[asterisk-users] Choppy audio
Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager proxy
Hello all, Some one is using asterisk and queuemetrics connected via astmanproxy? How about your experience? Which proxy do you use in this kind of connection? In my instalation asterisk and Queuemetrics are installed on diferent machines and I want to avoid manager problems Thanks in advance. VoipCrazy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] line goes silent for a few seconds at the start of outgoing calls
Hi there, I am experiencing a strange problem and am looking for advice to where to start looking. Or any clues, really. I have Asterisk running on our router, and it is configured to forward calls to a provider out there (who is also using Asterisk). On the inside of the Asterisk are several voip clients, among which a Siemens C450ip. If that phone makes a call, there's a few seconds right after the peer picks up when neither side can hear the other; after a few seconds, everything works. This only happens if that phone calls an outside phone via that provider; I cannot reproduce it from another phone via the same provider, nor if the Siemens phone is routed via a different provider. Also, internal calls and incoming calls via the provider to the Siemens are fine. I tried debugging but I could not find any hints. I am led to believe that there is some codec stuff going on, but I cannot figure out what, especially since my Asterisk does canreinvite=no, so the codecs are negotiated between my Asterisk and the phone and my Asterisk and the provider independently. Yet, it needs the Siemens on one side of the Asterisk, and this specific provider on the other side for the problem to appear. What's going on? How can I fix this? Where should I look? Thanks, -- martin; (greetings from the heart of the sun.) \ echo mailto: !#^.*|tr * mailto:; [EMAIL PROTECTED] with sufficient thrust, pigs fly just fine. however, this is not necessarily a good idea. it is hard to be sure where they are going to land, and it could be dangerous sitting under them as they fly overhead. -- rfc 1925 spamtraps: [EMAIL PROTECTED] digital_signature_gpg.asc Description: Digital signature (see http://martin-krafft.net/gpg/) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User unable to use DTMFs?
Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. --- On Tue, 7/1/08, Vincent [EMAIL PROTECTED] wrote: From: Vincent [EMAIL PROTECTED] Subject: [asterisk-users] User unable to use DTMFs? To: asterisk-users@lists.digium.com Date: Tuesday, July 1, 2008, 11:09 AM Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Click to Dial Service Providers in Australia
Is anyone on this list aware of any Click to Dial Service Providers in Australia. Eg. Someone who offers REST web services for clients web pages so they can implement Click to Call on a real estates web page OR Someone who offers value add voip services to real estate agents in Australia. I have a client looking to purchase services for something like this to Australian visitors of his website. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Panama SIP ITSP?
Does anyone know of an ITSP's offering single line SIP services in Panama. Mainly will be inbound with very low outbound (10 calls a month max) so cheapest per month costs is what's required. Also do you know what the deal with number portability is in Panama if this company uses the number but finds the service sucks and wants to go with another voip provider? Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call quality
Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User unable to use DTMFs?
On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob [EMAIL PROTECTED] wrote: Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. Users call into our Asterisk voice server through a Zaptel PCI interface from regular phones, usually from a PBX (virtually all of them ISDN-based). The only files I modified are zaptel.conf, zapata.conf, and extensions.conf, which don't have anything DTMF-related, so Asterisk uses the default options. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
2008/7/1 Loic Didelot [EMAIL PROTECTED]: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Which version of asterisk/zaptel, and which echo canceler is running in Zaptel? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I tried to get a little into cpu utilization and found the following results. Can they help me to come to a conclusion? Best regards, Loic Didelot. [EMAIL PROTECTED]:~# mpstat 1 Linux 2.6.22-14-server (ppsite1)07/01/2008 02:54:40 PM CPU %user %nice%sys %iowait%irq %soft %steal %idleintr/s 02:54:41 PM all0.000.001.000.000.000.000.00 99.00 4210.00 02:54:42 PM all0.000.000.000.000.000.000.00 100.00 4207.00 02:54:43 PM all0.000.000.000.000.000.000.00 100.00 4208.00 02:54:44 PM all0.000.000.000.00 45.000.000.00 55.00 4127.00 02:54:45 PM all0.000.000.000.00 97.000.000.00 3.00 4148.00 02:54:46 PM all0.000.000.000.00 93.004.000.00 3.00 4195.00 02:54:47 PM all0.000.000.000.00 92.006.000.00 2.00 4175.00 02:54:48 PM all0.000.000.001.00 91.002.000.00 6.00 4154.00 02:54:49 PM all0.000.000.000.00 100.000.000.00 0.00 4069.00 02:54:50 PM all0.000.000.000.00 23.000.000.00 77.00 4125.00 02:54:51 PM all2.000.000.000.00 19.000.000.00 79.00 4123.00 02:54:52 PM all0.000.000.000.000.000.000.00 100.00 4236.00 02:54:53 PM all0.000.006.002.000.003.000.00 89.00 4302.00 02:54:54 PM all0.000.005.000.000.003.000.00 92.00 4267.00 02:54:55 PM all0.000.00 20.000.000.001.000.00 79.00 4328.00 02:54:56 PM all0.000.000.000.009.000.000.00 91.00 4352.00 02:54:57 PM all0.000.000.00 49.00 46.000.000.00 5.00 4376.00 02:54:58 PM all0.000.000.000.000.000.000.00 100.00 4350.00 02:54:59 PM all 11.000.002.00 36.000.000.000.00 51.00 4237.00 02:55:00 PM all0.000.000.00 100.000.000.000.00 0.00 4221.00 02:55:01 PM all1.000.001.00 62.00 36.000.000.00 0.00 4318.00 02:55:02 PM all0.000.000.002.00 98.000.000.00 0.00 4219.00 02:55:03 PM all0.000.000.000.00 100.000.000.00 0.00 4342.00 02:55:04 PM all0.000.000.002.00 98.000.000.00 0.00 4236.00 02:55:05 PM all 14.000.004.00 20.00 62.000.000.00 0.00 4229.00 02:55:06 PM all 39.000.003.00 38.00 18.001.000.00 1.00 4346.00 02:55:07 PM all8.000.008.00 79.003.001.000.00 1.00 4240.00 02:55:08 PM all1.000.000.00 98.000.000.000.00 1.00 4217.00 02:55:09 PM all0.000.001.006.000.000.000.00 93.00 4167.00 02:55:10 PM all0.000.000.000.00 25.000.000.00 75.00 4132.00 02:55:11 PM all0.000.000.000.00 75.000.000.00 25.00 4117.00 02:55:12 PM all0.000.000.000.00 53.000.000.00 47.00 4130.00 02:55:13 PM all0.000.000.000.000.000.000.00 100.00 4103.00 02:55:14 PM all0.000.000.00 50.000.000.000.00 50.00 4124.00 02:55:15 PM all0.000.001.000.000.000.000.00 99.00 4216.00 02:55:16 PM all1.000.000.000.00 32.000.000.00 67.00 4214.00 02:55:17 PM all0.000.000.000.00 98.000.000.00 2.00 4209.00 02:55:18 PM all0.000.000.000.00 94.000.000.00 6.00 4220.00 02:55:19 PM all0.000.000.000.00 58.000.000.00 42.00 4216.00 02:55:20 PM all1.000.000.000.000.000.000.00 99.00 4204.00 02:55:21 PM all1.000.000.000.000.000.000.00 99.00 4210.00 02:55:22 PM all1.000.000.000.000.000.000.00 99.00 4234.00 02:55:23 PM all0.000.001.000.000.000.000.00 99.00 4202.00 02:55:24 PM all0.000.001.000.000.000.000.00 99.00 4109.00 02:55:25 PM all1.000.001.001.00 53.001.000.00 43.00 4179.00 02:55:26 PM all0.000.000.000.00 35.000.000.00 65.00 4213.00 02:55:27 PM all0.000.001.000.000.000.000.00 99.00 4204.00 02:55:28 PM all0.000.001.000.000.000.000.00 99.00 4169.00 02:55:29 PM all0.000.003.960.00 37.620.990.00 57.43 4149.50 02:55:30 PM all0.000.001.000.000.000.000.00 99.00 4208.00 02:55:31 PM all0.000.000.00 16.003.000.000.00
