[asterisk-users] Polycom IP321?
A client of mine asked about a Polycom IP321..anyone else heard about it? -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] First ever Open Source Asterisk / Wave bounty
Hi, On Mon, Jun 01, 2009 at 08:08:41PM -0400, Dean Collins wrote: I've just received an email from a colleague who told me to put my money where my mouth is So here it is - I'm offering $500 and looking for other people to add to this bounty. While not my money, and not osmething I consider important enough to put my money or time on, We can get a group of people putting matching funds up to finalize the scope of the first Open Source Asterisk / Wave conference call integration robot bounty but if you have any other suggestions feel free to add to the list below or to pass around / retweet this link http://bit.ly/t9c5C Functionality of the Open Source Asterisk / Wave conference call robot bounty Asterisk Conference server spawn waves event to all participants with the details of the call length, With details of who was on the call, What time they dialed in/out, their numbers, any notes that were taken by all parties during the call urls for the call voice recording access at a later date This requires quite a few things. For starters, it requires that you know the address of a participant in a conference. There are many useful applications of such a protocol even before that. Asterisk has currently very poor support of text messages. Asterisk cannot route text messages. Asterisk can send Jabber messages, SIP/SIMPLE messages, chan_mobile SMS messages, PSTN SMS messages and probably some other channel-specific SMS messages, but all with different syntaxes. So two interesting subgoals (with no specific order) would be: 1. A similar integration to that of res_jabber of today - the ability to send messages and handle them. I have no idea how that Wave of the Future handles authentication and authorization (authorizaiton: think spam). 2. A more common way to handle text messages. Would it be nice to be able to route text messages in the dialplan or is it outside the scope of Asterisk? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 with dial timeout
Hello, I am trying to do : Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Why? I forgot something ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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 with dial timeout
BERGANZ François schrieb: Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Syntax: Dial(Technology/resource[Tech2/resource2...][,timeout][,options][,URL]) You have to pass L() as the options argument. Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) ^ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 with dial timeout
Try this: Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of BERGANZ François Sent: 2009 m. birželio 2 d. 11:07 To: asterisk-users@lists.digium.com Subject: [asterisk-users] error with dial timeout Hello, I am trying to do : Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Why? I forgot something ? Thank you Cordialement, BERGANZ François P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. ___ -- 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 with dial timeout
Thank you! I did understood that i twas THAT timeout :-) I thought that it speak about my 'limit call' Thank you Cordialement, BERGANZ François Pensez à l'Environnement, n'imprimez ce mail que si nécessaire. -Message d'origine- De : asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] De la part de Philipp Kempgen Envoyé : mardi 2 juin 2009 10:37 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] error with dial timeout BERGANZ François schrieb: Exten =_X.,n,Dial(SIP/ser_sei0/1130,L(10208400:61000:1)) But it return that error: [Jun 2 10:04:44] WARNING[18920]: app_dial.c:1623 dial_exec_full: Invalid timeout specified: 'L(10208400:61000:1)' Syntax: Dial(Technology/resource[Tech2/resource2...][,timeout][,options][,URL]) You have to pass L() as the options argument. Exten =_X.,n,Dial(SIP/ser_sei0/1130,,L(10208400:61000:1)) ^ Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] zaptel to dahdi
Hi i have just recently installed asterisk 1.4 server with a digium card 410, i used the zaptel packages in debian. now I have notice the move to dahdi which seems to be a rename and some changes as well. is it a easy change from zaptel to dahdi ? any sort of gotchas to watch out for ? Alex signature.asc Description: Digital signature ___ -- 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] Transfer call from analog telephone
Remember that the time between the two digits is VERY short. You must press those two digits in quick succession or else the requested feature code will not activate. - Or set featuredigittimeout longer. ___ -- 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] problem with outgoing calls
hi, firstly excuse me for my bad English I configured my astrerisk, and it goes for internal call but when I want to make outgiong call I arriven't and the asterisk indicates the following error == Using SIP RTP CoS mark 5 -- Executing [0671735...@default:1] Dial(SIP/100-0826a070, SIP/ 0671735...@10.76.252.3) in new stack == Using SIP RTP CoS mark 5 -- Called 0671735...@10.76.252.3 -- Got SIP response 482 Loop Detected back from 0.0.0.0 -- Now forwarding SIP/100-0826a070 to 'Local/0671735...@default' (thanks to SIP/10.76.252.3-08267f08) -- Executing [0671735...@default:1] Dial(Local/0671735...@default-6b02;2, SIP/0671735...@10.76.252.3) in new stack [Jun 2 10:10:25] WARNING[6474]: app_dial.c:1437 dial_exec_full: Skipping dialing interface 'SIP/0671735...@10.76.252.3' again since it has already been dialed == Spawn extension (default, 0671735116, 1) exited non-zero on 'Local/0671735...@default-6b02;2' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/100-0826a070' status is 'CHANUNAVAIL' thanks for your help ___ -- 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] h323 guide for asterisk
Maybe this can help you? http://astrecipes.net/index.php?n=286 Thanks l. 2009/5/31 Tamer Higazi th9...@googlemail.com Hi people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.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] [Atcom] Asterisk + LAMP on 128MB RAM?
On Tue, 02 Jun 2009 02:17:02 +0800, Steve Underwood ste...@coppice.org wrote: Linux is not a given here. The Blackfin runs uCLinux, as it has non MMU. Don't get too enthusiastic about putting complex applications like Apache, MySQL or PHP on one of those boxes. The memory management limitations of uCLinux can be quite restricting. I'll keep that in mind, and see if works OK on that hardware. Thank you. ___ -- 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] [Atcom] Asterisk + LAMP on 128MB RAM?
On Tue, Jun 02, 2009 at 01:45:17PM +0200, Vincent wrote: On Tue, 02 Jun 2009 02:17:02 +0800, Steve Underwood ste...@coppice.org wrote: Linux is not a given here. The Blackfin runs uCLinux, as it has non MMU. Don't get too enthusiastic about putting complex applications like Apache, MySQL or PHP on one of those boxes. The memory management limitations of uCLinux can be quite restricting. I'll keep that in mind, and see if works OK on that hardware. For the record, the part was not too serious. Specifically I'm not really sure how many people actually use Lua. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Call quality - how to debug
Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, Thanks Adrian ___ -- 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] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I’ve a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I’ve one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I’ve turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. I’m going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 maximum user
How many asterisk voice user concurrently continue their voice only one Asterisk sever? Could it possible implementation in big environment (more than thousand users)? Hardware statistics: vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 9 cpu MHz : 2799.622 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe up pebs bts cid xtpr bogomips: 5604.90 clflush size: 64 Thanks Monzur ___ -- 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] Call quality - how to debug
- Steve Howes st...@geekinter.net wrote: On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I’ve a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I’ve one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I’ve turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. I’m going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ Physical or virtualised server ? Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ___ -- 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] SIP Response 181 - Is it possible in Asterisk?
