To: 95457878 sip:95457...@10.1.0.8;tag=as0b5e19fc Call-ID: 2edce254de2a77ab CSeq: 32330 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:95457...@10.1.0.8 Content-Length: 0 == Spawn extension (from-internal, 95457878, 4) exited non-zero on 'SIP/117-09c4fc20' -- Executing [...@from-internal:1] Macro(SIP/117-09c4fc20,
[asterisk-users] polarity on some channels
Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks ___ -- 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] Intermittent Low volume
Just looking for some ideas here... Single office with 1.4.26.2 - Frontend 1.4.26.2 w/sangoma A108 Gateway I have been getting a few complaints about caller cant hear me or I cant hear the caller I've listened to the recordings and can verify what they are complaining about, with this being said, most calls are fine. I know there are alot of issues that can happen once the call leaves the office that I will never be able to address (vonage, cell phones, etc) but I am trying to see if there is anything I could do to help alleviate the issue on my end. I never really messed with rxgain and txgain and was starting to play with dahdi_monitor to see my gain levels... Do you all think this could be a gain level issue? Thanks for any input ___ -- 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] RAMDisk vs Extarnal server for recording
Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] RAMDisk vs Extarnal server for recording
Hi Matt, ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a vicidial system. Anyway, the minimum length is 10-20 seconds, maximum can get as long as 15-20 minutes, and on average it's about 2-5 minutes, depending on the campaign. The server is now doing everything btw, but I'm going to dedicate it to only handle calling and recording. The rest (database and http) will be moved to other servers, which might help a bit too. *Off topic*: the company I work for went bankrupt a few months ago, but is back in business and we are making heavy use of vicidial (awesome stuff). Going to do loads of work on it, so hope to give loads of (usefull) code to the vicidial project by the end of the year. Looking forward to it! On Wed, Oct 21, 2009 at 17:11, Matt Florell astma...@gmail.com wrote: Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] troubleshooting NAT
Have a quick look at this guide on NAT and SIP - http://www.aocomputing.net/?p=3. This is the link given if you were to ask this same question in the IRC channel... --wcs On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is what i think the is helpful from wireshark OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport From: Unknown sip:unkn...@mypublicip;tag=as7b5287b3 To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 311d57516ef9649b7dfab93720393...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060 From: Unknown sip:unkn...@10.1.0.8 sip%3aunkn...@10.1.0.8 ;tag=as7b5287b3 To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340 Call-ID: 311d57516ef9649b7dfab93720393...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport From: Unknown sip:unkn...@mypublicip;tag=as20c07cef To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 09003fa1042464842df21c73339a1...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060 From: Unknown sip:unkn...@10.1.0.8 sip%3aunkn...@10.1.0.8 ;tag=as20c07cef To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e Call-ID: 09003fa1042464842df21c73339a1...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport From: Unknown sip:unkn...@mypublicip;tag=as271c263c To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060 From: Unknown sip:unkn...@10.1.0.8 sip%3aunkn...@10.1.0.8 ;tag=as271c263c To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4 Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport From: Unknown sip:unkn...@mypublicip;tag=as3913f8ae To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 05e50acb725d34f01bc78666422f5...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060 From: Unknown sip:unkn...@10.1.0.8 sip%3aunkn...@10.1.0.8 ;tag=as3913f8ae To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790 Call-ID: 05e50acb725d34f01bc78666422f5...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 -- From: sixfourimp...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 21 Oct 2009 14:00:20 + Subject: Re: [asterisk-users] troubleshooting NAT Date: Tue, 20 Oct 2009 21:02:29 -0500 From: asteriskl...@callthem.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT if you're using SIP then you look at SIP headers ... SDP part from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP Here is the SIP header that you see when you run the asterisk -r command. Reliably Transmitting (NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport From: Unknown sip:unkn...@ourpublicip;tag=as0186791c To: sip:216.82.224.202 Contact: sip:unkn...@ourpublicip Call-ID: 52019c8970f8727a04fd79f0083cc...@ourpublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 13:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Here is a debug from one of our phones calling an external number SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.46:5060
Re: [asterisk-users] RAMDisk vs Extarnal server for recording
Hello, Yep, I'm the ViciDial Guy :) In our most recent release we do have some instructions in the SCRATCH_INSTALL.txt doc on setting up a tmpfs partition for recording. 8GB should be fine for the 60 concurrent recordings under the times you gave, although with MySQL and Apache/PHP you may run into issues, so I would recommend moving MySQL/Apache/PHP off to a different server ASAP. Thanks for the compliments! MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Hi Matt, ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a vicidial system. Anyway, the minimum length is 10-20 seconds, maximum can get as long as 15-20 minutes, and on average it's about 2-5 minutes, depending on the campaign. The server is now doing everything btw, but I'm going to dedicate it to only handle calling and recording. The rest (database and http) will be moved to other servers, which might help a bit too. Off topic: the company I work for went bankrupt a few months ago, but is back in business and we are making heavy use of vicidial (awesome stuff). Going to do loads of work on it, so hope to give loads of (usefull) code to the vicidial project by the end of the year. Looking forward to it! On Wed, Oct 21, 2009 at 17:11, Matt Florell astma...@gmail.com wrote: Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] DAHDI: TCM PCI Master abort
I'm assuming this is an issue with DAHDI. I am running asterisk 1.4.26 on Fedora 11 with dahdi-linux kernel modules 2.2.0.2-65 (both from ATrpms). I have a Wildcard TDM400P REV I (4 modules) with one POTS line and three local extensions (never can remember which is FXS and which is FXO )-: and a couple of SIP phones; small home system. About once or twice a month this happens. What I observe is that the system is comatose. I can switch via CTRL-ALT-F2 to a console window and I can see the TCM PCI Master abort messages whizzing past. They are also written to the syslog over and over and over and over and... until the file system fills up and Bad Things (tm) happen. The only way to get the system back at this point is to do a hard reset, wait for it to come back, remove the /var/log/messages file, then reboot again (I tried just restarting syslog after removing the file but due to things other than asterisk on the system, this isn't enough). Has anyone else seen this? Is it due to some error in my DAHDI configuration? Thanks, --Greg ___ -- 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] RAMDisk vs Extarnal server for recording