Re: [asterisk-users] Call quality
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com Is this a new install or a new problem? If it is a new problem, what has changed? If it is a new install, I would not rule out the provider, the more historic may or may not be a good thing. Describe the audio when it is poor, popping, clicking, hissing? Have you tried running a debug on the spans? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I/O wait is very suspicious. What is your hardware platform? Is this just a plain Jane PBX or are you doing anything unusual? Thanks, Steve T On Tue, Jul 1, 2008 at 8:57 AM, Loic Didelot [EMAIL PROTECTED] wrote: I tried to get a little into cpu utilization and found the following results. Can they help me to come to a conclusion? Best regards, Loic Didelot. [EMAIL PROTECTED]:~# mpstat 1 Linux 2.6.22-14-server (ppsite1)07/01/2008 02:54:40 PM CPU %user %nice%sys %iowait%irq %soft %steal %idleintr/s 02:54:41 PM all0.000.001.000.000.000.000.00 99.00 4210.00 02:54:42 PM all0.000.000.000.000.000.000.00 100.00 4207.00 02:54:43 PM all0.000.000.000.000.000.000.00 100.00 4208.00 02:54:44 PM all0.000.000.000.00 45.000.000.00 55.00 4127.00 02:54:45 PM all0.000.000.000.00 97.000.000.00 3.00 4148.00 02:54:46 PM all0.000.000.000.00 93.004.000.00 3.00 4195.00 02:54:47 PM all0.000.000.000.00 92.006.000.00 2.00 4175.00 02:54:48 PM all0.000.000.001.00 91.002.000.00 6.00 4154.00 02:54:49 PM all0.000.000.000.00 100.000.000.00 0.00 4069.00 02:54:50 PM all0.000.000.000.00 23.000.000.00 77.00 4125.00 02:54:51 PM all2.000.000.000.00 19.000.000.00 79.00 4123.00 02:54:52 PM all0.000.000.000.000.000.000.00 100.00 4236.00 02:54:53 PM all0.000.006.002.000.003.000.00 89.00 4302.00 02:54:54 PM all0.000.005.000.000.003.000.00 92.00 4267.00 02:54:55 PM all0.000.00 20.000.000.001.000.00 79.00 4328.00 02:54:56 PM all0.000.000.000.009.000.000.00 91.00 4352.00 02:54:57 PM all0.000.000.00 49.00 46.000.000.00 5.00 4376.00 02:54:58 PM all0.000.000.000.000.000.000.00 100.00 4350.00 02:54:59 PM all 11.000.002.00 36.000.000.000.00 51.00 4237.00 02:55:00 PM all0.000.000.00 100.000.000.000.00 0.00 4221.00 02:55:01 PM all1.000.001.00 62.00 36.000.000.00 0.00 4318.00 02:55:02 PM all0.000.000.002.00 98.000.000.00 0.00 4219.00 02:55:03 PM all0.000.000.000.00 100.000.000.00 0.00 4342.00 02:55:04 PM all0.000.000.002.00 98.000.000.00 0.00 4236.00 02:55:05 PM all 14.000.004.00 20.00 62.000.000.00 0.00 4229.00 02:55:06 PM all 39.000.003.00 38.00 18.001.000.00 1.00 4346.00 02:55:07 PM all8.000.008.00 79.003.001.000.00 1.00 4240.00 02:55:08 PM all1.000.000.00 98.000.000.000.00 1.00 4217.00 02:55:09 PM all0.000.001.006.000.000.000.00 93.00 4167.00 02:55:10 PM all0.000.000.000.00 25.000.000.00 75.00 4132.00 02:55:11 PM all0.000.000.000.00 75.000.000.00 25.00 4117.00 02:55:12 PM all0.000.000.000.00 53.000.000.00 47.00 4130.00 02:55:13 PM all0.000.000.000.000.000.000.00 100.00 4103.00 02:55:14 PM all0.000.000.00 50.000.000.000.00 50.00 4124.00 02:55:15 PM all0.000.001.000.000.000.000.00 99.00 4216.00 02:55:16 PM all1.000.000.000.00 32.000.000.00 67.00 4214.00 02:55:17 PM all0.000.000.000.00 98.000.000.00 2.00 4209.00 02:55:18 PM all0.000.000.000.00 94.000.000.00 6.00 4220.00 02:55:19 PM all0.000.000.000.00 58.000.000.00 42.00 4216.00 02:55:20 PM all1.000.000.000.000.000.000.00 99.00 4204.00 02:55:21 PM all1.000.000.000.000.000.000.00 99.00 4210.00 02:55:22 PM all1.000.000.000.000.000.000.00 99.00 4234.00 02:55:23 PM all0.000.001.000.000.000.000.00 99.00 4202.00 02:55:24 PM all0.000.001.000.000.000.000.00 99.00 4109.00 02:55:25 PM all1.000.001.001.00 53.001.000.00 43.00 4179.00 02:55:26 PM all0.000.000.000.00 35.000.000.00 65.00 4213.00 02:55:27 PM all0.000.001.000.000.000.000.00 99.00 4204.00 02:55:28 PM all0.00
Re: [asterisk-users] Call quality
Hi, its a new installation in a new office. Customer moved in, so right moment to get a new PBX. The box is running asterisk, nothing else: - asterisk - postfix just to send out voicemails - no realtime - som AGIS at call setup and call end - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2 - zaptel-1.4.10 We use a Junghann BRI card and a XORCOMM Analog Astribank. But only one modem and 2 fax devices are connected to the astribank. I did not do a debug on the spans. Anythin special I should look for? Difficult to describe the audio: - basically echo is appearing - audio problems are only one way - audio has cuts when speaking Best regards, Loic Didelot. On Tue, 2008-07-01 at 08:58 -0400, Steve Totaro wrote: On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. Best regards, Loic Didelot. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com Is this a new install or a new problem? If it is a new problem, what has changed? If it is a new install, I would not rule out the provider, the more historic may or may not be a good thing. Describe the audio when it is poor, popping, clicking, hissing? Have you tried running a debug on the spans? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote: Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. If there are no legal reasons not to, consider recording all calls for a limited time. It's easier for engineers to debug a voice quality problem when they have a recording of exactly what it sounds like. It's possible that different people are complaining about different perceptions of what they consider a voice quality problem, and that the problem might not even be on your end of the conversation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hello, I forgot to include CPU information [EMAIL PROTECTED]:/usr/src/bristuff-0.4.0-RC2# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1000MHz stepping: 9 cpu MHz : 1000.127 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips: 2002.19 clflush size: 64 The box is running asterisk, nothing else: - asterisk - postfix just to send out voicemails - no realtime - som AGIS at call setup and call end - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2 - zaptel-1.4.10 Best regards, Loic Didelot. On Tue, 2008-07-01 at 13:54 +0100, Steve Davies wrote: 2008/7/1 Loic Didelot [EMAIL PROTECTED]: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Which version of asterisk/zaptel, and which echo canceler is running in Zaptel? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I considered that, but I fear that this would load the machine even more. So I guess I should take a more powerful box with a good harddrive (at the moment I have a solid state flash card) and start recording calls. Best regards, Loic Didelot. On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote: On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote: Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. If there are no legal reasons not to, consider recording all calls for a limited time. It's easier for engineers to debug a voice quality problem when they have a recording of exactly what it sounds like. It's possible that different people are complaining about different perceptions of what they consider a voice quality problem, and that the problem might not even be on your end of the conversation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Disto choice for Asterisk with AVM Fritz!PCI cards
