[asterisk-users] No RTP Engine problem in 1.8.2
hi guys, i have a problem with 1.8 branch no matter which release of 1.8 i'm using. i can't make any sip calls, this is the error message i get on each call: [Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found. Do you have one loaded? [Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got SDP but have no RTP session allocated. i'm sure that the rtp engine is loaded this is the messages i get when loading rtp engine: module load res_rtp_asterisk.so Loaded res_rtp_asterisk.so == Registered RTP engine 'asterisk' == Parsing '/etc/asterisk/rtp.conf': == Found == RTP Allocating from port range 1650 - 4650 Loaded res_rtp_asterisk.so = (Asterisk RTP Stack) any advice to get rid of this problem? thanks all paradise -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] channel variables not kept
hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- 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] channel variables not kept
I'm using AGI and set AGISIGHUP=no to make it keep on running on channel hangup On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: You are using AGI or DeadAGI ? Paradise Dove wrote: hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] channel variables not kept
Thanks, It works now! but i get this warning as well: Running DeadAGI on a live channel will cause problems, please use AGI is it serious? what problems will occur!?? On Fri, Aug 8, 2008 at 11:30 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: Try DeadAGI and it should work.. Paradise Dove wrote: I'm using AGI and set AGISIGHUP=no to make it keep on running on channel hangup On Fri, Aug 8, 2008 at 10:24 PM, Ruddy Gbaguidi [EMAIL PROTECTED] wrote: You are using AGI or DeadAGI ? Paradise Dove wrote: hi, i'm using asterisk 1.4.21.2, and i use channel variables in my agi scripts. the problem is that some variables (and maybe all, not sure) like ANSWEREDTIME does not kept if the caller hangs up. my agi script continues to run after caller/callee hangup but the variables are not set properly if callers hangs up. is there anything i should to to avoid this or it's a bug. thanks, paradise dove ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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] Call Forwarding Lopp Prevention
i have two extensions which have call forwarding enabled when they are busy to forward the caller to each other. 11 ==on busy== 12 12 ==on busy== 11 when both extensions are Busy a large number of stale calls will be made in the system! how can i prevent this mess in my system? ___ -- 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] detecting voltage on fxo
hi is there any way to find out that an fxo module is connected to telco line or not? paradise dove ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_rxfax vs (iaxmodem+hylafax)
On 6/15/07, Steve Underwood [EMAIL PROTECTED] wrote: Paradise Dove wrote: can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? app_rxfax+spandsp fits inside asterisk, and sends and receives TIFF files. 3rd party bits and pieces can integrate it with an e-mail environment. iaxmodem+spandsp+hylafax works as a IAX port on a network. It uses the same spandsp engine for its front end, but hylafax for its back end. If you want to use hylafax clients, this would be the appropriate choice. Also, it does a couple of things app_rxfax won't currently do, like colour faxing. Spandsp can do T.38, but not inside Asterisk, and currently not with Hylafax. T.38 for Hylafax might be added, but integrating T.38 with a class 1 FAX modem interface for Hylafax is an awful bodge. The t38modem program from openh323 does this, and it has to do some nasty things to work. :-\ ...and which method is more reliable and is recommended? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax vs (iaxmodem+hylafax)
can anybody help me to choose the most reliable fax solution for * . after googling the net i found that there are at least two solutions for this, app_rxfax+spandsp and iaxmodem+hylafax. - what's the differences between these two? - which one's better? why? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Modems with Asterisk
does astribank from xorcom do the same for me? asterisk-astribank-Modem/Fax On 6/13/07, Doug Lytle [EMAIL PROTECTED] wrote: Jeremy Mann wrote: So you're doing PRI-Channel bank? Yes, for inbound: PRI-Asterisk-Chanel Bank-Modem/Fax/Cheapy Phone For outbound: Modem/Fax/Cheapy Phone-Chanel Bank-Asterisk-PRI Before we moved to a PRI, it was: Phone Lines--Chanel Bank- Tellabs Echo canceller-Asterisk--Modem/Fax/Cheapy Phone Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Faxing
so how to avoid CPC?? On 6/14/07, C F [EMAIL PROTECTED] wrote: Its called CPC On 6/12/07, Kyle Vorster [EMAIL PROTECTED] wrote: Hello, Sorry if this is a real dumb question but when sending a fax and the end user does not enable fax on their side and then just hangs up does not force asterisk to end the call. So it keeps the trunk open until its killed by a Flash Operator. Please assist if any one understands me. Kind Regards, Kyle Virster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection problem on wctdm24xxp
hi all, i have problem with dtmf detection on wctdm24xxp with full fxo and vpm module. after pushing dtmf tones on my phone for several times the card just detects one or two digits randomly.so now i can't use any voice menu on my box with this card. i have tried the following scenarios: - the card with / without vpm module has the same dtmf detection problem. - relaxdtmf=yes/no didn't solve the problem - toneduration=300 / 350 / 400 didn't help also. - vpmdtmfsupport=1 / 0 didn't solve again. what else could be the possible cause for this problem? please help! - paradise dove ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
On 5/8/07, Kevin Collins [EMAIL PROTECTED] wrote: I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf being selected. And when reading rtp if 'f' character shows up vector to fax extension can i have your patched chan_sip.c ? Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, May 08, 2007 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Answer() command?
hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? The noise may indicate other problems. Yuan Liu Paradise Dove wrote: hi, is there anyway to Answer() the caller channel after the called number pickedup the phone. when an outside caller calls * system just continue ringing and not pick up the line and just dial an extension and then answer the caller channel after the called extension picked up the phone. is this possible in *? something like this: [incoming] exten = s,1,NoOp() exten = s,n,Dial(SIP/120) i've done this but when 120 extension picks up the phone just a noise will be heard and call won't be bridged to caller channel. i have also used Dial(SIP/120|M(answerme)) which runs a macro with Answer() command when called party picksup but it just re-answers the called extension!! thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? Don't use callprogress. It doesn't work. GOOD NEWS: Problem Fixed! i wrote a patch for dsp.c and chan_zap.c now both callprogress and answer problem work fine together. i also add a config option in zapata.conf to tune callprogress now it works with over 95 percent accuracy. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Answer() command?
On 2/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Feb 22, 2007 at 09:40:54PM +0330, Paradise Dove wrote: On 2/22/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Paradise Dove wrote: On 2/22/07, Yuan LIU [EMAIL PROTECTED] wrote: From: Pavel Jezek [EMAIL PROTECTED] Date: Thu, 22 Feb 2007 09:39:22 +0100 I think, this can be solved using phone autoanswer feature, look at wiki... exten = s,1,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten = s,2,Dial(SIP/myphone) Or without. One of my contexts is set up exactly like the original sample. Just Dial(), no Answer(). (I think I've seen textbook samples like that, too.) Asterisk bridges the call when the callee picks up. (That's the main work Asterisk does: bridging calls.) BUT, when callprogress=yes, asterisk doesn't bridge the call and just ring for the caller and noise for called!! is it a bug or it's normal? Don't use callprogress. It doesn't work. GOOD NEWS: Problem Fixed! i wrote a patch for dsp.c and chan_zap.c now both callprogress and answer problem work fine together. i also add a config option in zapata.conf to tune callprogress now it works with over 95 percent accuracy. Great! Mind posting your patch on http://bugs.digium.com ? my patch needs some tailoring its not very clean now! i will post it soon. ;-) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird problem in wctdm24xxp driver
Hi, I'm running FC3 with kernel 2.6.11. All the binary files and zaptel kernel modules is not available to system at boot time. They are extracted in a ram disk at system startup and then zaptel modules are loaded manually and so on. I have no problem with this boot routine and i've been tested all digium cards (expect tdm24) and they work fine till now. The problem appeared when i purchased the new TDM2400 card. after installing this new card system stopped on modprobe wctdm24xxp ot boot time! (randomly sleeping on Resetting the modules.. / During Resetting the modules... and sometimes After resetting the modules...) but no step further. Nothing happened till i killed modprobe manually so no modules detected on tdm24. (Port 1: Not Installed, ) My motherboard is a brand new Intel 3010 chipset from supermicro with 3.3Vpci slots. i also changed the motherboard to Intel 7230 chipset with 5V slot but nothing changed. I also switched back to zaptel-1.2.13 and removed the modules from the board and re-inserted them as digium support recommended, but nothing changed. I also changed to 2.6.12 kernel but still the same problem. Finally i changed the system startup routine and copied all extracted zaptel files to hard-disk into the standard location in kernel dirs and found that now system starts-up with no problem and detects TDM24 at boot time!!! it seemed that the problem is that the wctdm24xxp needs to be detected at boot time by the kernel. But now the problem is that when i rmmod the wctdm24xxp module and modprobe it again still system doesn't detect it and sleeps on modprobe utill i kill it. i dont have such a problem with all other cards from digium. i think this is a weird problem with wctdm24xxp driver. thanks, p. dove ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird problem in wctdm24xxp driver