I'm on it, going to get me some new hardware tomorrow and hope to have it up and running early next week. tnx! On Wed, Oct 21, 2009 at 17:42, Matt Florell astma...@gmail.com wrote: Hello, Yep, I'm the ViciDial Guy :) In our most recent release we do have some instructions in the SCRATCH_INSTALL.txt doc on setting up a tmpfs partition for recording. 8GB should be fine for the 60 concurrent recordings under the times you gave, although with MySQL and Apache/PHP you may run into issues, so I would recommend moving MySQL/Apache/PHP off to a different server ASAP. Thanks for the compliments! MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Hi Matt, ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a vicidial system. Anyway, the minimum length is 10-20 seconds, maximum can get as long as 15-20 minutes, and on average it's about 2-5 minutes, depending on the campaign. The server is now doing everything btw, but I'm going to dedicate it to only handle calling and recording. The rest (database and http) will be moved to other servers, which might help a bit too. Off topic: the company I work for went bankrupt a few months ago, but is back in business and we are making heavy use of vicidial (awesome stuff). Going to do loads of work on it, so hope to give loads of (usefull) code to the vicidial project by the end of the year. Looking forward to it! On Wed, Oct 21, 2009 at 17:11, Matt Florell astma...@gmail.com wrote: Hello, We use RAM to record to on almost all systems we set up, although we usually use tmpfs, instead of a fixed RAM drive, because it is more flexible. The number of recordings you can handle is dependant on how long the calls are. What would your average, minimum, maximum recording lengths be? We usually do not do more than 100 concurrent recordings on a single server, but we have done up to 250 before successfully. MATT--- On 10/21/09, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? btw, I found this thread somewhere: http://lists.digium.com/pipermail/asterisk-users/2005-October/120930.html, but this is rather old info. Is this documentation still usefull? And if not, do you happen to have any idea/url/doc where I can find a bit less old info? thanks, robin On Wed, Oct 21, 2009 at 13:21, Zoa zoach...@securax.org wrote: There are 2 issues i think, one is the seek time on harddisks and the lack of a big buffer in Asterisk (saving 10 streams at the same time will cause a lt of random writes). The other one is the interrupts being taken up by the harddisk. So an SSD might help, saving to an network drive might help (it moves the issue to another server, where it might not cause a problem), buffering to ram (but you will lack space). The best solution depends on your exact hardware and the amount of writes you want to do. Buffering to a ramdrive before moving it over NFS seems like the best idea to me. Zoa Robin wrote: I'm having loads of problems with recordings, as in crappy audio quality and lost pieces of the recordings. I've been searching for a solution and the solutions i find on the interwebs include a ramdisk, for local recording, or another machine, handling the recording. I guess the ramdisk would be the easy solution and the external machine would be little harder to set up. I do actually prefer the external machine, but i'm not exaclty sure how to set that one up... The reason I prefer the external machine, is that the recording have to be moved to an external machine anyway. Although I've come across a post somewhere, talking about recording to ramdisk and then move the files over a crosscable directly to another disk over 1000mbit. Which sound nice as well... What do you advise for bringing serverload down and get rid of the harddisk bottleneck? Is a ramdisk a better solution then an external machine? And if so, why? Sorry about this pro-con question, but I cannot find an answer which compares these pro-cons anywhere. thanks, robin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To
Re: [asterisk-users] troubleshooting NAT
i changed my sip_nat.conf file following the steps in that link. Still didn't work same debug info Date: Wed, 21 Oct 2009 10:33:18 -0500 From: wcse...@selbytech.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT Have a quick look at this guide on NAT and SIP - http://www.aocomputing.net/?p=3. This is the link given if you were to ask this same question in the IRC channel... --wcs On Wed, Oct 21, 2009 at 9:59 AM, Ott Rose sixfourimp...@hotmail.com wrote: Here is what i think the is helpful from wireshark OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK5aac5845;rport From: Unknown sip:unkn...@mypublicip;tag=as7b5287b3 To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 311d57516ef9649b7dfab93720393...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK5aac5845;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as7b5287b3 To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.e340 Call-ID: 311d57516ef9649b7dfab93720393...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK2e9c787f;rport From: Unknown sip:unkn...@mypublicip;tag=as20c07cef To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 09003fa1042464842df21c73339a1...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK2e9c787f;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as20c07cef To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.f75e Call-ID: 09003fa1042464842df21c73339a1...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 64.191.130.78:5060;branch=z9hG4bK4d179421;rport From: Unknown sip:unkn...@mypublicip;tag=as271c263c To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK4d179421;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as271c263c To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.25d4 Call-ID: 30a25dbd0ed352e5159b0b6a767ea...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP mypublicip:5060;branch=z9hG4bK61a8b371;rport From: Unknown sip:unkn...@mypublicip;tag=as3913f8ae To: sip:216.82.224.202 Contact: sip:unkn...@mypublicip Call-ID: 05e50acb725d34f01bc78666422f5...@mypublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 14:11:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.0.8:5060;branch=z9hG4bK61a8b371;rport=5060 From: Unknown sip:unkn...@10.1.0.8;tag=as3913f8ae To: sip:216.82.224.202;tag=f5da119de3db22dcaa2abb8ea9fec0ce.2790 Call-ID: 05e50acb725d34f01bc78666422f5...@10.1.0.8 CSeq: 102 OPTIONS Server: Bandwidth.com TRM (bw7.gold.13) Content-Length: 0 From: sixfourimp...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 21 Oct 2009 14:00:20 + Subject: Re: [asterisk-users] troubleshooting NAT Date: Tue, 20 Oct 2009 21:02:29 -0500 From: asteriskl...@callthem.info To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] troubleshooting NAT if you're using SIP then you look at SIP headers ... SDP part from INVITE's and 200 OK to INVITE. You check what IP/port is used for RTP Here is the SIP header that you see when you run the asterisk -r command. Reliably Transmitting (NAT) to 216.82.224.202:5060: OPTIONS sip:216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP ourpublicip:5060;branch=z9hG4bK42ef5901;rport From: Unknown sip:unkn...@ourpublicip;tag=as0186791c To: sip:216.82.224.202 Contact: sip:unkn...@ourpublicip Call-ID: 52019c8970f8727a04fd79f0083cc...@ourpublicip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Oct 2009 13:33:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Here is a debug from one of our phones calling an external number SIP/2.0 200 OK Via: SIP/2.0/UDP
[asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?
Hi, Siemens Gigaset line of products include an integrated web browser with which firmware download is possible. The trouble is you need to provide Internet access. We use a couple of these boxes in LANs not connected to Internet for security reasons. So I would prefer to download firmware upgrades from my own TFTP or HTTP server. Thanks to Wireshark, I could list downloaded files list. For instance, latest C450IP files include : http://gigaset.siemens.com/chagall/1/0/master.bin http://gigaset.siemens.com/chagall/1/0/../baselines.bin http://gigaset.siemens.com/chagall/1/0/../chagall072_01.bin All these files can be copied (using a wget command) and copied to a personal web server but information is missing to extend this process to each model. Any idea ? Reagrds ___ -- 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 - Gigaset Chagall - How to download firmware without Internet access ?