El mar, 01-07-2008 a las 06:10 +0200, Dave Cotton escribió: I go along with the above, Ive done this with Mandrake 10.1 and OpenSuse 10.1 and 10.3, What I found was that with the Mandrake I used chan_capi and patched the Suse supplied driver code to work with 2 Frtz cards a lot of work. With the Suse installs I switched to chan_misdn no patching and the config was handled bu misdn_init config automagically. I'me really happy with debian. Always you can use apt-get to install asterisk and modules without pain ;) Latest zaptel modules from debian repository have OSLEC as default echo canceler and works like a charm. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Between IAX Trunks
Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can´t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] music on hold realtime
Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
El mar, 01-07-2008 a las 10:35 -0300, Gustavo A Gonzalez escribió: Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can´t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. I've the same setup with FreePBX and NVFaxDetect. Works fine. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
OK, there could be a few items here: 1 Faxes usually do not work over straight IP. I know they can and many including myself have had success the mechanics of the IP network usually won't allow it. 2 If you are using anything other than a/u law forget it. 3 What is the connectivity between the Asterisk boxes? If they are plugged into the same private backbone (ie Same Ethernet network) you have a fighting chance. If not, it is up to the Gods of Network congestion. 4 You may want to read up on the IAXModem pages, Mr, Underwood, explains the use of IAXModem and gives a few pointers on useing the IAXModem to terminal/originate calls on the same machine that has the TDM circuits and use remote serial to bring the fax data back to the hylafax machine. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo A Gonzalez Sent: Tuesday, July 01, 2008 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fax Between IAX Trunks Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can´t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
As an addendum to my original message... In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, Doug On Tue, 1 Jul 2008, Doug Crompton wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I just switched over to this system from an older SUSE 2.4.10 kernel system. I am getting choppy audio in voicemail and general message playback. I installed Zaptel and ztdummy module and the following is zaptel status: slate:/etc/init.d # cat /proc/zaptel/1 Span 1: ZTDUMMY/1 ZTDUMMY/1 (source: RTC) 1 Is this indicating proper installation? Is there anything else I should try/do?? The choppyness is not extreme, just not perfect. I had no problem in my old system with 2.4. I had not even installed zaptel or ztdummy there. Doug * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.21 and CUT function
Hi all, does anybody know how to cut a chain using the pipe delimiter? I tried to escape it or to use $'x7c' as delimiter, no luck. Thanks for any help. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Recording the calls may or may not reveal an issue. I have personally done this exact same method of troubleshooting only to find the recordings were perfect but not the actual calls. I think you should try just putting a regular server in place of your appliance and then test. I have a feeling the I/O is choking your system, similar to recording many simultaneous calls, which to me would indicate a flash bottleneck. At least put in a real HD and copy over your configs. Thanks, Steve T On Tue, Jul 1, 2008 at 9:15 AM, Loic Didelot [EMAIL PROTECTED] wrote: I considered that, but I fear that this would load the machine even more. So I guess I should take a more powerful box with a good harddrive (at the moment I have a solid state flash card) and start recording calls. Best regards, Loic Didelot. On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote: On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot [EMAIL PROTECTED] wrote: Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. If there are no legal reasons not to, consider recording all calls for a limited time. It's easier for engineers to debug a voice quality problem when they have a recording of exactly what it sounds like. It's possible that different people are complaining about different perceptions of what they consider a voice quality problem, and that the problem might not even be on your end of the conversation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold realtime
If by realtime, you mean to be able to read the MOH class from a DB and set MusicOnHold, then I think you should try func_odbc. Have never tried it, but reading the workings of it, it seems to be possible to achieve this. Let me know if you succeed in it. - Ben. --- On Tue, 7/1/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: [asterisk-users] music on hold realtime To: asterisk-users@lists.digium.com Date: Tuesday, July 1, 2008, 1:33 PM Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
Comments inline. On Tue, Jul 1, 2008 at 9:52 AM, Alexander Lopez [EMAIL PROTECTED] wrote: OK, there could be a few items here: 1 Faxes usually do not work over straight IP. I know they can and many including myself have had success the mechanics of the IP network usually won't allow it. On the same LAN it should work OK, using a dedicated NIC and a crossover cable to the HylaFax server is the best way if possible. 2 If you are using anything other than a/u law forget it. I use SLIN 3 What is the connectivity between the Asterisk boxes? If they are plugged into the same private backbone (ie Same Ethernet network) you have a fighting chance. If not, it is up to the Gods of Network congestion. True. Again, a direct connection to the fax server via crossover cable is the best method. 4 You may want to read up on the IAXModem pages, Mr, Underwood, explains the use of IAXModem and gives a few pointers on useing the IAXModem to terminal/originate calls on the same machine that has the TDM circuits and use remote serial to bring the fax data back to the hylafax machine. I think his is the root of the problem. Not understanding how IAXmodem works with HylaFax. Thanks, Steve Totaro Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gustavo A Gonzalez Sent: Tuesday, July 01, 2008 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Fax Between IAX Trunks Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem + Hylafax installed on other box. I have setup IAX trunks between this boxes, all works fine but can´t send faxes from one to other, Im trying with or without NVFaxDetect application but does not work. Is there a way to get it working?. If I connect a fax machine directly to Asterisk with Iaxmodem and Hylafax, I have no problem. But between Iax Trunks nothing happened and the fax machine registered on the first PBX give me a communication error. Thanks for any help or idea to setup and get it working. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier Are the problems in SIP-PSTN calls? SIP-SIP calls? PSTN-Local? (echo test, playback, whatever) SIP-PSTN or PSTN-SIP (what direction is the call)? 7% is something you have hope of reproducing. Unless you miss the real factor. Have you managed to reproduce it yourself? The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] User unable to use DTMFs?