On 2/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Feb 17, 2007 at 07:52:25PM +0330, Paradise Dove wrote: Hi, I'm running FC3 with kernel 2.6.11. All the binary files and zaptel kernel modules is not available to system at boot time. They are extracted in a ram disk at system startup and then zaptel modules are loaded manually and so on. I have no problem with this boot routine and i've been tested all digium cards (expect tdm24) and they work fine till now. The problem appeared when i purchased the new TDM2400 card. after installing this new card system stopped on modprobe wctdm24xxp ot boot time! (randomly sleeping on Resetting the modules.. / During Resetting the modules... and sometimes After resetting the modules...) but no step further. Nothing happened till i killed modprobe manually so no modules detected on tdm24. (Port 1: Not Installed, ) My motherboard is a brand new Intel 3010 chipset from supermicro with 3.3Vpci slots. i also changed the motherboard to Intel 7230 chipset with 5V slot but nothing changed. I also switched back to zaptel-1.2.13 and removed the modules from the board and re-inserted them as digium support recommended, but nothing changed. I also changed to 2.6.12 kernel but still the same problem. Finally i changed the system startup routine and copied all extracted zaptel files to hard-disk into the standard location in kernel dirs and found that now system starts-up with no problem and detects TDM24 at boot time!!! Hmmm... How exactly do you load the modules? With insmod or modprobe? when i put zaptel modules in kernel dirs. it detects all the needed modules. and from the dmesg it seems (as it should be) to load zaptel first and then wctdm24xxp. so i don't need to do a insmod or modprobe at all. the problem comes when i rmmod these modules and modprobe or insmod them again. it seemed that the problem is that the wctdm24xxp needs to be detected at boot time by the kernel. But now the problem is that when i rmmod the wctdm24xxp module and modprobe it again still system doesn't detect it and sleeps on modprobe utill i kill it. i dont have such a problem with all other cards from digium. i think this is a weird problem with wctdm24xxp driver. Another theory: only the first modprobe after a boot is successful, and the modules are loaded automatically at boot. no it's not true that the first modprobe is the successful one. it seems the card works when kernel detects it and loads the modules itself. something that happens before init scripts. Test: remove the modules copletely, reboot, re-add the modules and modprobe again. you mean remove the physical modules? if yes i've done it once. made no sense -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird problem in wctdm24xxp driver
BAD News! the card doesn't seem to work at all. even when it's detected by kernel it doesn't send/recv any interrupts in system. /proc/interrupts shows this. i also send p to sysrq but nothing special was shown: Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r1686M Echo Canceller: KB1 ACPI: PCI interrupt :03:01.0[A] - GSI 48 (level, low) - IRQ 82 PCI Config reg is 02900117 WCTDM2400P: New Reg: fe59! Detected REG0: 0100 Detected REG1: 7849 Detected REG2: 001d (pre) Reg fc is 5027 (post) Reg fc is 5024 Detected REG2: wctdm2400p: reg is a04c0004 Resetting the modules... On 2/18/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Feb 17, 2007 at 11:52:32PM +0330, Paradise Dove wrote: On 2/17/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sat, Feb 17, 2007 at 07:52:25PM +0330, Paradise Dove wrote: Hi, I'm running FC3 with kernel 2.6.11. All the binary files and zaptel kernel modules is not available to system at boot time. They are extracted in a ram disk at system startup and then zaptel modules are loaded manually and so on. I have no problem with this boot routine and i've been tested all digium cards (expect tdm24) and they work fine till now. The problem appeared when i purchased the new TDM2400 card. after installing this new card system stopped on modprobe wctdm24xxp ot boot time! (randomly sleeping on Resetting the modules.. / During Resetting the modules... and sometimes After resetting the modules...) but no step further. Nothing happened till i killed modprobe manually so no modules detected on tdm24. (Port 1: Not Installed, ) Strange. You can kill the modprobe process? In't probing done in kernel with the process in state D? To see what the kernel is doing: alt-sysrq-p (or: echo p /proc/sysrq-trigger ) My motherboard is a brand new Intel 3010 chipset from supermicro with 3.3Vpci slots. i also changed the motherboard to Intel 7230 chipset with 5V slot but nothing changed. I also switched back to zaptel-1.2.13 and removed the modules from the board and re-inserted them as digium support recommended, but nothing changed. I also changed to 2.6.12 kernel but still the same problem. Finally i changed the system startup routine and copied all extracted zaptel files to hard-disk into the standard location in kernel dirs and found that now system starts-up with no problem and detects TDM24 at boot time!!! Hmmm... How exactly do you load the modules? With insmod or modprobe? when i put zaptel modules in kernel dirs. it detects