Olivier wrote: Hi, Siemens Gigaset line of products include an integrated web browser with which firmware download is possible. The trouble is you need to provide Internet access. We use a couple of these boxes in LANs not connected to Internet for security reasons. So I would prefer to download firmware upgrades from my own TFTP or HTTP server. Thanks to Wireshark, I could list downloaded files list. For instance, latest C450IP files include : http://gigaset.siemens.com/chagall/1/0/master.bin http://gigaset.siemens.com/chagall/1/0/../baselines.bin http://gigaset.siemens.com/chagall/1/0/../chagall072_01.bin All these files can be copied (using a wget command) and copied to a personal web server but information is missing to extend this process to each model. Change your local nameserver to resolve the address to a private IP instead of to the public IP? Leif! ___ -- 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] Syncronizing files on different Asterisk servers
Have you considered rsync? We use it to synchronize voicemail between offices connected through a VPN. Of course you need to run rsync somehow, which is easy with an external command every time someone checks their voice mail, but no reason it couldn't be done with a cron job. Sincerely, Brent A. Torrenga Sorry For the wording actually i need to send to a central server. then a central server to all others. Because all servers have VPN To central Server only. The Drive Mount Option seems cool to me but I dont have any Idea About it . ___ -- 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] Astricon
Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry ___ -- 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] Searching on how to keep local calls... local
Your best option without a local asterisk server is to set up the remote server to do reinvites when calls are going local-local The calls will end up routed through your internet router, but not beyond that. So by placing canreinvite=yes in sip.conf, the RTP-traffic would flow between the 2 IP-phones and through the router. Do I loose music on hold ? I guess I do... Try it first, asterisk could just reinvite the audio back to the server Also you might be able to program a SIP address for music on hold into the ip phones exten = moh,1,Answer() exten = moh,2,MusicOnHold() Downside: might have to make each ip phone available via port forwards And if I place nat=yes in sip.conf ?? Or will IP-phone 1 not know the local IP-address of IP-phone 2 for sending a re-invite ?? The remote asterisk server would be doing the reinvites with what it knows Jonas. ___ -- 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 ___ -- 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] DS3 capacity calls using asterisk
Thanks for the information, I will look into both cisco and adtran see which would be helpful On Thu, Oct 15, 2009 at 4:09 PM, Alex Balashov abalas...@evaristesys.comwrote: David Backeberg wrote: On Thu, Oct 15, 2009 at 4:58 PM, David Backeberg dbackeb...@gmail.com wrote: There's no one-step solution I'm aware of. Cisco sells something called an AS5300 that supposedly can terminate a DS3 and convert it all to SIP. Otherwise, you need a channel bank like the Adtran MX2800 I was close, but incorrect. Cisco sells the 5XXX series, but I think the AS5300 has a lower capacity that a full DS3. The 58xx series claims to terminate multiple DS3s. I've never played with anything nicer than a Cisco 3845, which maxes out at 24T1s, just shy of what you can get out of the Adtran MX 2800. Yes, the AS5300 chassis can only do 4 T1s. You're looking for an AS5400, or another big router chassis that can take a DS3 adaptor and VFCs (like a 7200). -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon
On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote: Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry And I was in the one that wasn't. So I guess I'll have to summarize... except I was a sleep one of the days :) /r ___ -- 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] Astricon
Is THAT a summary :)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Wednesday, October 21, 2009 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Astricon On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote: Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry And I was in the one that wasn't. So I guess I'll have to summarize... except I was a sleep one of the days :) /r ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120 connect_timeout=80 table_cache=1024 thread_concurrency=8 long_query_time=10 tmp_table_size=64M join_buffer_size=1M thread_cache_size=200 key_buffer=32M table_cache=1024 sort_buffer_size=2M read_buffer_size=2M read_rnd_buffer_size=4M And I am running on asterisk 1.4.22.1, Quadcore processor 2.4Ghz, 4GB RAM, mysql 5.0. Some times we get dead air even after 50-100 calls. Is there any other additional parameters or variables or resources (hardware) to be looked into to increase the speed of mysql connections? Your advice is really appreciated. Thanks Sandesh. ___ -- 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] RAMDisk vs Extarnal server for recording
On Wed, Oct 21, 2009 at 7:36 AM, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? I'm doing ramdisk recordings of about the same number of streams you're talking, in 4GB. I move out completed recordings once every 15 minutes or so via NFS, and as such, I never use very much of the ramdisk. There's no rule that says you have to use the whole 4GB of ram for recordings. I'm probably staying below 100MB or so. Strictly speaking, I'm using both ramdisk and external server, but the external server is just a centralized system with larger disks. However, I know that this arrangement isn't working for my load which is about to double again, so I'm upgrading to better hardware (and maintaining the status quo with my asterisk arrangement) If you read every single title of asterisk-users in the last few months, you'll find a similar discussion on this topic which went through the pros and cons of ramdisk versus centralized server. Somebody at that time mentioned particular names of programs that can do the centralized recordings by doing network hardware level replication and picking off the SIP packets. I've never done this, but if you find that mailing list thread you'll be able to find names of people who say they've done that. ___ -- 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] Asterisk and Nuance Vocalizer TTS Engine
Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran ___ -- 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] polarity on some channels
B.Masoud @ SH wrote: Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks ___ -- 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 Try turning off callerid. The 'standard' for POTS lines in the US is to put the caller id in between ring1 ring2. Asterisk waits for callerid before answering the line by default. usecallerid=off ___ -- 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 and Nuance Vocalizer TTS Engine
According to asterisk-guru this has been done. If you're just looking for TTS and not voice recognition, this shouldn't be too problematic. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vela Sivasankaran Sent: Wednesday, October 21, 2009 1:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk and Nuance Vocalizer TTS Engine Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran ___ -- 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 and Nuance Vocalizer TTS Engine
Hi, I think you should use the nvcmdline utility to synthesize your prompt to a certain file to be specified. Afterwards, you could play that on your asterisk, for example a wav file. But this could be some kind of long lasting as the TTS synthesizes in realtime, i.e. the longer the prompt is the longer you have to wait for the file to play. So, using AGI should be worthwile to take a look at. Using the nvcmdline utility you should use bash AGI or something more scripty. If there is a Java API for Nuance Vocalizer (I do not know that) you also could use that. Regards, Christophorus Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran ___ -- 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 ___ -- 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] RAMDisk vs Extarnal server for recording