On Tue, Jul 01, 2008 at 02:49:17PM +0200, Vincent wrote: On Tue, 1 Jul 2008 04:23:19 -0700 (PDT), Benjamin Jacob [EMAIL PROTECTED] wrote: Care to explain the scenario Vincent? Is it a SIP peer? what is the DTMF mode set? etc. Users call into our Asterisk voice server through a Zaptel PCI interface from regular phones, usually from a PBX (virtually all of them ISDN-based). So those phones are analog or BRI? The only files I modified are zaptel.conf, zapata.conf, and extensions.conf, which don't have anything DTMF-related, so Asterisk uses the default options. Hmmm any chance we could have a llok at them? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold realtime
Nhadie wrote: Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie Realtime music on hold does not exist in Asterisk versions prior to 1.6.0. Asterisk 1.6.0 supports realtime music on hold. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. I had the chance to notice the problem once myself but I could never again reproduce. Best regards, Loic Didelot. On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier Are the problems in SIP-PSTN calls? SIP-SIP calls? PSTN-Local? (echo test, playback, whatever) SIP-PSTN or PSTN-SIP (what direction is the call)? 7% is something you have hope of reproducing. Unless you miss the real factor. Have you managed to reproduce it yourself? The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The S word: Asterisk security
Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Try IOSTAT http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat Maybe you can correlate VM and/or emailing of VM to your IO spikes. Have you watched top and the Asterisk CLI when someone hits the panic button? Thanks, Steve T On Tue, Jul 1, 2008 at 11:17 AM, Loic Didelot [EMAIL PROTECTED] wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. I had the chance to notice the problem once myself but I could never again reproduce. Best regards, Loic Didelot. On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: Hello, one of my customers complained about bad voice quality on several calls, so I programmed a button on each phone which users can hit if they have audio drops and echo. I did this to check if there is a common recurrent problem to a given destination or just for one user etc... But till now I could not detect a pattern which could explain the problems This alert button is pressed between 7%-10% of all calls. The customer has 25 phones and around 300 calls per day. The SNOM phones are connected to Linksys switches and are totaly split from the computers network. The same goes for the asterisk box. No calls are routed trough the internet. Phone - Local Lan - Asterisk - Zaptel (Junghanns BRI card) - Carrier Are the problems in SIP-PSTN calls? SIP-SIP calls? PSTN-Local? (echo test, playback, whatever) SIP-PSTN or PSTN-SIP (what direction is the call)? 7% is something you have hope of reproducing. Unless you miss the real factor. Have you managed to reproduce it yourself? The carrier we use is known for his good quality and we never had a problem. It is the historic and most expensive carrier in Luxembourg. Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a maximum of 6 concurrent calls. Maybe someone can help me to track down the problem. What should I check, monitor test. Any ideas are welcome. -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21 and CUT function
On Tuesday 01 July 2008 09:20:52 Administrator TOOTAI wrote: does anybody know how to cut a chain using the pipe delimiter? I tried to escape it or to use $'x7c' as delimiter, no luck. 1.2 does not support escaping at all. 1.4 accepts only escapes relating to space characters (\t, \r, \n). 1.6 supports the space characters, plus hexadecimal (\xNN) and octal (\0NNN) escapes. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP-PSTN calls? I had the chance to notice the problem once myself but I could never again reproduce. So it doesn't happen with Local-PSTN calls (the type you can easily test remotely if we assume there's no voip access). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On 7/1/08, randulo [EMAIL PROTECTED] wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy Randy, I'd love to participate as long as no one minds me calling in from the beach... :) I'm interested in developing my SIP DoS script (and any similar solutions). While I'm reluctant to claim that it or anything like it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21 and CUT function
On Tuesday 01 July 2008 10:48:55 Tilghman Lesher wrote:
On Tuesday 01 July 2008 09:20:52 Administrator TOOTAI wrote:
does anybody know how to cut a chain using the pipe delimiter? I tried
to escape it or to use $'x7c' as delimiter, no luck.
1.2 does not support escaping at all. 1.4 accepts only escapes relating to
space characters (\t, \r, \n). 1.6 supports the space characters, plus
hexadecimal (\xNN) and octal (\0NNN) escapes.
Oh, there is a way to do what you want in 1.4, although it is non-obvious, due
to the insane amount of escaping that needs to be done:
exten = 8122,1,NoOp(${SET(string=one|two|three|four)})
exten = 8122,n,NoOp(${CUT(string,\\|,2)})
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/vidphone-0824ba08,
one|two|three|
four) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/vidphone-0824ba08, two) in
new
stack
Note that this insane escaping has been corrected in 1.6.0.
--
Tilghman
___
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Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
On Mon, Jun 30, 2008 at 4:31 PM, Duncan Turnbull [EMAIL PROTECTED] wrote: Specifically http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention Cheers Duncan This script look good but it doesn't work in my setup. My asterisk does not generate a messages file, neither does report SIP Login failures with souce ip in a single line. I think it is time to hack chan_sip.c :-s ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
Dan York gave a security presentation at Astricon. I've heard the recording he made of that session but it has yet to be published. He may be available, as least as a representative of VOIPSA. Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Original Message Subject: Re: [asterisk-users] The S word: Asterisk security From: Kristian Kielhofner [EMAIL PROTECTED] Date: Tue, July 01, 2008 10:56 am To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com On 7/1/08, randulo [EMAIL PROTECTED] wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy Randy, I'd love to participate as long as no one minds me calling in from the beach... :) I'm interested in developing my SIP DoS script (and any similar solutions). While I'm reluctant to claim that it or anything like it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
On Tue, Jul 01, 2008 at 11:13:27AM -0500, spectro wrote: On Mon, Jun 30, 2008 at 4:31 PM, Duncan Turnbull [EMAIL PROTECTED] wrote: Specifically http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention Cheers Duncan This script look good but it doesn't work in my setup. My asterisk does not generate a messages file, Fix your logger.conf, then. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Tue, Jul 1, 2008 at 5:56 PM, Kristian Kielhofner [EMAIL PROTECTED] wrote: I'd love to participate as long as no one minds me calling in from the beach... :) Why, do they care that I'm often naked? Hopefully, no one knew. PS, I did wear shorts for the Allison Smith sessions. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold realtime
thank you. will try to test 1.6.0. Mark Michelson wrote: Nhadie wrote: Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie Realtime music on hold does not exist in Asterisk versions prior to 1.6.0. Asterisk 1.6.0 supports realtime music on hold. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Tue, Jul 1, 2008 at 11:56 AM, Kristian Kielhofner [EMAIL PROTECTED] wrote: On 7/1/08, randulo [EMAIL PROTECTED] wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy Randy, I'd love to participate as long as no one minds me calling in from the beach... :) I'm interested in developing my SIP DoS script (and any similar solutions). While I'm reluctant to claim that it or anything like it could protect from a true DoS, it would offer some protection at the application level and that could make all the difference in some instances... As far as wider Asterisk/security issues I think J. Oquendo would be a great guest (hint, hint). -- Kristian Kielhofner NOT sent from my iPhone or Blackberry NOT sent from my iPhone or Blackberry very funny, you could add the typed with my thumbs line too. :) As far as your DoS script, do you have a general idea on how the conept would work? Would you just drop the packets from the offending IPs? For security, how about an authentication retry setting in the sip configuration? After X amounts of failed auth or registration attempts, block IP for Y amount of time. It would seem fairly easy to do using realtime with DB entries for IP blocks and expiration. Then a quick query of the same tables would allow an admin to put in permanent rules on a firewall or ACL and also contact that ISP's abuse dept. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip extension compromised, need help blocking brute force attempts