all the needed modules. and from the dmesg it seems (as it should be) to load zaptel first and then wctdm24xxp. so i don't need to do a insmod or modprobe at all. the problem comes when i rmmod these modules and modprobe or insmod them again. This is run somewhere in rc.sysinit . Specifically, where it loads modules of other devices. it seemed that the problem is that the wctdm24xxp needs to be detected at boot time by the kernel. But now the problem is that when i rmmod the wctdm24xxp module and modprobe it again still system doesn't detect it and sleeps on modprobe utill i kill it. i dont have such a problem with all other cards from digium. i think this is a weird problem with wctdm24xxp driver. Another theory: only the first modprobe after a boot is successful, and the modules are loaded automatically at boot. no it's not true that the first modprobe is the successful one. it seems the card works when kernel detects it and loads the modules itself. something that happens before init scripts. Test: remove the modules copletely, reboot, re-add the modules and modprobe again. you mean remove the physical modules? if yes i've done it once. made no sense Yes. A long-shot, but easy to accomploish (if you can afford a reboot). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM2400 and 3.3v pci
does TDM2400 work on 3.3v pci slot? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM2400 and 3.3v pci
my card has just fxo modules and is put in a 3.3v slot. when running modprobe wctdm24xxp it waits for ever and dmesg shows Resetting the modules what could be the problem? when i put this card in another system with 5v slot it works fine. On 2/12/07, William Moore [EMAIL PROTECTED] wrote: On 2/11/07, Paradise Dove [EMAIL PROTECTED] wrote: does TDM2400 work on 3.3v pci slot? Yes, all of Digium's analog cards are dual voltage and can work with either 3.3V or 5V slots. You just need to make sure you have an extra molex connector if you're going to be using FXS modules on the card. William ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with installing tdm2400
i have a full fxo TDM24 and i have problem with installing it. when i run modprobe wctdm24xxp dmesg shows the following messages. and it waits for ever and nothing will happen. i'm sure that: - the power is plugged into tdm24 board - udev is configured and is working with other tdm cards. - zaptel.conf is configured. Zapata Telephony Interface Registered on major 196 Zaptel Version: SVN-branch-1.2-r2128M Zaptel Echo Canceller: KB1 ACPI: PCI interrupt :02:01.0[A] - GSI 24 (level, low) - IRQ 74 PCI Config reg is 02900317 Wildcard TDM2400P: New Reg: fe59! Detected REG0: 0100 Detected REG1: 7849 Detected REG2: 001d (pre) Reg fc is 1027 (post) Reg fc is 1024 Detected REG2: wctdm24xxp: reg is a04c0004 Resetting the modules... During Resetting the modules... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] rx_fax problem
hi, rx_fax fails to get fax on a bit noisy lines but real fax devices can do that on the same line with no problem! what's the problem? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error in starting * with latest trunk
hi, i've just upgraded to latest trunk. everything compiles fine but when starting this message appears and fails to start. WARNING[3990] loader.c: module chan_zap.so error /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call thanks, paradise dove ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp 0.0.2pre25
pre25 is working fine for me. On 2/19/06, Jesse Guardiani [EMAIL PROTECTED] wrote: Hello, Is anyone successfully using spandsp 0.0.2pre25 with either asterisk 1.0.x or 1.2.4? I've built a Gentoo ebuild for this version of spandsp and app_rtxfax, and it builds, but I'm not having any luck getting it working. 99% of my test faxes fail. Reverting to 0.0.2pre20 yields a much higher success rate. I've bumped the console debugging level in logger.conf to include debug and verbose, as well as the defaults. I'm running libtiff-3.7.1, because Mr. Underwood's site recommends it, even though it's a vulnerable version of libtiff. I'm faxing from a working, production asterisk 1.0.9 + spandsp 0.0.2pre20 system to a testing asterisk 1.0.10 + spandsp 0.0.2pre25 system, and 1 of 3 things usually happens: 1.) The fax goes through (very rare in testing) 2.) The fax loops indefinitely like this: Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:04 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:07 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4059 zt_read: DTMF digit: f on Zap/1-1 Feb 19 11:46:07 DEBUG[5089]: chan_zap.c:4101 zt_read: Fax already handled Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier up Feb 19 11:46:10 DEBUG[5089]: app_rxfax.c:70 span_message: FLOW HDLC carrier down
Re: [Asterisk-Users] jitterbuffer causes no sound?