On 10/21/09, David Backeberg dbackeb...@gmail.com wrote: On Wed, Oct 21, 2009 at 7:36 AM, Robin ro...@zoap.org wrote: Thanks for your response. The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)... But memory is rather cheap nowadays. If i'd buf up the server with 8 extra gigs for use as a ramdrive, do you think that might be enough to record between 30-60 simultanious streams? Or should it be way more? I'm doing ramdisk recordings of about the same number of streams you're talking, in 4GB. I move out completed recordings once every 15 minutes or so via NFS, and as such, I never use very much of the ramdisk. There's no rule that says you have to use the whole 4GB of ram for recordings. I'm probably staying below 100MB or so. Strictly speaking, I'm using both ramdisk and external server, but the external server is just a centralized system with larger disks. However, I know that this arrangement isn't working for my load which is about to double again, so I'm upgrading to better hardware (and maintaining the status quo with my asterisk arrangement) If you read every single title of asterisk-users in the last few months, you'll find a similar discussion on this topic which went through the pros and cons of ramdisk versus centralized server. Somebody at that time mentioned particular names of programs that can do the centralized recordings by doing network hardware level replication and picking off the SIP packets. I've never done this, but if you find that mailing list thread you'll be able to find names of people who say they've done that. We have a few clients that use Oreka(from OrecX) that does network-based SIP packet-capture recording. It works very well on their multi-server setups and the core of Oreka is Open Source. MATT--- ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 and Nuance Vocalizer TTS Engine
On Wed, 21 Oct 2009, Christophorus Laube wrote: Using the nvcmdline utility you should use bash AGI or something more scripty. I'd suggest something way less scripty like C and a proper API if available. You can execute xxx AGIs written in C in the time it takes a PHP or Perl interpreter to load. (I haven't benched bash.) Executing agi() creates a process. Every shell command you execute creates a process. Creating all these processes is not free. Creating a process takes a huge amount of resources (time, CPU, memory, disk). Using a proper API eliminates all the shell command nonsense. Recently, somebody posted a shell script to extract lines from a queue log and warned that it could take a long time to execute. Re-coding the script in C reduced the execution time to almost 1/3,000th of the original execution time. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Unstable PRI interface: Link restart after few min::
Hello Team I have connected * running centos 5.2, asterisk 1.6.1 dahdi 2.1 to the telco but the link is very unstable (D-Channel restart after some few min) Below please find part of 'pri intensive debug span 2' for your advice. Looks like telco is sending disconnect request but cant establish reason for this Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 002 P/F: 1 0 bytes of data INV-VOICESW01*CLI [ 00 01 01 05 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 002 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 1 to (but not including) 2 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Active q931.c:3015 q931_disconnect: call 6321 on channel 1 enters state 11 (Disconnect Request) [ 00 01 04 04 08 02 98 b1 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 002 0: 0 N(R): 002 P: 0 9 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6321/0x18B1) (Terminator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Destroying the call, ourstate Disconnect Request, peerstate Disconnect Indication -- Hungup 'DAHDI/1-1' == Spawn extension (from-outside, 0222112211, 3) exited non-zero on 'DAHDI/32-1' == End MixMonitor Recording DAHDI/32-1 -- Hungup 'DAHDI/32-1' -- T200 counter expired, What to do... -- Retransmitting 13 bytes [ 00 01 04 05 08 02 98 b1 45 08 02 81 90 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 002 0: 0 N(R): 002 P: 1 9 bytes of data -- Rescheduling retransmission (1) INV-VOICESW01*CLI [ 00 01 01 07 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 003 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 1 to (but not including) 3 -- ACKing packet 2, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer INV-VOICESW01*CLI [ 02 01 04 06 08 02 18 b1 4d ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 002 0: 0 N(R): 003 P: 0 5 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 2 to (but not including) 3 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 6321/0x18B1) (Originator) Message type: RELEASE (77) -- Making new call for cr 6321 [ 00 01 06 06 08 02 98 b1 5a 08 02 81 d1 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 003 0: 0 N(R): 003 P: 0 9 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6321/0x18B1) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Restarting T203 timer INV-VOICESW01*CLI [ 00 01 01 08 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 004 P/F: 0 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 2 to (but not including) 4 -- ACKing packet 3, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Restarting T203 timer INV-VOICESW01*CLI Disconnected from Asterisk server [r...@inv-voicesw01 asterisk]# = The maximum call duration I have made so far is 3min Kind regards Sam ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On Wed, Oct 21, 2009 at 2:30 PM, das sandesh sandesh...@gmail.com wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: My guess is DNS taking a long time to timeout? Trying changing the connection string to use straight ip address rather than hostname. ___ -- 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] Astricon
Sounds like it wasn't a very interesting track. ;) N. Danny Nicholas wrote: Is THAT a summary :)? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Wednesday, October 21, 2009 1:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Astricon On Wed, Oct 21, 2009 at 7:46 PM, Barry L. Kline blkl...@attglobal.net wrote: Randy R wrote: I missed the first part of this, but has anyone said: not all the presentations were recorded. Hi Randy. Yes, that was mentioned. Actually, three of the four tracks were videotaped IIRC. Barry And I was in the one that wasn't. So I guess I'll have to summarize... except I was a sleep one of the days :) /r ___ -- 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 ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Hi Steve, Thanks for your reply. I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? Regards Sandesh. On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
I think the key point is how many calls per second. That's what mysql is concerned about. Other than that it is just asterisk. Did you monitor the mysql, try log-slow-queries and set the time to 1 second. -Jai On Wed, Oct 21, 2009 at 12:57 PM, das sandesh sandesh...@gmail.com wrote: Hi Steve, Thanks for your reply. I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? Regards Sandesh. On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot oftime for execution
Not my cup of tea, but I think I'd be trying an ODBC connection to reduce some overhead here. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of das sandesh Sent: Wednesday, October 21, 2009 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Concurrent calls including mysql taking lot oftime for execution Hi Steve, Thanks for your reply. I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? Regards Sandesh. On Wed, Oct 21, 2009 at 2:21 PM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 21 Oct 2009, das sandesh wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: This isn't a MySQL performance list and I'm not an expert, but... I cobbled up a little C program that created 1,000 concurrent connections to my database and it takes 0.15 seconds on an AMD Phenom(tm) 8650 Triple-Core Processor. I confirmed via netstat that there were 1,000 connections. Opening and closing a single connection 1,000 times was still less than a second. This was connecting to localhost so it used the UNIX socket. Changing to a TCP socket took 0.19 seconds. I'd look elsewhere -- it's not the MySQL connection that's the problem. How are you connecting? Is in in an AGI? What language are you using? What are you doing with MySQL? A few more details will help :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot oftime for execution