Hello all, I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html http://www.tech-invite.com/Ti-sip-service-8.html I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. Thanks, Marco Cordeiro mhcorde...@gmail.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] Call quality - how to debug
Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] SIP Response 181 - Is it possible in Asterisk?
Marco Cordeiro schrieb: I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html http://www.tech-invite.com/Ti-sip-service-8.html I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Call quality - how to debug
I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] Call quality - how to debug
Scratch that, my inventory tool says the system has 256Mb not 1Gb. I wonder if a memory upgrade would help it out... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 02 June 2009 14:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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] Call quality - how to debug
Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- 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] RES: SIP Response 181 - Is it possible in Asteri sk?
Thanks Philipp, Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could I find info about it? Thanks again, Marco -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Enviada em: terça-feira, 2 de junho de 2009 11:02 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk? Marco Cordeiro schrieb: I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html http://www.tech-invite.com/Ti-sip-service-8.html I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Call quality - how to debug
Yeah, I know, but when I last tried an upgrade to 1.4.18 it broke the whole IAX connectivity and I was forced to drop back. I'll go: 1) Memory upgrade first 2) Clone the machine, and upgrade to latest 1.4.x However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: 02 June 2009 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- 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] DAHDI Linux 2.2.0-rc5 and Tool 2.2.0-rc3 Release Announcement
The Asterisk Development Team is pleased to announce the fifth release candidate of DAHDI Linux 2.2.0 and the third release candidate of DAHDI tools. Both release candidates are available for immediate download at http://downloads.asterisk.org/pub/telephony In addition to various bug fixes, these release candidates include: * Support for new Xorcom Astribanks with the TwinStar[tm] option. * Improved hardware echo canceler performance for Digium VPMADT032. * Improved fax tone detection and echo canceler / fax handling. * Improved timing accuracy of dahdi_dummy, including when running in virtual environments. * Fixes for Dahdi-perl for non-Xorcom hardware. * BRI Astirbank modules no longer need the bri_dchan patch. * Explicit ordering of Astribanks for multi-Astribanks setups. Please report issues found in this release candidate on http://issues.asterisk.org/. For a full list of the changes in these release candidates, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/dahdi-linux/ChangeLog-2.2.0-rc5 http://downloads.asterisk.org/pub/telephony/dahdi-tools/ChangeLog-2.2.0-rc3 Thank you for your continued support of Asterisk! ___ -- 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] Realtime LDAP passwords
Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Call quality - how to debug
On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote: However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Wireshark has a lot of RTP tools for looking at the latency and jitter and dropped packets on the line, which are the most common problems I find when helping people diagnose poor audio connections. It won't tell you what is *causing* the problem, but it will help you know what the problem actually is. From there, you can start to track down the source of the problem one network segment at a time. For example... is the poor audio being caused by network problems between the phone and Asterisk, or between Asterisk and your upstream provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: I’m at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I’m not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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] extensions not being detected consistently
Extensions that are dialed within macros like the following lines could cause the type of problems as you mentioned: exten = s,n,Macro(dial-us) exten = s,n,Macro(hangupcall) This line: exten = s,n,Wait(0.5); should be changed to exten = s,n,WaitExten(0.5); and these lines: exten = Wait(10) exten = s,n(open),NoOp(open) are not valid. Try this: exten = s,1,Set(TIMEOUT(digit)=10) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(cassandra/CustomWelcomeMessage) exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open) exten = s,n,Background(cassandra/OfficeHours) exten = s,n,Background(cassandra/NextRep) exten = t,n,Macro(dial-us) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,open) exten = #,1,Macro(hangupcall) On Mon, Jun 1, 2009 at 1:47 PM, Terry Nathan tnat...@aiinc.ca wrote: G'afternoon everybody, I'm having a problem with consistently being able to ring our extensions from an outside line. I don't have a problem reaching the number, but during our calls to Background(msg) that I am having a problem. It seems to be an issue with timing. If I press the extension towards the end of the Background(msg) the it often works. However, in the middle of the message it will not work at all. What is also strange is that I can dial an extension any time if I call from one of our ip phones. This seems to be strictly a problem with regular phones, then the timing of dialing the extension becomes important. The fact that the ip phones always work seems to suggest that I need to look at tone detection, but after googling and searching the bowels of every conf file I could find, I haven't found any magic bullet. I should mention that the first call to Background() usually works, even for the regular phones, I think this is because it is short enough that the timing of dialing the extension is relatively easy. I don't know if it is significant or not but it seems that once a callee tries to dial an extension and it doesn't work, even the next few calls will also not work. And similarly, sometimes it works and then a few calls will go through, but then it will go back to not detecting properly again. Asterisk is running on its own box and there is nothing unusual happening with the system, or even people on other lines, that is happening. Checking the log files when I call in Asterisk tells me that either it only detects 1 of the 3 digits (usually the second or third one) or, if I dial the extension at a different point in the message, that the first digit was pressed twice e.g. '22' instead of just '2'. The inconsistency of the problem is starting to drive me bonkers as I can't accurately nail down the problem. Ideally I'd like our callees to be able to dial an extension as soon as the call to Background() is hit in the context, from any phone that calls in, not just ip phones. My setup is an installation of asterisk-now (Centos 5 with Asterisk 1.4.24) If anyone has seen a problem like this before or has even an inkling of what it might be, that would be awesome :D Thanks in advance. My dial plan: [incoming-our-number] exten = s,1,Answer exten = s,n,NoOp(incoming-our-number) exten = s,n,Background(cassandra/CustomWelcomeMessage) This line is usual fine, I think because the message is short enough that timing the dialing of the extension is less of an issue. exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open) exten = s,n,Wait(0.5); exten = s,n,Background(cassandra/OfficeHours) exten = Wait(10) exten = s,n(open),NoOp(open) exten = s,n,WaitExten(0.5); exten = s,n,Set(TIMEOUT(digit)=10) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(cassandra/NextRep) This is the line where I have a problem with dialing an extension. The timing is very fickle and heaps of our callees cannot get to the right extension properly. exten = s,n,WaitExten(10,m[default]) exten = s,n,Macro(dial-us) exten = s,n,Macro(hangupcall) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,open) exten = t,1,Macro(hangupcall) exten = #,1,Macro(hangupcall) ___ -- 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] Call quality - how to debug
I think you're overlooking your internet uplink, which is what I'm talking about: snip Inbound calls are via in IAX trunk from Gradwell. /snip You certainly DO need QOS to maintain call quality over the INTERNET link. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update
Re: [asterisk-users] Call quality - how to debug
Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
Hi Dave, You're quite right, it's a dedicated down and uplink to my ISP, and Gradwell also has fibre connection into that ISP (so short hop to them) The reason I don't think it's the fiber link, is that Asterisk recorded the conversation as two channels. IN (from Gradwell), and OUT (from the Cisco phone, that's on the same LAN as the asterisk server). And I hear distortion on both sides, at the same time. As thats what asterisk hears, and that part of the call is a same-LAN RTP stream, pre-ISP, then that's why I don't think it's the IAX link. That said, I've not got complaints from users making internal calls. So my thinking was maybe its an IAX/SIP conversion thing As a test, I've switched my account, and the problem account to inbound SIP, to see if that makes a difference. That makes it 100% SIP. Next step, memory upgrade and the A*k upgrade. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 16:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? snip So theres no web browsing etc on that 2mb circuit. /snip In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with 50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but 10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, What internet connection do you have... ___ -- 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:
Re: [asterisk-users] Call quality - how to debug
Hi Steve, Mainly because, if it were a CPU utilisation issue, then putting an extra load on the server because of tcpdump isn't going to help. If I go that route then I'll port mirror on the switch. But thanks for the reply, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 16:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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] RES: SIP Response 181 - Is it possible in Asteri sk?