this is a time issue. change your date to older value. everything works again. paradise dove On 1/25/06, stevanus [EMAIL PROTECTED] wrote: Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having the same symptoms (i.e: no voice). There is no sound when the call is pickup, no matter the call is from sip to sip, sip to zap, zap to sip ,sip to zap through iax, nor sip to sip through iax... Is jitterbuffer really the culprit or it's just a coincidence that I activated the jitterbuffer and my asterisks stopped working? Is asterisk 1.2.2 not meant for production use? Has there someone success story implemented asterisk 1.2.2? If there's, please share me as it can encouraged me to try this beast again :)... Currently, I'm rollback to asterisk 1.0.10 to avoid any unprecedented issue... Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suddenly No audio
yes, it's a bad pain. btw, i've submitted a bug. On 1/25/06, Peter Fern [EMAIL PROTECTED] wrote: Yep, I just got stung by this too - an hour of extreme pain, multiple * boxen all failed at precisely the same moment, and they're in different timezones, so must be a calc on epoch or UTC. Anyone shed any light on this? I'm hacking our CDRs currently to work around the difference in year, but I've obviously also had to disable ntp and I hate to think what setting the date by hand will have done to our CDR collation between machines... Paradise Dove wrote: this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349 change your system date to an older value. everything will work again. paradise dove On 1/25/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Hi, I set up a small system over the last couple of days and all was fine. (* 1.2.2 - Fedora Core 1 System has 2xTDM400P with 7xFXO and 1xFXS, a couple of SIP phones (snom320 and IP300), fax machine on the FXS channel and an IAX2 trunk through a local provider. All worked fine until this morning suddenly no audio, neither on internal or external calls. Even calls between two zap channels have no audio. The call is connected OK but there is just no sound. Frankly I do not even know where to begin looking because it is not a SIP problem (ZAP channels do the same). Any suggestions will be appreciated. Tx M -- Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suddenly No audio
this is a new bug which is submitted: http://bugs.digium.com/view.php?id=6349 change your system date to an older value. everything will work again. paradise dove On 1/25/06, Marnus van Niekerk [EMAIL PROTECTED] wrote: Hi, I set up a small system over the last couple of days and all was fine. (* 1.2.2 - Fedora Core 1 System has 2xTDM400P with 7xFXO and 1xFXS, a couple of SIP phones (snom320 and IP300), fax machine on the FXS channel and an IAX2 trunk through a local provider. All worked fine until this morning suddenly no audio, neither on internal or external calls. Even calls between two zap channels have no audio. The call is connected OK but there is just no sound. Frankly I do not even know where to begin looking because it is not a SIP problem (ZAP channels do the same). Any suggestions will be appreciated. Tx M -- Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp-0.0.2pre22 not working!
DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:51 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:52 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:53 DEBUG[5157] app_rxfax.c: Got hangup thanks, paradise dove ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call-limit kills hints
i have the same problem and also have submitted it as bug http://bugs.digium.com/view.php?id=5281. the Patch-5281-v2.txt in the mentioned bug will solve your problem. Paradise Dove On 1/3/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Joseph Rothstein wrote: I am setting up 10 SNOM 320s for a customer, and there seems to be a problem with call-limit and hints. Here is my sip config for one phone: [944] type=friend context=x language=de accountcode=x notifyringing=yes host=dynamic dtmfmode=rfc2833 [EMAIL PROTECTED] callerid=x 944 canreinvite=no disallow=all allow=g729 nat=yes If I add to this, call-limit=1, hint does not work at all. I get no status change from the hinted devices/extensions. I believe the incominglimit outgoinglimit and limit options will be removed in the next version of Asterisk. They were replaced by the *Group applications in 1.0 and by the *GROUP functions in 1.2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regular Crashes
i have the same problem. but when i remove all hints from my dialplan
in extensions.conf.
on more crash will occur.