On Wed, 21 Oct 2009, Danny Nicholas wrote: Not my cup of tea, but I think I'd be trying an ODBC connection to reduce some overhead here. [snip] Does that reduce overhead or add it? Seems that direct mysql-client code should be more efficient than adding ODBC in the middle... j ___ -- 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] Incorrect voice mail format on transfer
Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] file.c: Failed to write frame Then the recipient trying to open the transferred voicemail: [Oct 21 13:27:25] WARNING[13565] file.c: File /var/spool/asterisk/voicemail/a10/612/INBOX/msg does not exist in any format [Oct 21 13:27:25] WARNING[13565] file.c: Unable to open /var/spool/asterisk/voicemail/a10/612/INBOX/msg (format 0x4 (ulaw)): No such file or directory [Oct 21 13:27:25] WARNING[13565] app_voicemail.c: Playback of message /var/spool/asterisk/voicemail/a10/612/INBOX/msg failed [Oct 21 13:27:37] WARNING[1678] app_voicemail.c: IMAP Warning: Unknown message data: 63 FETCH [Oct 21 13:27:40] WARNING[13565] file.c: File /var/spool/asterisk/voicemail/a10/612/INBOX/msg does not exist in any format [Oct 21 13:27:40] WARNING[13565] file.c: Unable to open /var/spool/asterisk/voicemail/a10/612/INBOX/msg (format 0x4 (ulaw)): No such file or directory [Oct 21 13:27:40] WARNING[13565] app_voicemail.c: Playback of message /var/spool/asterisk/voicemail/a10/612/INBOX/msg failed [Oct 21 13:28:20] WARNING[13565] channel.c: Unexpected control subclass '17' [Oct 21 13:28:50] WARNING[13572] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 13:29:03] WARNING[13572] file.c: File /var/spool/asterisk/voicemail/a10/612/INBOX/msg does not exist in any format [Oct 21
Re: [asterisk-users] Concurrent calls including mysql taking lot oftime for execution
I don't use ODBC or MYSQL, but the problem the OP mentions is that MYSQL takes .X seconds longer each time he calls it until it takes 5-10 seconds to connect on the 100th call. I know some guru out there is probably handling 1000 calls using a MYSQL database, so maybe yall can tell OP what is hosed. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, October 21, 2009 3:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Concurrent calls including mysql taking lot oftime for execution On Wed, 21 Oct 2009, Danny Nicholas wrote: Not my cup of tea, but I think I'd be trying an ODBC connection to reduce some overhead here. [snip] Does that reduce overhead or add it? Seems that direct mysql-client code should be more efficient than adding ODBC in the middle... j ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On 22/10/09 7:30 AM, das sandesh wrote: Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: Without knowing what you're optimising you're unlikely to have much luck just setting values. We have had quite good success with the tunish-primer.sh script: http://www.day32.com/MySQL/ http://www.day32.com/MySQL/tuning-primer.sh We run with MySQL at about 500 queries per second with no problems - we don't however use Asterisk's MySQL libraries. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Concurrent calls including mysql taking lot oftime for execution
On 22/10/09 9:16 AM, Jeff LaCoursiere wrote: On Wed, 21 Oct 2009, Danny Nicholas wrote: Not my cup of tea, but I think I'd be trying an ODBC connection to reduce some overhead here. [snip] Does that reduce overhead or add it? Seems that direct mysql-client code should be more efficient than adding ODBC in the middle... Yep, ODBC would add overhead - you may want to look at using FastAGI and keeping a MySQL connection open inside your script (i.e. connection pooling). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On 22/10/09 8:56 AM, David Backeberg wrote: On Wed, Oct 21, 2009 at 2:30 PM, das sandeshsandesh...@gmail.com wrote: I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: My guess is DNS taking a long time to timeout? Trying changing the connection string to use straight ip address rather than hostname. Alternatively install a caching name server. In debian just do apt-get install bind9 then change your nameserver to 127.0.0.1 -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Incorrect voice mail format on transfer
It should be reproducible in some way, how was asterisk installed on the server its having a problem? If its from source compare the apps/app_voicemail.c from whats in production with whats getting compiled in the lab. when imap is used only one format is stored you could specify just one format: format=wav49 On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] file.c: Failed to write frame Then the recipient trying to open the transferred voicemail: [Oct 21 13:27:25] WARNING[13565] file.c: File /var/spool/asterisk/voicemail/a10/612/INBOX/msg does not exist in any format [Oct 21 13:27:25] WARNING[13565] file.c: Unable to open /var/spool/asterisk/voicemail/a10/612/INBOX/msg (format 0x4 (ulaw)): No such file or directory [Oct 21 13:27:25] WARNING[13565] app_voicemail.c: Playback of message /var/spool/asterisk/voicemail/a10/612/INBOX/msg failed [Oct 21 13:27:37] WARNING[1678] app_voicemail.c: IMAP Warning: Unknown message data: 63 FETCH [Oct 21 13:27:40] WARNING[13565] file.c: File /var/spool/asterisk/voicemail/a10/612/INBOX/msg does not exist in any format [Oct 21 13:27:40] WARNING[13565] file.c: Unable to open /var/spool/asterisk/voicemail/a10/612/INBOX/msg (format 0x4 (ulaw)): No such file or directory [Oct 21 13:27:40]
Re: [asterisk-users] Incorrect voice mail format on transfer
I'm sorry - by the lab I meant the end points - it is the same server. I was not aware that IMAP only stored one format. If I change the setting in voicemail.conf, do I still have to worry about the grievous warning message about being sure to delete all messages not using that format? I would think not but it's a dire enough message that I thought I had better ask - John On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: It should be reproducible in some way, how was asterisk installed on the server its having a problem? If its from source compare the apps/app_voicemail.c from whats in production with whats getting compiled in the lab. when imap is used only one format is stored you could specify just one format: format=wav49 On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303]
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
On Wed, 21 Oct 2009, das sandesh wrote: I am using only asterisk code (dial plan) in extensions.conf which also includes connection to the database: like exten =n,1, MYSQL(connect connid ipaddr uname pwd database) and then the required select queries and the clear and Disconnect the connection. I'm not a big fan of doing anything performance sensitive in scripting languages. I'm also not a big fan of doing MySQL in dialplan. I think it makes for a very ugly and difficult to maintain dialplan. Since there is no substantial syntax checking, every time you edit your dialplan you risk fat-fingering something that you (or somebody less skilled than you) may not notice and may take considerable effort to debug. When the live calls are made to test and at 200th or at around 250th call there is a point where it took like 5-10 sec just to connect to the database and in the mean time we get dead air for that period of time..how can we change the type of connection that you mentioned? Since I don't do MySQL in dialplan, I may be wrong here, but in C all you have to do is change the host (or IP address) to localhost. I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. Here's how I broke down part of a recent project: ) block-ani -- lookup the caller's ANI in the database and set STATUS (a channel variable) to BLOCK, PASS, FAILURE. ) lookup-dnis -- lookup the dialed number in the database and set a bunch of channel variables from the database. My current project sets around 350 