Marco Cordeiro schrieb: Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could I find info about it? IIRC = If I remember correctly. Asterisk trunk is the bleeding-edge development version of Asterisk. See How source code is organized at http://www.asterisk.org/developers/getting-started and Get the source at http://www.asterisk.org/developers/get-source -Mensagem original- [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Marco Cordeiro schrieb: I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html http://www.tech-invite.com/Ti-sip-service-8.html I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] SIP Response 181 - Is it possible in Asterisk?
Philipp Kempgen schrieb: Marco Cordeiro schrieb: I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. However as a rule of thumb you could probably say that SIP B2BUAs send 302 Moved temporarily whereas SIP proxies send 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 maximum user
I think you are looking at this incorrectly, First, if you are seriously looking at thousands of calls, you won't want to be doing this without the help of somebody who is experienced with Asterisk and knows how to solve scaling issues. Secondly, a TDM call between two PRI channels is going to involve very little overhead, and you probably could theoretically handle many hundreds of calls (perhaps even a thousand) like this in a single Asterisk system. But, since that is very expensive to do because PRI ports are not cheap, that is probably not what you are doing with your low end single CPU Asterisk system. Software echo cancellation and transcoding between audio codecs will cost a lot of CPU time, while call recorded will use a high level of hard disk I/O, so thse issues must be considered. Assuming no CPU transcoding, no CPU echo cancelling and no call recording, your worry should be about likely capacity of a single server given your specifics and then scaling issues (i.e. I have a 1/2/4/8/16 CPU Asterisk server handling many calls, but the CPU load is spiking above 2/2/4/8/16. How do I scale to 2+ servers?). Some capacity hints: assuming SIP telephones over a high bandwidth network (g.711 codec by all SIP devices) and multiple SIP or PRI trunks for connecting calls to the PSTN, you might solve this problem by using a modern quad CPU server with redundant power supplies and redundant hard disks and configure 100 SIP phones per server with a Quad port PRI card (TE420B) or SIP trunking for PSTN connections. This might work for a business if the system will hit 50% capacity often with capacity spikes of 75% possible. If the number of calls will normally reach 10% of capacity, you might be able to configured 5 times more SIP phones/users per server. Brandon. On Tue, Jun 2, 2009 at 7:31 AM, M.Monzur Alam mon...@citechco.net wrote: How many asterisk voice user concurrently continue their voice only one Asterisk sever? Could it possible implementation in big environment (more than thousand users)? Hardware statistics: vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping : 9 cpu MHz : 2799.622 cache size : 512 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe up pebs bts cid xtpr bogomips : 5604.90 clflush size : 64 Thanks Monzur ___ -- 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] Segfault on unload of chan_h323 in asterisk-1.4.25
When the support for h323plus was announced for Asterisk 1.4.25, I tried to build this support in Asterisk. For this, I checked out the h323plus CVS from SourceForge, which reported version 1.20.beta5, and also the ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now I see that I get a segfault whenever I issue the command module unload chan_h323, or stop gracefully. I have yet to file a bug on either Asterisk or h323plus because I believe this to be my own error in configuration, rather than an intrinsic bug in chan_h323. Does anyone else have chan_h323 running with h323plus? If so, how did you compile your support? Have you experienced the segfault on shutdown? [r...@rpmbuild64 channels]# ldd /usr/sbin/asterisk libdl.so.2 = /lib64/libdl.so.2 (0x00399180) libcap.so.1 = /lib64/libcap.so.1 (0x00399ca0) libpthread.so.0 = /lib64/libpthread.so.0 (0x003991c0) libtermcap.so.2 = /lib64/libtermcap.so.2 (0x00399ce0) libresolv.so.2 = /lib64/libresolv.so.2 (0x00399a60) libh323_linux_x86_64_n.so.1.20-beta5 = /usr/lib64/libh323_linux_x86_64_n.so.1.20-beta5 (0x003c9240) libpt.so.2.4.2 = /usr/lib64/libpt.so.2.4.2 (0x003c92e0) libssl.so.6 = /lib64/libssl.so.6 (0x00399c60) libcrypto.so.6 = /lib64/libcrypto.so.6 (0x00399a20) libz.so.1 = /usr/lib64/libz.so.1 (0x00399200) libodbc.so.1 = /usr/lib64/libodbc.so.1 (0x00399080) libstdc++.so.6 = /usr/lib64/libstdc++.so.6 (0x003999e0) libm.so.6 = /lib64/libm.so.6 (0x00399140) libgcc_s.so.1 = /lib64/libgcc_s.so.1 (0x00399680) libc.so.6 = /lib64/libc.so.6 (0x00399100) /lib64/ld-linux-x86-64.so.2 (0x00399000) libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x00399c20) libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x00399b60) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x00399ae0) libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x00399b20) libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x00399ba0) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x00399aa0) libselinux.so.1 = /lib64/libselinux.so.1 (0x003995c0) libsepol.so.1 = /lib64/libsepol.so.1 (0x00399600) [r...@rpmbuild64 channels]# ldd /usr/lib64/asterisk/modules/chan_h323.so libh323_linux_x86_64_n.so.1.20-beta5 = /usr/lib64/libh323_linux_x86_64_n.so.1.20-beta5 (0x2af569a25000) libresolv.so.2 = /lib64/libresolv.so.2 (0x2af56a3f1000) libpt.so.2.4.2 = /usr/lib64/libpt.so.2.4.2 (0x2af56a606000) libpthread.so.0 = /lib64/libpthread.so.0 (0x2af56aa84000) libssl.so.6 = /lib64/libssl.so.6 (0x2af56ac9f000) libcrypto.so.6 = /lib64/libcrypto.so.6 (0x2af56aee9000) libdl.so.2 = /lib64/libdl.so.2 (0x2af56b23b000) libz.so.1 = /usr/lib64/libz.so.1 (0x2af56b43f000) libodbc.so.1 = /usr/lib64/libodbc.so.1 (0x2af56b653000) libstdc++.so.6 = /usr/lib64/libstdc++.so.6 (0x2af56b8b8000) libm.so.6 = /lib64/libm.so.6 (0x2af56bbb8000) libgcc_s.so.1 = /lib64/libgcc_s.so.1 (0x2af56be3b000) libc.so.6 = /lib64/libc.so.6 (0x2af56c04a000) /lib64/ld-linux-x86-64.so.2 (0x00399000) libgssapi_krb5.so.2 = /usr/lib64/libgssapi_krb5.so.2 (0x2af56c3a) libkrb5.so.3 = /usr/lib64/libkrb5.so.3 (0x2af56c5cf000) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x2af56c864000) libk5crypto.so.3 = /usr/lib64/libk5crypto.so.3 (0x2af56ca66000) libkrb5support.so.0 = /usr/lib64/libkrb5support.so.0 (0x2af56cc8c000) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x2af56ce94000) libselinux.so.1 = /lib64/libselinux.so.1 (0x2af56d097000) libsepol.so.1 = /lib64/libsepol.so.1 (0x2af56d2af000) -- perl -e '$x=2.3;printf(%.0f + %.0f = %.0f\n,$x,$x,$x+$x);' ___ -- 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] RES: SIP Response 181 - Is it possible in Asteri sk?