On Tue, Jul 1, 2008 at 11:19 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: Fix your logger.conf, then. -- Tzafrir Cohen What am I missing? [EMAIL PROTECTED] ~]# cat /etc/asterisk/logger.conf ; ; Logging Configuration ; ; In this file, you configure logging to files or to ; the syslog system. ; ; For each file, specify what to log. ; ; For console logging, you set options at start of ; Asterisk with -v for verbose and -d for debug ; See 'asterisk -h' for more information. ; ; Directory for log files is configures in asterisk.conf ; option astlogdir ; [logfiles] ; ; Format is filename and then levels of debugging to be included: ;debug ;notice ;warning ;error ;verbose ; ; Special filename console represents the system console ; ;debug = debug ;console = notice,warning,error ;console = notice,warning,error,debug ;messages = notice,warning,error full = notice,warning,error,debug,verbose ;syslog keyword : This special keyword logs to syslog facility ; ;syslog.local0 = notice,warning,error ; [EMAIL PROTECTED] ~]# ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Jul 1, 2008, at 11:29 AM, randulo wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy I love it. I'm celebrating the 4th with a 2000 mile motorcycle ride :) I'll do my best to make it for the conference. Fred Posner www.voiptechchat.com smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Broadvoice and Asterisk 1.6.0-beta9
I am still failing to get a Broadvoice SIP peer to work correctly with Asterisk 1.6.0-beta9. My setup works fine with Asterisk 1.2. After upgrading to 1.6, I getting a FORBIDEN error from Broadvoice when Asterisk attempts to connect. I'm wondering if anyone has a working SIP connect to Broadvoice with the latest release of Asterisk, and if they do, could they share their configuration setup with me. Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
Headset mic? Drive safe ;-) Michael Graves mgraves at mstvp.com o(713) 861-4005 c(713) 201-1262 sip:[EMAIL PROTECTED] skype mjgraves FWD 54245 Original Message Subject: Re: [asterisk-users] The S word: Asterisk security From: Fred Posner [EMAIL PROTECTED] Date: Tue, July 01, 2008 12:30 pm To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com On Jul 1, 2008, at 11:29 AM, randulo wrote: Hi all, As I mentioned briefly in the SIP takeover thread, I'd like to try to talk about security this coming Friday. I realize it is a holiday in the USA, but do geeks ever take a day off, especially security-conscious geeks? Mark Spencer once said The Bug Tracker is never on vacation!. We will try to start this subject this Friday, but I have no experience at all with this. If you know anyone who is good in this area and would like to share their expertise and talk about security in the asterisk and voip contexts, I'd like to hear from them, especially next Friday July 4th. tia, Randy I love it. I'm celebrating the 4th with a 2000 mile motorcycle ride :) I'll do my best to make it for the conference. Fred Posner www.voiptechchat.comhr___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Tue, Jul 1, 2008 at 7:30 PM, Fred Posner [EMAIL PROTECTED] wrote: a 2000 mile motorcycle ride :) I'll Where to where? What Michael said. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On 7/1/08, Steve Totaro [EMAIL PROTECTED] wrote: NOT sent from my iPhone or Blackberry very funny, you could add the typed with my thumbs line too. :) I know, although it looks like I'll be waiting in line in a couple of weeks for iPhone 2.0/3G... I just need to remember to update my prefs not to include that in the sig line! As far as your DoS script, do you have a general idea on how the conept would work? Would you just drop the packets from the offending IPs? Sort of... I would make it configurable to allow for dropping, rejecting (using ICMP unreachable for UDP, TCP RST for TCP), and logging. Although I tend to not like anything other than a silent drop for what is attack traffic. I'm not going to be the sucker that's going to saturate my upstream bandwidth by actually responding to DoS traffic. DROP by default, I say. Same thing goes for logging... Using disk I/O and space for attack traffic only makes sense if you've got a properly tuned and dimensioned configuration and/or you're running on a separate box. The hashlimit extension already provides for a /proc interface that (along with standard iptables accounting) could provide for enough information without using something like the LOG target. Ideally you could get really fancy and report the source of these attacks and block them at your upstream (via BGP blackhole or some other means). As long as we remember that any large enough, sophisticated enough DoS/DDoS WILL TAKE YOU OUT unless you have ample resources to deal with it. Even then if one of the larger botnets comes after you, good luck! ;) For security, how about an authentication retry setting in the sip configuration? After X amounts of failed auth or registration attempts, block IP for Y amount of time. It would seem fairly easy to do using realtime with DB entries for IP blocks and expiration. Then a quick query of the same tables would allow an admin to put in permanent rules on a firewall or ACL and also contact that ISP's abuse dept. My main concern with implementing these protections in Asterisk is the expense of starting the thread to deal with the (SIP) traffic in the first place. Although I'm not aware of the specifics, Asterisk reserves a bit of resources for each open SIP channel. Ideally I'd intercept attack traffic in the kernel (or better yet in the kernel on a different machine) before it ever got a chance to use any Asterisk resources in userland. Adding any realtime queries or other DB foo would only serve to amplify the effects of the attack (exceed max number of connections in MySQL, die). Of course the other benefit with a generic Linux solution is the same protections (script) would work for any other SIP application or network device. -- Kristian Kielhofner NOT sent from my iPhone or Blackberry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The S word: Asterisk security
On Tue, Jul 1, 2008 at 7:30 PM, Fred Posner [EMAIL PROTECTED] wrote: a 2000 mile motorcycle ride :) I'll Where to where? Gainesville, FL to Ann Arbor, MI to Gainesville, FL What Michael said. I had a blueant bluetooth, which was awesome on the motorcycle. Clear conversations at 70+ mph. But my buddy has failed to return it, so it's time to test the JawBone on the bike. smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR
Greetings
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the time
the choice is made to the time the caller hangup or choice another option
i.e.
exten = s,1,Answer()
exten = s,n,Background(PLEASE ENTER YOU OPTION)
exten = s,n,XXX ; //IS IT POSSIBLE TO START A STOPWATCH (COUNTER) HERE
exten = s,n,WaitExten(10)
exten = s,n,Goto(s,1)
exten = 1,1,Answer()
exten = 1,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 1,n,XXX ;(RE)START A COUNTER HERE
exten = 1,n,PLAYBACK(OPTION1 SELECTED);(RE)START A COUNTER HERE
exten = 1,n,Hangup
exten = 2,1,Answer()
exten = 2,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 2,n,XXX ;(RE)START A COUNTER HERE
exten = 2,n,PLAYBACK(OPTION2 SELECTED);(RE)START A COUNTER HERE
exten = 2,n,Hangup
i believe we can set something very powerful here
Kili
Thanks Anselm
Its true that is a lot of calls but i have a separate mysql database on
different server (HP DL580G5 with 16cores). what am currently doing is
capturing the information right after selection and insert that record
into mySql.
[macro-capture-input]
;
;
; Macro that feeds data into mysql through perl script:
; ${ARG1} - MSISDN
; ${ARG2} - src
; ${ARG3} - MainMenu Application
; ${ARG4} - Channel
; ${ARG5} - calldatetime
; ${ARG6} - Sub Menu Application
;
;
exten = s,1,System(/var/lib/asterisk/agi-bin/capture.pl ${ARG1} ${ARG2}
${ARG3} ${ARG4} ${ARG5} ${ARG6})
[Data-Services-Options]
; This menu is aimed to provide user with info about data services
offered by Vodacom, including
; 1 - SUBMenu 1
; 2 - SUBMenu 2
; 3 - SUBMenu 3
; 4 - SUBMenu 4
;
;
;SUBMENU 1
;
exten =
1,1,Macro(capture-input,${MSISDN},${OPT},APPLICATION1,${CHANNEL},now(),SUBMENU1)
exten = 1,n,Background(IVR/(1110) MENU 1)
;SUBMENU 2
exten = 2,1,Macro(sendsms,${MSISDN},1,${LANGUAGE})
exten =
2,n,Macro(ivrcdr,${MSISDN},${OPT},APPLICATION2,${CHANNEL},now(),SUBMENU2)
exten = 2,n,Background(IVR/(1120) MENU 2)
.
.
.
what i will also want to capture is how long a caller took to listen to
say SUBMENU1
It should be noted that CDR doesnot capture such detailed info (Tzafrir)
Regards
Kili
On Sat, Jun 28, 2008 at 03:37:56PM +0200, Anselm Martin Hoffmeister
wrote:
Am Samstag, den 28.06.2008, 08:15 -0500 schrieb
[EMAIL PROTECTED]:
Hi List
I have build an IVR on Asterisk from 1.2 to now 1.4.18 and has
already
processed more than 10million calls!
I have one big challenge which is reporting... it is the requirement
to
have a web reporting module which should the following info based on
selected time frame
- Number of calls on specific branch- Done
- Number of calls to branch 1 that came from branch 2 (this should
be
flexible)
- talktime on specified branch (say how long caller listened to
option
1
before choosing option 2 or hangup)
On IVR, it is so important to understand how many callers select a
specific branch and how long they spent on that branch. CDR stats can
not
provide these type of information and on trying freepbx, still can
not
go
so detailed
Dear Kili,
in my opinion this is a good application for Database backends. You
could, for example, write entries to a DB whenever someone presses a
key
(or is re-routed in the dialplan, which comes to a similar scheme). In
data mining time some SQL logic can produce nearly any data you want,
provided the input data is there.
Millions of calls sounds a lot though, so be sure to have a reasonable
database backend: The asterisk included one might be a bit on the small
side here.
This is just an idea, I did not implement anything the like (yet).
Asterisk already has this separate database backend: CDR.