variables in the blink of an eye -- at least less than a second. Sets STATUS to SUCCESS or FAILURE. ) auth-card -- creates a thread to play Please hold while your card is being verified... while the mainline code checks to see if the credit card is in a known bad database and issues an authorization request via TCP to the card processor. Usually we get the response before the prompt completes playing so it appears instantaneous to the caller. Sets STATUS to SUCCESS or FAILURE. ) messages -- kind of like a voicemail system where callers can record messages for other callers and listen to messages left for them. Lots of database activity. ) most-idle-agent -- find the online agent who has been idle the longest and has the skills (from the database) needed for the caller. Sets AGENT (a channel variable) to the agent's ID or GROUP. ) settle-card -- called when the caller hangs up, rates the call based on how much time they spent in each product and issues the card sale request. Most of these could have been done in the dialplan, but it would have been completely un-maintainable and prone to failure. Or might be is it good to go with dual quad core processor instead of just one inorder to handle the call capacity as well as connections? I'm not a big fan of throwing hardware at something that may be easy to fix. What will you do if your business doubles? You mentioned 400-500 simultaneous calls. You may want to re-think your architecture to split that across several hosts. I'd rather tell my client a host smoked and only took 100 calls with it -- each call in the above project is worth about US$30. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? Thanks Sandesh On Wed, Oct 21, 2009 at 3:57 PM, Matt Riddell li...@venturevoip.com wrote: On 22/10/09 7:30 AM, das sandesh wrote: Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: Without knowing what you're optimising you're unlikely to have much luck just setting values. We have had quite good success with the tunish-primer.sh script: http://www.day32.com/MySQL/ http://www.day32.com/MySQL/tuning-primer.sh We run with MySQL at about 500 queries per second with no problems - we don't however use Asterisk's MySQL libraries. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On Wed, 21 Oct 2009, Steve Edwards wrote: I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. And you can debug them (AGIs in any language) from the command line completely outside of Asterisk. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] polarity on some channels
It's not caller ID issue, I can make asterisk answer the line by omitting the line answeronpolarityswitch=no , but this will take effect on all 24 TDM channels, I want some to have answer on polarity, and some without polarity. Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, October 21, 2009 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity on some channels B.Masoud @ SH wrote: Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks _ ___ -- 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 Try turning off callerid. The 'standard' for POTS lines in the US is to put the caller id in between ring1 ring2. Asterisk waits for callerid before answering the line by default. usecallerid=off ___ -- 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] Incorrect voice mail format on transfer
If you're using file storage and specify three formats, app_voicemail will save to those formats. The dire warning is because when renaming (for example listening to new/msg and it gets moved to old messages) and deleting files, app_voicemail only touches the formats in the configuration file. set format=wav49|gsm, reload config Record a message set format=wav49, reload config delete the message, doesn't delete msg.gsm Record a message set format=wav49|gsm, reload config Connect to app_voicemail with gsm codec and hear that old message again just like its the first time. On Wed, Oct 21, 2009 at 2:37 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: I'm sorry - by the lab I meant the end points - it is the same server. I was not aware that IMAP only stored one format. If I change the setting in voicemail.conf, do I still have to worry about the grievous warning message about being sure to delete all messages not using that format? I would think not but it's a dire enough message that I thought I had better ask - John On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: It should be reproducible in some way, how was asterisk installed on the server its having a problem? If its from source compare the apps/app_voicemail.c from whats in production with whats getting compiled in the lab. when imap is used only one format is stored you could specify just one format: format=wav49 On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44]
Re: [asterisk-users] Concurrent calls including mysql taking lot of time for execution
On 22/10/09 10:57 AM, das sandesh wrote: Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? The thing is, concurrent calls won't make any difference, it's the calls per second. And really you're unlikely to use too many queries per sec. Seriously, use at least AGI (fastAGI would be better but AGI will at least give you a start). So: 1. Do you get the same delay if you use MySQL command line at the same time? 2. Do you have a programming language you know well enough to connect to MySQL in? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Concurrent calls including mysql taking lot of time for execution
The thing is, concurrent calls won't make any difference, it's the calls per second. And really you're unlikely to use too many queries per sec. Exactly and you can see the slow-log-queries if mysql is taking time. -Jai On Wed, Oct 21, 2009 at 3:51 PM, Matt Riddell li...@venturevoip.com wrote: On 22/10/09 10:57 AM, das sandesh wrote: Hi Matt, I already used the tuning-primer.sh script to enhance the values for the parameters, but still it was being slow to connect when there are lot of calls (calls around 150-200 calls). Also I reduced mysql queries in the code as well as many other steps, but only problem coming is with repect to the connection from asterisk to mysql (also I am using direct ip address and not the dns name).is it better to use any additional mysql server apart from this application server? or adding additional hardware would help (like dual quad core)? The thing is, concurrent calls won't make any difference, it's the calls per second. And really you're unlikely to use too many queries per sec. Seriously, use at least AGI (fastAGI would be better but AGI will at least give you a start). So: 1. Do you get the same delay if you use MySQL command line at the same time? 2. Do you have a programming language you know well enough to connect to MySQL in? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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 ___ -- 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] IMAP voicemail using subfolders fails.
Barry L. Kline wrote: Kevin P. Fleming wrote: It's not present in the current 1.4 doc/imapstorage.txt file, or any later version. I don't even know why the storage format would matter, since that would be very specific to the IMAP server that is managing that folder. Hmmm http://markmail.org/message/up3rfmdk2kjf6r7y is a link that contains the contents of a README file that looks like it came from Digium. About half-way down is: -- Mailbox Format -- Mailboxes should use the mbx mailbox format. The mbox format does not support concurrent access to mailboxes, which can cause deadlock or strange behaviors. You can convert mailboxes from mbox to mbx using mailutil: Perhaps that came from a different product? I think that I'm going to just go ahead and implement IMAP VM and see what happens. Barry, I don't think that Maildir or a database backend solution (such as Exchange) suffers from this same limitation. I would be more interested in knowing how sensitive this would be to latency if using an IMAP server that isn't on the same device as the Asterisk server (or perhaps even a remote IMAP server)? Darrick ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Steve Edwards asterisk@sedwards.com wrote: On Wed, 21 Oct 2009, Steve Edwards wrote: I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. And you can debug them (AGIs in any language) from the command line completely outside of Asterisk. OK, are there include files available for the appropriate functionality? Sounds like it might be very nice. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On 22/10/09 1:41 PM, cov...@ccs.covici.com wrote: Steve Edwardsasterisk@sedwards.com wrote: On Wed, 21 Oct 2009, Steve Edwards wrote: I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. And you can debug them (AGIs in any language) from the command line completely outside of Asterisk. OK, are there include files available for the appropriate functionality? Sounds like it might be very nice. It's really simple you just read from standard input and write to standard output. If you tell us a programming language you'd like to use (i.e. php/c/perl/bash etc) we can give you a link to some docs and examples. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On Wed, 21 Oct 2009, Steve Edwards wrote: I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. And you can debug them (AGIs in any language) from the command line completely outside of Asterisk. On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote: OK, are there include files available for the appropriate functionality? Sounds like it might be very nice. If you're referring to debugging outside of Asterisk, it's dead obvious -- once you know the secret. The AGI protocol is just communications over STDIN and STDOUT in a specific format. Thus, running outside of Asterisk just means feeding the right stuff going in and observing the right stuff coming out. For example: ./block-ani dummy-input-for-block-ani where dummy-input-for-block-ani contains: agi_accountcode: agi_callerid: 1234567890 agi_calleridname: sedwards agi_callingani2: 0 agi_callingpres: 0 agi_callingtns: 0 agi_callington: 0 agi_channel: SIP/201-09456478 agi_context: newline agi_dnid: * agi_enhanced: 0.0 agi_extension: * agi_language: en agi_priority: 1 agi_rdnis: unknown agi_request: block-ani agi_type: SIP agi_uniqueid: 1195070681.28 200 result=1 (551212) 200 result=1 (localhost) 200 result=1 (example) 200 result=1 (example) 200 result=1 (example) The first block is the standard AGI environment. The second block is specific to this AGI and supplies the answers to the AGI requests GET VARIABLE ANI, GET VARIABLE DATABASE-SERVER, GET DATABASE-DATABASE, GET DATABASE-USERNAME, and GET DATABASE PASSWORD. I prefer to use an executable script so I can include comments. The script looks like: # agi-environment.sh # the standard AGI environment variables echo agi_accountcode: echo agi_callerid: 1234567890 echo agi_calleridname: sedwards echo agi_callingani2: 0 echo agi_callingpres: 0 echo agi_callingtns: 0 echo agi_callington: 0 echo agi_channel: SIP/201-09456478 echo agi_context: newline echo agi_dnid: * echo agi_enhanced: 0.0 echo agi_extension: * echo agi_language: en echo agi_priority: 1 echo agi_rdnis: unknown echo agi_request: block-ani echo agi_type: SIP echo agi_uniqueid: 1195070681.28 echo # cruft specific to my AGI # AGI Rx GET VARIABLE ANI echo 200 result=1 (551212) # AGI Rx GET VARIABLE DATABASE-SERVER echo 200 result=1 (localhost) # AGI Rx GET VARIABLE DATABASE-DATABASE echo 200 result=1 (example) # AGI Rx GET VARIABLE DATABASE-USERNAME echo 200 result=1 (example) # AGI Rx GET VARIABLE DATABASE-PASSWORD echo 200 result=1 (example) # (end of agi-environment.sh) And you use it like: ./agi-environment.sh | ./block-ani or ./agi-environment.sh dummy-input-for-block-ani ./block-ani dummy-input-for-block-ani Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Beats the hell out of peppering your code with prints/puts/echos and crossing your fingers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Steve Edwards asterisk@sedwards.com wrote: On Wed, 21 Oct 2009, Steve Edwards wrote: I'd take a look at using AGIs written in C. They make nice little building blocks. They execute very quickly and can cleanup your dialplan. And you can debug them (AGIs in any language) from the command line completely outside of Asterisk. On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote: OK, are there include files available for the appropriate functionality? Sounds like it might be very nice. If you're referring to debugging outside of Asterisk, it's dead obvious -- once you know the secret. The AGI protocol is just communications over STDIN and STDOUT in a specific format. Thus, running outside of Asterisk just means feeding the right stuff going in and observing the right stuff coming out. For example: ./block-ani dummy-input-for-block-ani where dummy-input-for-block-ani contains: agi_accountcode: agi_callerid: 1234567890 agi_calleridname: sedwards agi_callingani2: 0 agi_callingpres: 0 agi_callingtns: 0 agi_callington: 0 agi_channel: SIP/201-09456478 agi_context: newline agi_dnid: * agi_enhanced: 0.0 agi_extension: * agi_language: en agi_priority: 1 agi_rdnis: unknown agi_request: block-ani agi_type: SIP agi_uniqueid: 1195070681.28 200 result=1 (551212) 200 result=1 (localhost) 200 result=1 (example) 200 result=1 (example) 200 result=1 (example) The first block is the standard AGI environment. The second block is specific to this AGI and supplies the answers to the AGI requests GET VARIABLE ANI, GET VARIABLE DATABASE-SERVER, GET DATABASE-DATABASE, GET DATABASE-USERNAME, and GET DATABASE PASSWORD. I prefer to use an executable script so I can include comments. The script looks like: # agi-environment.sh # the standard AGI environment variables echo agi_accountcode: echo agi_callerid: 1234567890 echo agi_calleridname: sedwards echo agi_callingani2: 0 echo agi_callingpres: 0 echo agi_callingtns: 0 echo agi_callington: 0 echo agi_channel: SIP/201-09456478 echo agi_context: newline echo agi_dnid: * echo agi_enhanced: 0.0 echo agi_extension: * echo agi_language: en echo agi_priority: 1 echo agi_rdnis: unknown echo agi_request: block-ani echo agi_type: SIP echo agi_uniqueid: 1195070681.28 echo # cruft specific to my AGI # AGI Rx GET VARIABLE ANI echo 200 result=1 (551212) # AGI Rx GET VARIABLE DATABASE-SERVER echo 200 result=1 (localhost) # AGI Rx GET VARIABLE DATABASE-DATABASE echo 200 result=1 (example) # AGI Rx GET VARIABLE DATABASE-USERNAME echo 200 result=1 (example) # AGI Rx GET VARIABLE DATABASE-PASSWORD echo 200 result=1 (example) # (end of agi-environment.sh) And you use it like: ./agi-environment.sh | ./block-ani or ./agi-environment.sh dummy-input-for-block-ani ./block-ani dummy-input-for-block-ani Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Beats the hell out of peppering your code with prints/puts/echos and crossing your fingers. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 OK, but how do write the C program -- the Perl and php agis have defined functions for the agi commands, how do you do this in c? -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Steve Edwards asterisk@sedwards.com wrote: Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Bah. If you were really old school you would use vi. [ducking!] :) j ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Hey now, I'm a newschool programmer and I use vim (and vi, when necessary). Andrew On Wed, Oct 21, 2009 at 8:02 PM, Jeff LaCoursiere j...@jeff.net wrote: Steve Edwards asterisk@sedwards.com wrote: Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Bah. If you were really old school you would use vi. [ducking!] :) j ___ -- 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 ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On 22/10/09 2:54 PM, cov...@ccs.covici.com wrote: OK, but how do write the C program -- the Perl and php agis have defined functions for the agi commands, how do you do this in c? There is a library (haven't used it myself) http://sourceforge.net/projects/cagi/ Basically you read from the standard input (i.e. fgets or similar) and write to the standard output (printf or similar). -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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] Concurrent calls including mysql taking lot of time for execution