Paradise Dove
On 1/2/06, Andrew Gough [EMAIL PROTECTED] wrote:
I don't think this is the same problem I am experiencing. As you can see
below the two BT's are almost identical and I have others the same too. so
the fault is fairly consistent, unfortunately I have been unable to determine
the exact reason for it yet. It is not the whole box crashing it is merely
Asterisk core dumps. sometimes in the middle of a call and sometimes when
there is no-one even in the office. Unless I get solution soon I'll be
forced to give up on asterisk, which would be a real shame.
Regards
Andrew
From: [EMAIL PROTECTED] on behalf of Zafer Khodr
Sent: Fri 30/12/2005 15:32
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Regular Crashes
I have been experiencing a similar problem.
I have not yet been able to figure out what the exact problem is but I know
that the errors are inconsitant.
Sometimes nothing for 2 days and sometimes 5 times a day.
I thought about it a lot and I have found only one thing in common.
The area where my server is stored gets pretty stuffy, especially on a hot
day.
I occasionally turn on the aircon as I need to go in and do some work.
From my best recollection the server has never crashed when the aircon has
been on.
This is my third day of testing my theory, and with the aircon controlling
the room tempreture to make sure it is always nice and cool in there I have
not seen any errors for 3 days (Keeping in mind that the day I decided to try
this theory by constantly keeping the room cool my server encountered around
4 errors in just a few hours).
So to put in short I think but cant be sure that somehow when the room gets
too hot the server goes awol and somehow causes this error.
Don't ask me how or why... all I know is that now with controlled room temp I
have not had a problem.
Good Luck
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Gough
Sent: Saturday, 31 December 2005 1:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Regular Crashes
I have just setup asterisk on a debian sarge box. I am running Asterisk
1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz)
ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions
(SIP) configured all using CounterPath(Xten) eyebeam softphone.
After many hours of Googling I have finally got it all setup and
working. We can transfer calls internally and make and receive external
calls. Its all great except for stability issues!!
Essentially every now and again, asterisk simply dies (2-3 times a
day). No warning, no error, just my console session outputs a
disconnected from console message.
Sometimes the crashes happen when you are on a call, other times when
there is no-one in the office.
The server is a brand new AMD 3400+ with 512Mb RAM. The other issue
experienced is occasional break up on inbound sound quality.
Below are traces of the last two crashes
Any Help much appreciated
Regards
Andrew Gough
FIRST TRACE
#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x672e33fc) at lock.h:597
No locals.
#2 0x0806175a in ast_queue_hangup (chan=0x672e3330) at channel.c:671
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0, data = 0x0, delivery = {tv_sec =
0,
tv_usec = 0}, prev = 0x0, next = 0x0}
#3 0x408fc2d9 in __sip_autodestruct (data=0x81be208) at chan_sip.c:1315
p = (struct sip_pvt *) 0x81be208
#4 0x08056c3e in ast_sched_runq (con=0x8172f28) at sched.c:373
current = (struct sched *) 0x8174868
tv = {tv_sec = 1135275568, tv_usec = 989877}
x = 0
res = 1083432672
#5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253
res = 0
sip = (struct sip_pvt *) 0x0
peer = (struct sip_peer *) 0x0
t = 1135275568
fastrestart = 0
lastpeernum = -1
curpeernum = 6
reloading = 0
#6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0
No symbol table info available.
#7 0x401ac18a in clone () from /lib/tls/libc.so.6
No symbol table info available.
SECOND TRACE
#0 0x400268b7 in pthread_mutex_trylock () from /lib/tls/libpthread.so.0
No symbol table info available.
#1 0x0806c146 in ast_mutex_trylock (pmutex=0x120010c) at lock.h:597
No locals.
#2 0x0806175a in ast_queue_hangup (chan=0x1200040) at channel.c:671
f = {frametype = 4, subclass = 1, datalen = 0, samples = 0,
mallocd = 0, offset = 0, src = 0x0,
data = 0x0, delivery = {tv_sec = 0, tv_usec = 0
[Asterisk-Users] GROUP_COUNT and AGI
hi, is it possible to use GROUP_COUNT function in AGIs. i could not make it work. :-( thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
i'm using 1.2.
get the right patch from http://bugs.digium.com/view.php?id=5281
patch fie is: Patch-5281-v2.txt
On 12/6/05, Alvaro Parres [EMAIL PROTECTED] wrote:
which version of Asterisk do you have ?, Becouse when i change the function
to your code, every time that one phone with call-limit the Asterisk crash.