Hi Philipp, So, what you are saying is that SIP trunks between 2 Asteriks might be able to handle SIP Response 181 ? Marco -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Enviada em: terça-feira, 2 de junho de 2009 13:06 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] SIP Response 181 - Is it possible in Asterisk? Philipp Kempgen schrieb: Marco Cordeiro schrieb: I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. However as a rule of thumb you could probably say that SIP B2BUAs send 302 Moved temporarily whereas SIP proxies send 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] RES: SIP Response 181 - Is it possible in Asteri sk?
Marco Cordeiro schrieb: So, what you are saying is that SIP trunks between 2 Asteriks might be able to handle SIP Response 181 ? Looks like it, but I didn't test it. (Note to self: Here's the diff: https://reviewboard.asterisk.org/r/201/diff/ ) -Mensagem original- De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen Philipp Kempgen schrieb: Marco Cordeiro schrieb: I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP extension with the forwarding activated, I get only the SIP Response 302 (MOVED_TEMPORARILY) instead of the SIP Response 181 CALL_IS_BEING_FORWARDED). The forwarding of the SIP extensions is being set with AstDB. My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181, or if it would be possible with an Asterisk Server. IIRC Asterisk trunk can send and handle 181 Call is being forwarded. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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] Play a file while transfering a call
Hi, I would like to play a file please wait while we transfer your call ... while dialing I could use music on hold (Dial CMD option m) but, the file can change very frequently and it could be problematic to edit musiconhold.conf and reload everytime there is a new file available. Is there a suggestion on how to simply specify one file ? or create one directory with one file only without having to edit musiconhold.conf ? or is there a different alternative ? winmail.dat___ -- 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] PSTN Interface Card
Dear expert Users Very new to asterisks, seeking an expert advice on the subject matter. Understand that Digium PCI cards could be used to connect Asterisk to POTS/PSTN Lines. However is there any portable cards available that need not be fitted into the PCI slots that could server the same purpose? I have a Fritz!Box 7270 with me, but I am not sure how to use it for the purpose. I have configured the box as a sip client and I can talk to the analog devices attached to the fritz!box from any of my Asterisk clients. However I am not able to use it as a modem to dial any number from my SIP phones to the outside PSTN world. Any experience on this also will be appreciated. Please help. Thanks Manoj ___ -- 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] FritzBox 7270
Thanks hw, you are assumption is right. I want to use the box as an ata. I can make the calls from Astrisk Phones to the outside PSTN world thru the Box in very crude way. However it allows me to call only a pre-configured number in the FritzBox in the form of call forwaring. I would like to work exaclty like a digium card -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 25 May 2009 10:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FritzBox 7270 On Sun, 2009-05-24 at 16:21 +0400, Manoj Panicker - FOES wrote: Hi , Any idea as how to divert the Incoming PSTN calls on the FritzBox to one of the Numbers in the Asterisk domian? and vice versa. I want ot use the FritzBox as the bridge between the PSTN and Astrisk Thanks Manoj __ You mean, you want to use the 7270 as an isdn-ata ? perhaps i'm wrong, but afaics the pbx-part in any DSL-modem works only on the ip-stream (wan/lan). Would be nice, but as far as i know, such a thing, an isdn-ata, does not exist in any appliance. (Perhaps a business opportunaty for someone) hw ___ -- 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] FritzBox 7270
Ngo,I dint know that Astrisk for that matter anything could be installed on the box. I guess I should dwell a bit more. Thanks Manoj -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ngo-Vi Hoai-Anh Sent: 25 May 2009 19:07 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FritzBox 7270 is installing asterisk directly on FritzBox an option for you? If yes I 'v found an interesting link http://www.ip-phone-forum.de/showthread.php?t=146132 Manoj Panicker - FOES schrieb: Hi , Any idea as how to divert the Incoming PSTN calls on the FritzBox to one of the Numbers in the Asterisk domian? and vice versa. I want ot use the FritzBox as the bridge between the PSTN and Astrisk Thanks Manoj -- -- *From:* Manoj Panicker - FOES *Sent:* 24 May 2009 12:39 *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* RE: [asterisk-users] FritzBox 7270 Kare, Thanks much appreciated. It connected as soon as I created a SIP account. However I must try and figure out as how to get this box use IAX2. The vendors are not very helpful. Thanks for your help. Regards Manoj -- -- *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *IT-Connect *Sent:* 20 May 2009 16:59 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] FritzBox 7270 I only tried to connect my 7270 Fritz Box over a sip account on asterisk! There are some points, you have to note: - you have to select Using internet number - in the area select other Provider - in field internet number your asterisk number - in field user name your number too - your password - field registrar a name of your choice - other fields are blank I hope, these are the same options on you Fritz Box, because my gui is in German! regards, Kare Manoj Panicker - FOES schrieb: Dear Users Good day, need a help on connecting the FritzBox with my Asterisk Server. Both are in LAN and from the Asterisk Server I can ping the FritzBox. However the Username I gave in the box is somehow is not geeting registered in the Asterisk application. The usetname I configured in the box is of IAX2 type, is that the reason? Any information on how to connect the FritzBoz 7270 with Asterisk will be appreciated. I did not seem to get much help from the net. Can somebody help? Thanks Manoj - --- ___ -- 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 -- Fa. IT-Connect Karl Tischler Tel.