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406 mailto:[EMAIL PROTECTED]
http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Call quality
Yes, most calls are SIP-PSTN calls. Thanks for your help. I will try a faster box. Are VIA CPUs known to cause problems? Loic On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP-PSTN calls? I had the chance to notice the problem once myself but I could never again reproduce. So it doesn't happen with Local-PSTN calls (the type you can easily test remotely if we assume there's no voip access). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I don't think your issue is the VIA CPU but the I/O of your flash drive. Voicemail is what I suspect being the I/O bottleneck. Thanks, Steve T On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote: Yes, most calls are SIP-PSTN calls. Thanks for your help. I will try a faster box. Are VIA CPUs known to cause problems? Loic On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP-PSTN calls? I had the chance to notice the problem once myself but I could never again reproduce. So it doesn't happen with Local-PSTN calls (the type you can easily test remotely if we assume there's no voip access). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
I am trying with ulaw/alaw codecs, both boxes are in the same LAN. The first box have Asterisk 1.2 and the other have installed Asterisk 1.4, I can do calls between boxes without problem. So the solution for this model is connect both server through a crossover cable?. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Yes, but they get like 10 voicemails per day. That feature isnt really used alot. Loic On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote: I don't think your issue is the VIA CPU but the I/O of your flash drive. Voicemail is what I suspect being the I/O bottleneck. Thanks, Steve T On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote: Yes, most calls are SIP-PSTN calls. Thanks for your help. I will try a faster box. Are VIA CPUs known to cause problems? Loic On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP-PSTN calls? I had the chance to notice the problem once myself but I could never again reproduce. So it doesn't happen with Local-PSTN calls (the type you can easily test remotely if we assume there's no voip access). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Between IAX Trunks
Crossover will certainly help with IP issues such as latency. I think your problem is that you don't really understand how IAXmodem and Hylafax work. Here is a link to a well documented Hylafax installation that I did for a fairly large mortgage company. Alex came in after me and did a great job documenting my work (and taking credit). http://blog.evaristesys.com/?p=24 Thanks, Steve Totaro On Tue, Jul 1, 2008 at 3:01 PM, Gustavo A Gonzalez [EMAIL PROTECTED] wrote: I am trying with ulaw/alaw codecs, both boxes are in the same LAN. The first box have Asterisk 1.2 and the other have installed Asterisk 1.4, I can do calls between boxes without problem. So the solution for this model is connect both server through a crossover cable?. Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue show name - callerID
Hi, Is there a way to show callerID of calls waiting in queue? queue show shows only channel not callerID Cheers, Marcin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Run top along with the tool that indicated the high I/O and see what is going on. Are you doing G729 or anything like that? Thanks, Steve T On Tue, Jul 1, 2008 at 3:06 PM, Loic Didelot [EMAIL PROTECTED] wrote: Yes, but they get like 10 voicemails per day. That feature isnt really used alot. Loic On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote: I don't think your issue is the VIA CPU but the I/O of your flash drive. Voicemail is what I suspect being the I/O bottleneck. Thanks, Steve T On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot [EMAIL PROTECTED] wrote: Yes, most calls are SIP-PSTN calls. Thanks for your help. I will try a faster box. Are VIA CPUs known to cause problems? Loic On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP-PSTN calls? I had the chance to notice the problem once myself but I could never again reproduce. So it doesn't happen with Local-PSTN calls (the type you can easily test remotely if we assume there's no voip access). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Waiting time to send the call
Dear List; Can anyone advise how to increase the waiting time to consider the number is dialed (if user pickup the analoge phone and start dialing the digits, then we need to asterisk to enough time to consider that number is done and sent it). How? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote: Run top along with the tool that indicated the high I/O and see what is going on. Are you doing G729 or anything like that? vmstat will probably provide more useful data (vmstat 1 etc. for a continous run). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
I saw that bug. Most of my files are WAV though. Would it apply to them also? Doug On Tue, 1 Jul 2008, Noah Miller wrote: Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR
On Tue, 1 Jul 2008, [EMAIL PROTECTED] wrote:
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the time
the choice is made to the time the caller hangup or choice another option
i.e.
exten = s,1,Answer()
exten = s,n,Background(PLEASE ENTER YOU OPTION)
exten = s,n,XXX ; //IS IT POSSIBLE TO START A STOPWATCH (COUNTER) HERE
exten = s,n,WaitExten(10)
exten = s,n,Goto(s,1)
exten = 1,1,Answer()
exten = 1,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 1,n,XXX ;(RE)START A COUNTER HERE
exten = 1,n,PLAYBACK(OPTION1 SELECTED);(RE)START A COUNTER HERE
exten = 1,n,Hangup
exten = 2,1,Answer()
exten = 2,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 2,n,XXX ;(RE)START A COUNTER HERE
exten = 2,n,PLAYBACK(OPTION2 SELECTED);(RE)START A COUNTER HERE
exten = 2,n,Hangup
You could start your stopwatch with
exten = s,n,set(STOPWATCH=${EPOCH})
instead of your extraneous answer()s and then in your h extension, stop
the stopwatch with:
exten = h,n,set(STOPWATCH=$[${EPOCH} -
${STOPWATCH}])
You could also use resetcdr(w) at the start of each option. This will
create a new CDR with the time spent on the previous option at each step
in your dialplan. By setting the option number in a CDR variable after
each CDR is written, the time spent in each option can be identified.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Asterisk 1.4.21.1: Bugs in IAX
Hi All; I used Asterisk 1.4.21.1 and I discovered the following bugs, I do not know if other used it and discover it: 1) In the IAX trunk, it suddenly stop working and I have to restart the machine. 2) An FXS station, suddenly loose the tone and I have to re-modprobe for zaptel driver. 3) CLI command stuck sometimes. Any advise. Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
On Tue, Jul 1, 2008 at 2:15 AM, Doug Crompton [EMAIL PROTECTED] wrote: Using 1.2.29 on Linux version 2.6.18.8-0.3 (SUSE 10.2) Intel 686 dual core. I am getting choppy audio in voicemail and general message playback. see if disabling APM in your kernel solves the issue, add apm=off to kernel boot options. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice and Asterisk 1.6.0-beta9
ill verify my asterisk verison in the morning as its 3AM here, it does work woth broadvoice, ill send you my configs if needed, i used asterisk-gui on FreeBSD to configure it had no issues with it from the gui level On Wed, Jul 2, 2008 at 12:50 AM, David Siegel [EMAIL PROTECTED] wrote: I am still failing to get a Broadvoice SIP peer to work correctly with Asterisk 1.6.0-beta9. My setup works fine with Asterisk 1.2. After upgrading to 1.6, I getting a FORBIDEN error from Broadvoice when Asterisk attempts to connect. I'm wondering if anyone has a working SIP connect to Broadvoice with the latest release of Asterisk, and if they do, could they share their configuration setup with me. Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
What did you do to setup a button for alerts? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Waiting time to send the call