Jeff LaCoursiere wrote: Steve Edwards asterisk@sedwards.com wrote: Since I'm an old-school C programmer, I use emacs as my editor. I fire up gdb (the GNU C (amongst other languages) debugger) in a window, give it a command like b main; r dummy-input-for-block-ani and I can step through my program line by line, examining and changing variables at will. Bah. If you were really old school you would use vi. [ducking!] :) j Old school? I tried to use 'ed' the other day, and failed. PaulH ___ -- 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] Asterisk 1.6.1.6 crashing -- multiple phone entries
Folks, Not sure what's going on, but suddenly Asterisk 1.6.1.6 is crashing, usually when I exit the console or use asterisk -rx. The sip peers entry always shows duplicate entries (once I had an extension over half a dozen times) just before it crashes. 3182/3182 172.17.0.126 D N 5060 OK (14 ms) 3183/3183 172.17.0.128 D N 5060 OK (14 ms) 3183/3183 172.17.0.128 D N 5060 OK (14 ms) 3184/3184 (Unspecified)D N 5060 UNKNOWN Is this a known issue? It did just start crashing (and again, only after exhibiting this bizarre behavior of multiple duplicate entries). My other system running 1.6.1.6 does not appear to have this problem. Any thoughts? Thanx, David A. Bandel -- Focus on the dream, not the competition. - Nemesis Air Racing Team motto Visit my blog at: http://www.pananix.com/cgi-bin/blosxom ___ -- 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] Concurrent calls including mysql taking lot oftime for execution
On Wednesday 21 October 2009 15:16:31 Jeff LaCoursiere wrote: On Wed, 21 Oct 2009, Danny Nicholas wrote: Not my cup of tea, but I think I'd be trying an ODBC connection to reduce some overhead here. [snip] Does that reduce overhead or add it? Seems that direct mysql-client code should be more efficient than adding ODBC in the middle... In this case, it would reduce overhead. The example he provides creates a unique connection for every channel, which is massive overkill for MySQL. Only Sybase and MS SQL Server require a distinct connection for each live query. MySQL can very effectively run multiple queries on a single connection. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- 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] Concurrent calls including mysql taking lot of time for execution
On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote: OK, but how do write the C program -- the Perl and php agis have defined functions for the agi commands, how do you do this in c? The same way. All languages need a library. Either you find a library that talks AGI or you write one. I wrote mine because when I started writing AGIs about 5 years ago, I didn't have much luck finding one. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] ivr menu not hanging up call
I am testing an ivr but I'm having problems. The call keeps looping and it
doesn't hangup the call after passing three times through the menu. Here's my
conf:
exten = s,n,NoOp(Here's Count)
exten = s,n,NoOp(${COUNT})
;123,n,Set(COUNT=$[${COUNT} - 1])
exten = s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 )
exten = 1,1,goto(tech-support,s,1)
exten = 2,1,goto(sales,s,1)
exten = 3,1,goto(cust-service,s,1)
exten = 100,1,goto(wilson,s,1)
exten = 102,1,goto(sales,s,1)
exten = i,1,Playback(invalid)
exten = i,n,Playback(please-try-again)
exten = i,n,goto(ivr,s,5)
exten = i,n,Playback(goodbye)
exten = i,n,Hangup
exten = 33,1,PlayBack(please-try-again-later)
exten = 33,n,PlayBack(call-terminated)
exten = 33,n,PlayBack(goodbye)
exted = 33,n,HangUp()
exten = 44,1,goto(ivr,s,5)
exten = t,1,goto(ivr,s,2)
exten = h,1,Hangup
When it enters extension 33 it should hangup the call but, if the caller stays
on the line the exten = t,1,goto(ivr,s,2) takes over and the menu keeps
repeating. Should I just remove that t extension?
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Re: [asterisk-users] ivr menu not hanging up call
On Wed, 21 Oct 2009, Landy Landy wrote:
I am testing an ivr but I'm having problems. The call keeps looping and
it doesn't hangup the call after passing three times through the menu.
When it enters extension 33 it should hangup the call but, if the caller
stays on the line the exten = t,1,goto(ivr,s,2) takes over and the
menu keeps repeating. Should I just remove that t extension?
If this is the actual dialplan...
[snip]
exten = 33,1,PlayBack(please-try-again-later)
exten = 33,n,PlayBack(call-terminated)
exten = 33,n,PlayBack(goodbye)
exted = 33,n,HangUp()
exted != exten
If this isn't a cut paste, a cut paste from show dialplan may shed
some light.
exten = s,n,NoOp(Here's Count)
exten = s,n,NoOp(${COUNT})
Just a suggestion...
There is an application specifically designed to output to the console
named verbose(). It's more flexible and obvious, rather than relying on a
side effect of noop().
I know everybody does it, but it's kind of like using a screwdriver to
open a can of paint. You can, but there is a tool made just for that
purpose. (And it has a beer bottle opener on the other end!)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] OT - Gigaset Chagall - How to download firmware without Internet access ?
2009/10/21 Leif Madsen leif.mad...@asteriskdocs.org Olivier wrote: Hi, Siemens Gigaset line of products include an integrated web browser with which firmware download is possible. The trouble is you need to provide Internet access. We use a couple of these boxes in LANs not connected to Internet for security reasons. So I would prefer to download firmware upgrades from my own TFTP or HTTP server. Thanks to Wireshark, I could list downloaded files list. For instance, latest C450IP files include : http://gigaset.siemens.com/chagall/1/0/master.bin http://gigaset.siemens.com/chagall/1/0/../baselines.bin http://gigaset.siemens.com/chagall/1/0/../chagall072_01.bin All these files can be copied (using a wget command) and copied to a personal web server but information is missing to extend this process to each model. Change your local nameserver to resolve the address to a private IP instead of to the public IP? Yes but the hard part is to properly identify and copy the files to download. With C450IP, you can edit a text field from which the base station will download its firmware (using HTTP or TFTP). The trouble is I don't know which files exactly to put in the HTTP or TFTP server. Reading at the example above, I would say 3 files are needed for this 072_01 firmware. When the next XXX_YY firmware will be published, should I just add the chagallXXX_YY.bin file in the appropriate directory ? Leif! ___ -- 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 ___ -- 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 and Nuance Vocalizer TTS Engine
2009/10/21 Christophorus Laube christophorus.la...@semanticedge.de I think you should use the nvcmdline utility Is this nvcmdline bundled with every Nuance TTS ? ___ -- 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] Concurrent calls including mysql taking lot of time for execution
There were 2 problems that we faced, one was at around 50 calls, few calls were just dead air, and when I saw the logs I could see that it was sent to the sip provider and after that there was no log for that particular call that was having dead air, but at around 200 to 250, we could see that MySQL(Connect connid ipaddr uname pwd db) statement took around 5-10 sec to connect to the database and then the 2 queries in that code got executed pretty fast (1-2sec), and so here we had the dead air untill the call got connected (after 5-10sec). We also monitored the processor usage and it was around 15-20% CPU and memory was around 300M to 400M, so we concluded that it was not the hardware issue.based on all of your opinions i will try to see whether I can use any other language and try to do those operations.Thanks for all of your information! On Wed, Oct 21, 2009 at 10:51 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 21 Oct 2009, cov...@ccs.covici.com wrote: OK, but how do write the C program -- the Perl and php agis have defined functions for the agi commands, how do you do this in c? The same way. All languages need a library. Either you find a library that talks AGI or you write one. I wrote mine because when I started writing AGIs about 5 years ago, I didn't have much luck finding one. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 ___ -- 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