+49 941 705 49 01 Fax +49 941 378 70 39 Mobil +49 160 98 38 23 00 Diese Email und etwaige Anhaenge sind nur fuer den angegebenen Empfaenger bestimmt. Sie kann privilegierte und vertrauliche Informationen enthalten. Wenn der Leser dieser Email nicht der bestimmungsgemaesse Empfaenger oder ein authorisierter Vertreter ist, werden Sie hiermit darauf aufmerksam gemacht, dass jede Weitergabe dieser Email streng verboten ist. Wenn Sie diese Email faelschlicherweise erhalten haben, unterrichten Sie uns bitte sofort und loeschen Sie die Email und etwaige Anhaenge von Ihrem Rechner. This message and any attachments are intended only for the use of the addressee(s) and may contain information that is privileged and/or confidential. If the reader of the message is not the intended recipient(s) or an authorized representative of the intended recipient(s), please do not use, copy, distribute this email or its attachments or take action based on them. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. -- -- ___ -- 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
Re: [asterisk-users] FritzBox 7270
Hans and all My apologies for the delayed response, was not in town. Many thanks for the attempt to help so far. Much appreciated All I want todo is use Fritz as a gateway between my Asterisk and PSTN lines. Remember I don't have any PCI based ATA card. It is the case 2 I require as the VOIP can be handled by Asterisk itself. I did configure fritz to be a SIP client for Asterisk thereby establishing capability to call from any of the Astrisk client phones to call any analog or DECT phones attached to Fritz. But I don't know how do I get to call an outside PSTN number using this integration. However I can always call any one pre-configured PSTN number using the call forwarding feature, however I should be able to use my sogtphone and dial a PSTN number using the integration which is not happening today. Hope this explais Thanks Manoj -Original Message- From: Hans Witvliet [mailto:h...@a-domani.nl] Sent: 26 May 2009 01:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Manoj Panicker - FOES Subject: Re: [asterisk-users] FritzBox 7270 On Mon, 2009-05-25 at 22:19 +0200, Philipp von Klitzing wrote: Hi! looks interesting, indeed, but as the O.P. wanted to divert PSTN call, one would need chan_dahdi.so or chan_misdn.so/chan_capi.so (If the hardware of Fritz is capable of it) Divert-ing is a misleading term in this case. As I said, use the new firmware and register Asterisk to the Fritz!Box as Internet telephone and the OP is ready to go. http://www.ip-phone-forum.de/showpost.php?p=1298093postcount=202 http://www.ip-phone-forum.de/showthread.php?t=184818; Well, to avoid further confusion, lets check with the O.P. ;-) So Manoj, which case were you refering to: case-1: Incoming VOIP-calls are answered by Fritz, and then forwarded to your own Asterisk. (And for outgoing calls, Asterisk uses Fritz as an VOIP-gateway) case-2 Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your own Asterisk. Incoming VOIP-calls are answered by your own Asterisk. In case-1, Fritz has an ordinairy DSL-modem, and on the lan-side there is an VOIP-pbx, and a FXS interface for a local phone. In case-2 Fritz has on the PSTN/ISDN-line an ordinairy DSL-modem, But also a FXO interface. While on the lan-side there is an VOIP-pbx, and a FXS interface for a local phone. AFAICS, case-1 is do-able, but you don't gain anything with it. case-2 would give you two places for incoming calls (voip PSTN) But i wonder if the HW would allow that (no fxo) If case-2 is feasable, i'll dash-off for an 7270 ___ -- 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] FritzBox 7270
Philipp, can you please elaborate, I did mean case 2 onl -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Sent: 26 May 2009 04:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FritzBox 7270 Hi! case-2 Incoming PSTN/ISDN are answered by Fritz, and then forwarded to your own Asterisk. Incoming VOIP-calls are answered by your own Asterisk. - the Fritz!Box usually doesn't answer unless you set it up for voicemail or fax - define forwarded: Do you mean normally an anlog phone would answer, but based on some condition (which?) you now want this call temporarily to go to your Asterisk box? That's what I meant with divert is a misleading term, in the ISDN world diverting means a slightly different thing: Don't call me nor my PBX, call him! In case-2 Fritz has on the PSTN/ISDN-line an ordinairy DSL-modem, But also a FXO interface. While on the lan-side there is an VOIP-pbx, and a FXS interface for a local phone. AFAICS, case-1 is do-able, but you don't gain anything with it. case-2 would give you two places for incoming calls (voip PSTN) But i wonder if the HW would allow that (no fxo) If case-2 is feasable, i'll dash-off for an 7270 Go run! :-) There is ISDN, S0 and also FXO in that box. Philipp ___ -- 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] Realtime LDAP passwords
Where do they currently change their password? If it's somewhere you control, why not add some to create the realmed password? Gavin. On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- 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] Realtime LDAP passwords
It also depends where you are registering your users. If merely using Asterisk for a media server, do the auth via LDAP in Kamailio, which will just use the userPassword attribute (or however the Kamailio LDAP module binds to check auth or what you script it to do) then a normal password change will do. On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- 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] Realtime LDAP passwords
Sorry, lastly I defined it as auxilary to do exactly that; add it to any existing entry. Thanks. On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- 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] Realtime LDAP passwords
One last thing ;-) use OpenLDAP! On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ ___ -- 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] Realtime LDAP passwords
Most of the desktops are KDE and they use the KDE change password facility. It works via pam I believe. Is there an Asterisk interface with pam that would cause it to simultaneously change the Asterisk SIP realm password? If there is, I wonder how we pass it the requisite information? Thanks - John On Tue, 2009-06-02 at 21:04 +0100, Gavin Henry wrote: Where do they currently change their password? If it's somewhere you control, why not add some to create the realmed password? Gavin. On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] does transcoding take place when a SIP call (both ends using same codec) gets forwarded over T1?