On Tuesday 01 July 2008 14:27:26 bilal ghayyad wrote: Can anyone advise how to increase the waiting time to consider the number is dialed (if user pickup the analoge phone and start dialing the digits, then we need to asterisk to enough time to consider that number is done and sent it). How? An analog phone, assuming you're not set up in immediate mode, never does this. If you're talking about an Asterisk IVR, then simply set TIMEOUT(digit) to something higher than the default (5). If you're using an analog phone connected to an ATA device, then you need to check with the manufacturer of that ATA device. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choppy audio
Doug Crompton wrote: I saw that bug. Most of my files are WAV though. Would it apply to them also? Doug On Tue, 1 Jul 2008, Noah Miller wrote: Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start If it is a relatively slow box, try getting the exact sound files you will be playing back, if you have the space (make menuselect). -bk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cannot dial on E1 cards
Hello, I have a Digium TE110P connected to a GSM Gateway via E1 (Euro-IDSN). It works fine, but after an hour or so, it cannot setup a call anymore. The error it sends is: app_dial.c:1215 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown) When that happens, the only way to have it working again, is doing a restart row. Incoming calls (from the gateway to the PCI card), are always working. The problem is with outbound calls, and when some time has elapsed since the last restart of Asterisk. I am using 1.4.18, but tested with 1.2.13 also. Also tested different cards (the cards are OK, they work fine in other E1s). Also tested setting internal or loopback clock, with or without CRC. Everything seems to be configured right, I still don't know what is the origin of this erratic behavior. Would like to hear any advice or suggestion. Regards, Lars Knopf ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best Practices: Empirical measure of call latency
I would like to hear your favored method to obtain an empirical measure of latency in the media path. I'm doing several things that bring the media path through asterisk, and this would allow me to make informed decisions about (a)PSTN termination providers (b)DIDs in local and remote locations (and variance between ITSP's) (c)time to/from various cellular networks (and variance between ITSP's) Thanks! Your opinion would be greatly appreciated -Karl Fife p.s. Speaking of latency, I've noticed that some sip endpoints (i.e. Aastra 57i Wireless) add significant latency. It would be interesting to do an apples-to-apples comparison between with various fxo/dect, sip/dect, wi/sip, fxo/Spread-spectrum digital , and fxo/analog 47/900/2400mhz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Depends on the phone. On many devices you can setup buttons to call a url. Thats what I did. Loic On Tue, 2008-07-01 at 21:19 +0100, Gavin Henry wrote: What did you do to setup a button for alerts? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hi, I am using g711a everywhere. I checked on a completely idle system (no calls at all) and idle CPU is dropping from 100% to 0% more than once per minute. procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 1 0 0 891124 4644 4286800 028 4047 85757 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4042 68342 0 94 6 0 0 0 0 891124 4644 4287600 0 0 4042 72429 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4065 158878 0 100 0 0 0 0 0 891124 4644 4287600 0 0 4033 59033 0 98 2 0 0 0 0 891124 4644 4287600 0 0 4012 14464 0 96 4 0 0 0 0 891124 4652 4286800 076 4013 19727 0 37 62 1 0 0 0 891124 4652 4287600 0 0 4011 20225 0 4 96 0 0 0 0 891124 4652 4287600 0 0 4011 23901 0 20 80 0 0 1 0 891124 4652 4287600 0 4 4025 21165 0 40 55 5 0 0 0 891124 4660 4287600 032 4028 20190 0 1 95 4 0 0 0 891124 4660 4287600 0 0 4022 23295 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4111 20508 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4102 25239 0 30 70 0 0 0 0 891124 4660 4287600 0 0 4112 23148 0 0 100 0 0 0 0 891124 4668 4286800 052 4116 19031 0 0 100 0 1 0 0 891124 4668 4287600 0 0 4110 21776 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4150 20332 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4114 26285 0 0 100 0 0 0 0 891124 4668 4287600 032 4118 23029 1 0 99 0 0 0 0 891124 4668 4287600 0 0 4121 23284 0 0 100 0 0 0 0 891124 4676 4286800 060 4112 25232 0 36 64 0 0 0 0 891124 4676 4287600 0 0 4134 21583 0 99 1 0 0 0 0 891124 4676 4287600 0 0 4105 26029 0 100 0 0 0 0 0 891124 4676 4287600 076 4143 22795 0 25 75 0 0 0 0 891124 4676 4287600 0 0 4118 21418 0 0 54 46 0 0 0 891124 4676 4287600 0 0 4108 25499 0 0 100 0 0 0 0 891124 4684 4286800 052 4081 20778 0 0 100 0 0 0 0 891124 4684 4287600 0 0 4011 25463 0 13 87 0 0 0 0 891124 4684 4287600 0 0 4021 23502 0 86 14 0 0 0 0 891124 4684 4287600 0 0 4015 21693 0 1 99 0 On Tue, 2008-07-01 at 22:28 +0300, Tzafrir Cohen wrote: On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote: Run top along with the tool that indicated the high I/O and see what is going on. Are you doing G729 or anything like that? vmstat will probably provide more useful data (vmstat 1 etc. for a continous run). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hi its me again. Here is the output of zttest of a completely idle system (no calls). Acoording to some documents those values do not seem to be good. The IRQ of my zaptel card is shared with other devices. But not sure if this causes a problem. lspci -v | grep IRQ 22 -B4 00:0c.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b55b Flags: medium devsel, IRQ 22 00:0f.0 IDE interface: VIA Technologies, Inc. VIA VT6420 SATA RAID Controller (rev 80) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: VIA Technologies, Inc. VIA VT6420 SATA RAID Controller Flags: bus master, medium devsel, latency 32, IRQ 22 00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT82C586/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE Flags: bus master, medium devsel, latency 32, IRQ 22 Opened pseudo zap interface, measuring accuracy... 99.989845% 99.979881% 99.987305% 99.987297% 99.988190% 99.986824% 99.987999% 99.987701% 99.984970% 99.987892% 99.987587% 99.987595% 99.987885% 99.988968% 99.987885% 99.989449% 99.987595% 99.989250% 99.988571% 99.987106% 99.990044% 99.990921% 99.986519% 99.990822% 99.978127% 99.985054% 99.984482% 99.963478% 99.978722% 99.950005% 99.974609% 99.955170% 99.969528% 99.967972% 99.964066% 99.979797% 99.962898% 99.976852% 99.980072% 99.972946% 99.989937% 99.972359% 99.986908% 99.987694% 99.988770% 99.993660% 99.991516% 99.992577% 99.993164% 99.992470% 99.984276% 99.991600% 99.983200% 99.992279% 99.979790% 99.990036% 99.981544% 99.988770% 99.981346% 99.988182% 99.988190% 99.986717% 99.991211% 99.986618% 99.986824% 99.987991% 99.988869% 99.989265% 99.987015% 99.987396% 99.987495% 99.985657% 99.987396% 99.986229% 99.987206% 99.986908% 99.986618% 99.987411% 99.988579% 99.989059% 99.987106% 99.986336% 99.987114% 99.988190% 99.983200% 99.958191% 99.986031% 99.989357% 99.985939% 99.988678% 99.989746% 99.990341% 99.988762% 99.989159% 99.976067% 99.991798% 99.962799% 99.976173% 99.972366% 99.962898% 99.972855% 99.951462% 99.983986% 99.952049% 99.985733% 99.963776% 99.977440% 99.980186% 99.973915% 99.977333% 99.990341% 99.969032% 99.995110% 99.988770% 99.989555% 99.991211% 99.992386% 99.990929% 99.992294% 99.991119% 99.991997% 99.992088% 99.980865% 99.988670% 99.982712% 99.989059% 99.981934% 99.982903% 99.981850% 99.989845% 99.981628% 99.989258% 99.872566% 99.988678% --- Results after 134 passes --- Best: 99.995 -- Worst: 99.873 -- Average: 99.982824, Difference: 99.983075 Loic On Wed, 2008-07-02 at 01:36 +0200, Loic Didelot wrote: Hi, I am using g711a everywhere. I checked on a completely idle system (no calls at all) and idle CPU is dropping from 100% to 0% more than once per minute. procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 1 0 0 891124 4644 4286800 028 4047 85757 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4042 68342 0 94 6 0 0 0 0 891124 4644 4287600 0 0 4042 72429 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4065 158878 0 100 0 0 0 0 0 891124 4644 4287600 0 0 4033 59033 0 98 2 0 0 0 0 891124 4644 4287600 0 0 4012 14464 0 96 4 0 0 0 0 891124 4652 4286800 076 4013 19727 0 37 62 1 0 0 0 891124 4652 4287600 0 0 4011 20225 0 4 96 0 0 0 0 891124 4652 4287600 0 0 4011 23901 0 20 80 0 0 1 0 891124 4652 4287600 0 4 4025 21165 0 40 55 5 0 0 0 891124 4660 4287600 032 4028 20190 0 1 95 4 0 0 0 891124 4660 4287600 0 0 4022 23295 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4111 20508 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4102 25239 0 30 70 0 0 0 0 891124 4660 4287600 0 0 4112 23148 0 0 100 0 0 0 0 891124 4668 4286800 052 4116 19031 0 0 100 0 1 0 0 891124 4668 4287600 0 0 4110 21776 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4150 20332 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4114 26285 0 0 100 0 0 0 0 891124 4668 4287600 032 4118 23029 1 0 99 0 0 0 0 891124 4668 4287600 0 0 4121 23284 0 0 100 0 0 0 0 891124 4676 4286800 060 4112 25232 0 36 64 0 0 0 0 891124 4676 4287600 0 0 4134 21583 0 99 1 0 0 0 0 891124 4676 4287600
Re: [asterisk-users] Milliwatt-sounding tone recorded over voicemail message
Using a frequency analyzer, the tone is composed of 1Khz multiples at (1, 2, 3, and 4Khz). Any ideas? On Mon, Jun 30, 2008 at 2:46 PM, James Lamanna [EMAIL PROTECTED] wrote: Hi, A couple of our users are reporting that intermittently, their voicemails are unable to be heard because there is a milliwatt-sounding tone recorded over the top of it. Has anyone else encountered this issue? I have put a recording of the voicemail up online for people to listen to to see what I am talking about here: http://www.ugcs.caltech.edu/~jlamanna/msg.mp3 I've compressed it to MP3 for file size savings. If you would like the uncompressed .wav, please let me know. Also, please CC me directly on any replies. Thank you for any help, -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Config help with ISDN Fritzcard
Hi There,
Ive managed to get a AVM ISDN Fritzcard working with debian etch (see
capiinfo output below), and compiled chan_capi and got everything
finished (i think). So i have: Etch + Asterisk + Zaptel + ChanCapi +
Asterisk Addons + Asterisk-GUI and the chan_capi driver is loaded into
asterisk:
asterisk*CLI module show like capi
Module Description
Use Count
chan_capi.so Common ISDN API Driver (1.1.1) 0
1 modules loaded
Would someone be able to help me with the config to setup the incoming
calls from the ISDN card? I dont know where to start here.
Thanks
Simon
asterisk:~# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
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Re: [asterisk-users] Choppy audio
OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start If it is a relatively slow box, try getting the exact sound files you will be playing back, if you have the space (make menuselect). -bk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Practices: Empirical measure of call latency
On Tue, 01 Jul 2008 17:57:31 -0500, [EMAIL PROTECTED] wrote: I would like to hear your favored method to obtain an empirical measure of latency in the media path. I'm doing several things that bring the media path through asterisk, and this would allow me to make informed decisions about (a)PSTN termination providers (b)DIDs in local and remote locations (and variance between ITSP's) (c)time to/from various cellular networks (and variance between ITSP's) Thanks! Your opinion would be greatly appreciated -Karl Fife p.s. Speaking of latency, I've noticed that some sip endpoints (i.e. Aastra 57i Wireless) add significant latency. It would be interesting to do an apples-to-apples comparison between with various fxo/dect, sip/dect, wi/sip, fxo/Spread-spectrum digital , and fxo/analog 47/900/2400mhz. I had a project not long ago where I thought I was going to have to make a comparison between the latency presented by two different call paths. In the end it wasn't necessary, but it did get me thinking about what I could do, lacking for any special equipment. I had thought that I'd locate an echo test on a remote server. Free World Dialup still runs one that's accessible by both SIP and IAX2. My hosted PBX provider has one accessible via PSTN or SIP. Then I'd use a mechanical click generator (impulse) at the handset while recording the call. Then take the recording into a waveform editor software and measure the timing differences between the various paths. I can't say that this would be any kind of recommended practice, but I do think that I could get a sense of the comparative path lengths/timings. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as an IVR
Oh Edward
You are my Hero... Simple but perfect. Option II is ideal but as you know
this is Asterisk/*/everything..
Thanks to list
Kill
Can someone assist to unfold the secret on how to atleast to a count on
particular branch, say, if 2 is chosen, then we start count from the
time
the choice is made to the time the caller hangup or choice another
option
i.e.
exten = s,1,Answer()
exten = s,n,Background(PLEASE ENTER YOU OPTION)
exten = s,n,XXX ; //IS IT POSSIBLE TO START A STOPWATCH (COUNTER) HERE
exten = s,n,WaitExten(10)
exten = s,n,Goto(s,1)
exten = 1,1,Answer()
exten = 1,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 1,n,XXX ;(RE)START A COUNTER HERE
exten = 1,n,PLAYBACK(OPTION1 SELECTED);(RE)START A COUNTER HERE
exten = 1,n,Hangup
exten = 2,1,Answer()
exten = 2,n,AGI(POPULATE PREVIOUS BRANCH DATA INTO THE DB)
exten = 2,n,XXX ;(RE)START A COUNTER HERE
exten = 2,n,PLAYBACK(OPTION2 SELECTED);(RE)START A COUNTER HERE
exten = 2,n,Hangup
You could start your stopwatch with
exten = s,n,set(STOPWATCH=${EPOCH})
instead of your extraneous answer()s and then in your h extension, stop
the stopwatch with:
exten = h,n,set(STOPWATCH=$[${EPOCH} -
${STOPWATCH}])
You could also use resetcdr(w) at the start of each option. This will
create a new CDR with the time spent on the previous option at each step
in your dialplan. By setting the option number in a CDR variable after
each CDR is written, the time spent in each option can be identified.
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Choppy audio
modprobe zaptel; modprobe ztdummy That will start zaptel and ztdummy after the 'zaptel stop'. Then restart asterisk. --- On Wed, 7/2/08, Doug Crompton [EMAIL PROTECTED] wrote: From: Doug Crompton [EMAIL PROTECTED] Subject: Re: [asterisk-users] Choppy audio To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wednesday, July 2, 2008, 1:58 AM OK just to be clear on what you recommend... Stop everything, unload zaptel and zrdummy modules... then just restart asterisk? Does it start zaptel? This is NOT a slow box. P6 dual core 4 gig cache, 3800 bogomips. Doug On Tue, 1 Jul 2008, bkruse wrote: I would recommend stopping asterisk (/etc/init.d/asterisk stop) /etc/init.d/zaptel stop (unload all modules) modprobe zaptel; modprobe ztdummy (in the case that you don't have another card for a timing device) /etc/init.d/asterisk start ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.21.1: Bugs in IAX
On 1 Jul 2008, at 20:30, bilal ghayyad wrote: Hi All; I used Asterisk 1.4.21.1 and I discovered the following bugs, I do not know if other used it and discover it: 1) In the IAX trunk, it suddenly stop working and I have to restart the machine. 2) An FXS station, suddenly loose the tone and I have to re-modprobe for zaptel driver. 3) CLI command stuck sometimes. Yep, I had that too. http://bugs.digium.com/view.php?id=12925 I have not had the chance to test the fix, but the bug is marked as 'resolved' so the next release should be ok. In the meanwhile, you have 3 options: 1) Go back a version or 2 - say 1.4.19 and live with the slight security vulnerability 2) Run the current SVN 3) Run 1.4.21 - but compile it without optimization, it seems the problem was only triggered in the optimized code. Depends on your situation which is best. Tim.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