I am trying to understand this from a CPU performance perspective. We have SIP Phones/Asterisk using G729 codec. The calls are being forwarded over T1 to the PSTN. 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
Re: [asterisk-users] does transcoding take place when a SIP call (both ends using same codec) gets forwarded over T1?
- Deepak dlal...@gmail.com wrote: I am trying to understand this from a CPU performance perspective. We have SIP Phones/Asterisk using G729 codec. The calls are being forwarded over T1 to the PSTN. Thanks Your Asterisk box will transcode those calls into whatever format is native for your PRI. T1 will be ulaw and E1 will be alaw most likely. --Tim ___ -- 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] Realtime LDAP passwords
grin OpenLDAP isn't an option. And thanks very much for all the responses. I've not had a chance to mock it up yet and see how it works hands on. I am planning that the users ultimately interface SIP to Kamailio and use Asterisk for the call tree, voice mail, conference, etc. I was assuming they would need to authenticate to Asterisk as well as Kamailio but I suppose it may be more a matter of Asterisk trusting Kamailio rather than the individual users. I would also assume voice mail passwords will be very different from user passwords as they should be designed to be entered from a phone keypad rather than a keyboard (I told you I'm a real Asterisk newbie!). I guess I'll find out as I start to set it up. As I want to build it piecemeal and add complexity rather than diving into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with interaction as described above), any suggestions on whether I should build and test Kamailio or Asterisk first? Thanks - John On Tue, 2009-06-02 at 21:08 +0100, Gavin Henry wrote: One last thing ;-) use OpenLDAP! On 02/06/2009, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm afraid I've been dropped into the deep end even though I am an Asterisk novice. I've set up a few tiny, tiny systems in the past and have now been asked to pull together Asterisk, FreePBX, Kamailio, RTPProxy, and Fedora Directory Server into a VoIP service. After googling and reading for most of the last 24 hours, I finally have my head around the components and how they work but am a little stumped by password synchronization using existing LDAP accounts. Maintaining separate accounts with a shared database between Kamailio and Asterisk seems quite reasonable. Integrating with the existing LDAP database seems like much more of a challenge. I did find http://www-rocq.inria.fr/who/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html and http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/ very helpful. For security reasons, we keep internal UIDs different from public email IDs. Thus, we might use john.doe internally and j...@example.com for email. Since it is a multi-tenant environment, I'd imagine we will use the Kamailio domain module, make the SIP domain match the email domain, and use the email user portion of the email address as the SIP ID. I think this is straightforward using LDAP and Kamailio as we would query LDAP for the email address and have return the password. Asterisk seems a little trickier. I've looked at the schema extensions and it looks like we add an auxiliary objectclass of AstSIPUser. I suppose we would add this objectclass to a structure inetOrgPerson object. We could then use the email name for the AstAccountName (or whatever the actual attribute is) but the password befuddles me. I notice we add an AstAccountRealmedPassword attribute. I suppose this is because of the need to furnish SIP a hash derived from username:realm:password. We would prefer our users only need to change their passwords in one place. Is there anyway beside deploying something like IPA to have Asterisk use the regular posix password stored in LDAP rather than a separate AstAccountRealmedPassword? I'm looking forward to diving in; I just wish it was with a little less time pressure! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Realtime LDAP passwords
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com: Most of the desktops are KDE and they use the KDE change password facility. It works via pam I believe. Is there an Asterisk interface with pam that would cause it to simultaneously change the Asterisk SIP realm password? If there is, I wonder how we pass it the requisite information? Thanks - John No, but you could write one. You never mentioned how Asterisk is used with Kamailio? http://search.cpan.org/~nikip/Authen-PAM-0.16/d/PAM.pm -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.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] Realtime LDAP passwords
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com: grin OpenLDAP isn't an option. And thanks very much for all the responses. I've not had a chance to mock it up yet and see how it works hands on. I am planning that the users ultimately interface SIP to Kamailio and use Asterisk for the call tree, voice mail, conference, etc. I was assuming they would need to authenticate to Asterisk as well as Kamailio but I suppose it may be more a matter of Asterisk trusting Kamailio rather than the individual users. I would also assume voice mail passwords will be very different from user passwords as they should be designed to be entered from a phone keypad rather than a keyboard (I told you I'm a real Asterisk newbie!). I guess I'll find out as I start to set it up. OK, depends how you set it up. You might not authenticate at all like some ITSPs do (based on IP). Is this for your company? I committed a patch for voicemail passwords in the Asterisk LDAP schema last week, so you'll need svn for that: https://issues.asterisk.org/view.php?id=15155 As I want to build it piecemeal and add complexity rather than diving into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with interaction as described above), any suggestions on whether I should build and test Kamailio or Asterisk first? Thanks - John So, Asterisk and FreePBX? Why both? This is a mighty big pie to take a bite out of, so it doesn't really matter. Kamailio is harder is you don't know SIP. Depends, depends, depends ;-) What is the overall project goal here? We should have asked that first. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.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] Realtime LDAP passwords
Thanks. I do appreciate the input as I am jumping into the deep end as I said :) On Tue, 2009-06-02 at 21:43 +0100, Gavin Henry wrote: 2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com: grin OpenLDAP isn't an option. And thanks very much for all the responses. I've not had a chance to mock it up yet and see how it works hands on. I am planning that the users ultimately interface SIP to Kamailio and use Asterisk for the call tree, voice mail, conference, etc. I was assuming they would need to authenticate to Asterisk as well as Kamailio but I suppose it may be more a matter of Asterisk trusting Kamailio rather than the individual users. I would also assume voice mail passwords will be very different from user passwords as they should be designed to be entered from a phone keypad rather than a keyboard (I told you I'm a real Asterisk newbie!). I guess I'll find out as I start to set it up. OK, depends how you set it up. You might not authenticate at all like some ITSPs do (based on IP). Is this for your company? We are launching a new company whose primary product is a complete, hosted, virtualized environment including desktops for micro-businesses, charitable organizations, schools, and municipalities. Unexpectedly, though not surprisingly, our initial customers are asking for a VoIP solution utilizing the same infrastructure. Hence the plunge into VoIP. We will be contracting with an ITSP for SIP trunking into our data center and need to set up the whole shooting match. I committed a patch for voicemail passwords in the Asterisk LDAP schema last week, so you'll need svn for that: https://issues.asterisk.org/view.php?id=15155 As I want to build it piecemeal and add complexity rather than diving into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with interaction as described above), any suggestions on whether I should build and test Kamailio or Asterisk first? Thanks - John So, Asterisk and FreePBX? Why both? From looking at the press release for AsteriskNOW (which I don't plan to use as I'd like a little tighter control over the system), it appears FreePBX and Asterisk 1.6 are a nice pairing and might ease some of our administration. Just going on what I'm reading and not experience. This is a mighty big pie to take a bite out of, so it doesn't really matter. Kamailio is harder is you don't know SIP. Depends, depends, depends ;-) I'm reasonably comfortable with protocols and how they work (my background is as a network engineer although the skills are a bit rusty). SIP seems quite comprehensible and all the docs I read through the night on the innards of Kamailio and SER made perfect sense. What is the overall project goal here? We should have asked that first. In effect, we will become a voice aggregator for micro-businesses and a shared PBX services provider to complement our data offerings. I was going to build Asterisk first to have complete standalone functionality but, if the user authentication will be primarily to Kamailio, it may make sense to start there. I'll probably circle the pool a few times and then jump in wherever I stop unless someone with more experiences advises specifically! Thanks again - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Realtime LDAP passwords
2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com: Thanks. I do appreciate the input as I am jumping into the deep end as I said :) On Tue, 2009-06-02 at 21:43 +0100, Gavin Henry wrote: 2009/6/2 John A. Sullivan III jsulli...@opensourcedevel.com: grin OpenLDAP isn't an option. And thanks very much for all the responses. I've not had a chance to mock it up yet and see how it works hands on. I am planning that the users ultimately interface SIP to Kamailio and use Asterisk for the call tree, voice mail, conference, etc. I was assuming they would need to authenticate to Asterisk as well as Kamailio but I suppose it may be more a matter of Asterisk trusting Kamailio rather than the individual users. I would also assume voice mail passwords will be very different from user passwords as they should be designed to be entered from a phone keypad rather than a keyboard (I told you I'm a real Asterisk newbie!). I guess I'll find out as I start to set it up. OK, depends how you set it up. You might not authenticate at all like some ITSPs do (based on IP). Is this for your company? We are launching a new company whose primary product is a complete, hosted, virtualized environment including desktops for micro-businesses, charitable organizations, schools, and municipalities. Unexpectedly, though not surprisingly, our initial customers are asking for a VoIP solution utilizing the same infrastructure. Hence the plunge into VoIP. We will be contracting with an ITSP for SIP trunking into our data center and need to set up the whole shooting match. OK, to be honest then, since it's for a commercial solution and you're so new, I'd buy something. I've seen: http://www.sipwise.com/index.php/products?start=2 http://www.asipto.com/ http://www.voice-system.ro/ I prefer the last one, but all vary on price and the money spent will be saved on your dev time and learning curve. Then send yourself to the training course. That way you know all the loop holes are closed to allowing fraudulent calls etc. I committed a patch for voicemail passwords in the Asterisk LDAP schema last week, so you'll need svn for that: https://issues.asterisk.org/view.php?id=15155 As I want to build it piecemeal and add complexity rather than diving into the end product (RTPProxy, Kamailio, Asterisk, FreePBX with interaction as described above), any suggestions on whether I should build and test Kamailio or Asterisk first? Thanks - John So, Asterisk and FreePBX? Why both? From looking at the press release for AsteriskNOW (which I don't plan to use as I'd like a little tighter control over the system), it appears FreePBX and Asterisk 1.6 are a nice pairing and might ease some of our administration. Just going on what I'm reading and not experience. Sorry, I thought I read FreeSWITCH! This is a mighty big pie to take a bite out of, so it doesn't really matter. Kamailio is harder is you don't know SIP. Depends, depends, depends ;-) I'm reasonably comfortable with protocols and how they work (my background is as a network engineer although the skills are a bit rusty). SIP seems quite comprehensible and all the docs I read through the night on the innards of Kamailio and SER made perfect sense. What is the overall project goal here? We should have asked that first. In effect, we will become a voice aggregator for micro-businesses and a shared PBX services provider to complement our data offerings. I was going to build Asterisk first to have complete standalone functionality but, if the user authentication will be primarily to Kamailio, it may make sense to start there. I'll probably circle the pool a few times and then jump in wherever I stop unless someone with more experiences advises specifically! Thanks again - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.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] extensions not being detected consistently
Hi Brandon, Thanks for getting back to me about my problem, I really appreciate it (going a bit nuts with this problem) I made the changes that you suggested but I can't tell if they've made a difference or not, as the problem seems to have shifted a bit but is still inconsistent and difficult to pin point. However you did get me thinking about what to try next and that you mentioned dialing extensions in macros could be a problem was news to me. Our dialplan is a bit of a kluge so I'm going to fix it first and see if the problem goes away on its own. Either way, thanks for your help! Cheers, Terry Brandon B. wrote: Extensions that are dialed within macros like the following lines could cause the type of problems as you mentioned: exten = s,n,Macro(dial-us) exten = s,n,Macro(hangupcall) This line: exten = s,n,Wait(0.5); should be changed to exten = s,n,WaitExten(0.5); and these lines: exten = Wait(10) exten = s,n(open),NoOp(open) are not valid. Try this: exten = s,1,Set(TIMEOUT(digit)=10) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(cassandra/CustomWelcomeMessage) exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open) exten = s,n,Background(cassandra/OfficeHours) exten = s,n,Background(cassandra/NextRep) exten = t,n,Macro(dial-us) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,open) exten = #,1,Macro(hangupcall) On Mon, Jun 1, 2009 at 1:47 PM, Terry Nathan tnat...@aiinc.ca wrote: G'afternoon everybody, I'm having a problem with consistently being able to ring our extensions from an outside line. I don't have a problem reaching the number, but during our calls to Background(msg) that I am having a problem. It seems to be an issue with timing. If I press the extension towards the end of the Background(msg) the it often works. However, in the middle of the message it will not work at all. What is also strange is that I can dial an extension any time if I call from one of our ip phones. This seems to be strictly a problem with regular phones, then the timing of dialing the extension becomes important. The fact that the ip phones always work seems to suggest that I need to look at tone detection, but after googling and searching the bowels of every conf file I could find, I haven't found any magic bullet. I should mention that the first call to Background() usually works, even for the regular phones, I think this is because it is short enough that the timing of dialing the extension is relatively easy. I don't know if it is significant or not but it seems that once a callee tries to dial an extension and it doesn't work, even the next few calls will also not work. And similarly, sometimes it works and then a few calls will go through, but then it will go back to not detecting properly again. Asterisk is running on its own box and there is nothing unusual happening with the system, or even people on other lines, that is happening. Checking the log files when I call in Asterisk tells me that either it only detects 1 of the 3 digits (usually the second or third one) or, if I dial the extension at a different point in the message, that the first digit was pressed twice e.g. '22' instead of just '2'. The inconsistency of the problem is starting to drive me bonkers as I can't accurately nail down the problem. Ideally I'd like our callees to be able to dial an extension as soon as the call to Background() is hit in the context, from any phone that calls in, not just ip phones. My setup is an installation of asterisk-now (Centos 5 with Asterisk 1.4.24) If anyone has seen a problem like this before or has even an inkling of what it might be, that would be awesome :D Thanks in advance. My dial plan: [incoming-our-number] exten = s,1,Answer exten = s,n,NoOp(incoming-our-number) exten = s,n,Background(cassandra/CustomWelcomeMessage) This line is usual fine, I think because the message is short enough that timing the dialing of the extension is less of an issue. exten = s,n,GotoIfTime(09:00-17:30,mon-fri,*,*?open) exten = s,n,Wait(0.5); exten = s,n,Background(cassandra/OfficeHours) exten = Wait(10) exten = s,n(open),NoOp(open) exten = s,n,WaitExten(0.5); exten = s,n,Set(TIMEOUT(digit)=10) exten = s,n,Set(TIMEOUT(response)=15) exten = s,n,Background(cassandra/NextRep) This is the line where I have a problem with dialing an extension. The timing is very fickle and heaps of our callees cannot get to the right extension properly. exten = s,n,WaitExten(10,m[default]) exten = s,n,Macro(dial-us) exten = s,n,Macro(hangupcall) exten = i,1,Playback(pbx-invalid) exten = i,2,Goto(s,open) exten = t,1,Macro(hangupcall) exten = #,1,Macro(hangupcall) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] 400 calls at g711 how much cpu power
I totally agree with you Jeff, however some of us do not actually sell viagra over the phone. This is a campaign to spread a message to the population about the health prevention steps that should be taken in order to prevent diseases that are affecting our population. I do understand all of you to be reluctant to help with this post. However judging before listening has been the most devastating problem humans have. We simply do not trust each other. However, just for the sake of posterity: Hardware/Software just one server Dell 2950 / 4GB RAM / four 72Gb ultra320 SCSI hard disks built as RAID-0 Debian as the OS (in 32 bit mode) Asterisk 32 bit 1.4 compiled manually (codecs removed, modules removed,etc, a ton of pure CRAP out!) Only g711/SIP was used 20 second clip was served from ramdisk Dialer: SmoothTorque (those guys simply ROCK!)( setup outbound mode ONLY!) Network: 50 Mbit fiber link to telco provider. Pure IP, no QoS. We were pumping 3k calls-setup/second to the session controller at telco's side. Until we reached controller's max of 10k calls. Server load was NEVER above 3.2 thanks to all for your help. On Thu, Apr 2, 2009 at 7:36 PM, Jon Pounder j...@inline.net wrote: Erick, how about posting your home phone number here so we can all call you and play a 20second audio clip - I am sure you would see nothing wrong with that would you ? ContactTel Business wrote: Your right, i don't think we would help someone asking on advice to send 1 million emails for Viagra would we ? So why the hell aren't we thinking straight and tell the poor guy? Ive seen dialer app that where legit, even worked on some for the military. But this is just spam /pham (phone spam) send 10USD to my email ;) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: April-02-09 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 400 calls at g711 how much cpu power My only comment is that I am having moral issues with assisting anyone that is planning to call one million phone numbers to play a message and hang up. Doesn't sound like an opt-in kind of campaign to me. When such a thing happens to me on my home phone I get extremely angry. j On Wed, 1 Apr 2009, Erick Perez wrote: We are planning to run an outbound only campaign. A 20-second voice message will be played to callers and our dialer on machine1 will send to machine2-asterisk (1.4) instructions to dial 400 calls, play the message and hang up. This will be done for about 1 million phones. The asterisk box will communicate via SIP to a voice carrier. the voice carrier will then place the calls on pstn. The codec will be g711. So we will never do any transcoding. I have been calculating the CPU power required to do the calls and in previous posting the usual calculation is about 40MHZ per leg when no transcoding is involved. So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. Comments? -- Erick ___ -- 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 -- Erick Perez Cel +(507) 6675-5083 ___ -- 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] 400 calls at g711 how much cpu power
On Wed, 2009-04-01 at 22:46 -0500, Erick Perez wrote: So if we use the 40MHZ rule, we are talking about 40*400=16000MHZ or 1.6Ghz. It's been my experience that CPU load of Asterisk don't scale linearly with call volume. I don't pretend to understand all the reasons why, but it probably has a lot to do with call structures inside of Asterisk. For example, searching a linked list is simple when there are only a few items in the list, but the more items that get added to the list, the more CPU time it takes to finish the task, on average. I know the Asterisk developers spent a lot of time and effort improving the performance of the internal structures between the 1.4 branch and the 1.6.0 branch... if I were you, I'd at least give the 1.6.0 branch a shot. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] extensions not being detected consistently
If you are using an analog card (dahdi or zaptel driver) - have you tried, in [chan_dahdi.conf | zapata.conf]: relaxdtmf=yes I had similar symptoms as the paragraph below, but it was only for a small handful of specific callers and not repeatable on demand. relaxdtmf made a significant improvement to the rate of DTMF false positive/negatives. - Original Message - Checking the log files when I call in Asterisk tells me that either it only detects 1 of the 3 digits (usually the second or third one) or, if I dial the extension at a different point in the message, that the first digit was pressed twice e.g. '22' instead of just '2'. The inconsistency of the problem is starting to drive me bonkers as I can't accurately nail down the problem. ___ -- 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] Play a file while transfering a call
Hi, I do this by creating a directory waitingtransfer with only 1 file (the audio message, the name isn't important, so you can change it everytime you want) and then add new musiconhold class with specific waitingtransfer directory. In your extensions.conf you change the musiconhold class to waitingmessage class, and that's it! For me works great! 2009/6/2 Julien Chavanton j...@atlastelecom.com Hi, I would like to play a file please wait while we transfer your call ... while dialing I could use music on hold (Dial CMD option m) but, the file can change very frequently and it could be problematic to edit musiconhold.conf and reload everytime there is a new file available. Is there a suggestion on how to simply specify one file ? or create one directory with one file only without having to edit musiconhold.conf ? or is there a different alternative ? ___ -- 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