I have 1.2.0
On 12/3/05, Paradise Dove [EMAIL PROTECTED] wrote:
hi,
This is the new update_call_counter() which works fine for me:
/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount
* Thought: For realtime, we should propably update storage with inuse
counter... */
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
int *inuse, *call_limit;
int outgoing = ast_test_flag(fup, SIP_OUTGOING);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
if (option_debug 2)
ast_log(LOG_DEBUG, Updating call counter for %s call\n,
outgoing ? outgoing : incoming);
/* Test if we need to check call limits, in order to avoid
realtime lookups if we do not need it */
if (!ast_test_flag(fup, SIP_CALL_LIMIT))
return 0;
ast_copy_string(name, fup-username, sizeof(name));
/* Check the list of users */
// paradise dove
p = find_peer(name, NULL, 1);
if (p) {
inuse = p-inUse;
call_limit = p-call_limit;
} else if (!u) {
/* Try to find user */
u = find_user(name, 1);
if (u) {
inuse = u-inUse;
call_limit = u-call_limit;
} else {
if (option_debug 1)
ast_log(LOG_DEBUG, %s is not a local user, no call
limit\n, name);
return 0;
}
}
switch(event) {
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
if ( *inuse 0 ) {
(*inuse)--;
} else {
*inuse = 0;
}
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' removed from call
limit %d\n, outgoing ? to : from, u ? user:peer
}
break;
case INC_CALL_LIMIT:
if (*call_limit 0 ) {
if (*inuse = *call_limit) {
ast_log(LOG_ERROR, Call %s %s '%s' rejected due
to usage limit of %d\n, outgoing ? to : from, u ? u
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return -1;
}
}
(*inuse)++;
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' is %d out of
%d\n, outgoing ? to : from, u ? user:peer, name, *in
}
break;
default:
ast_log(LOG_ERROR, update_call_counter(%s, %d) called
with no event!\n, name, event);
}
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return 0;
}
Paradise Dove
On 12/2/05, Alvaro Parres [EMAIL PROTECTED] wrote:
Could you send it patch please.
On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote:
btw, i've patched this part of code and now its working fine for me.
i'm going to upload it.
Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the
same.
i agree with you, it seems to be a bug which i've submited before
(bug
#5281) but it's now closed by bug marshals!
It's not closed. It's suspended waiting input from you:
Closing until the appropriate debug/trace output can be provided.
On 10/30 you said you were still trying to get the debug output.
Cheers,
Kevin
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
hi,
This is the new update_call_counter() which works fine for me:
/*! \brief update_call_counter: Handle call_limit for SIP users
* Note: This is going to be replaced by app_groupcount
* Thought: For realtime, we should propably update storage with inuse
counter... */
static int update_call_counter(struct sip_pvt *fup, int event)
{
char name[256];
int *inuse, *call_limit;
int outgoing = ast_test_flag(fup, SIP_OUTGOING);
struct sip_user *u = NULL;
struct sip_peer *p = NULL;
if (option_debug 2)
ast_log(LOG_DEBUG, Updating call counter for %s call\n,
outgoing ? outgoing : incoming);
/* Test if we need to check call limits, in order to avoid
realtime lookups if we do not need it */
if (!ast_test_flag(fup, SIP_CALL_LIMIT))
return 0;
ast_copy_string(name, fup-username, sizeof(name));
/* Check the list of users */
// paradise dove
p = find_peer(name, NULL, 1);
if (p) {
inuse = p-inUse;
call_limit = p-call_limit;
} else if (!u) {
/* Try to find user */
u = find_user(name, 1);
if (u) {
inuse = u-inUse;
call_limit = u-call_limit;
} else {
if (option_debug 1)
ast_log(LOG_DEBUG, %s is not a local user, no call
limit\n, name);
return 0;
}
}
switch(event) {
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
if ( *inuse 0 ) {
(*inuse)--;
} else {
*inuse = 0;
}
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' removed from call
limit %d\n, outgoing ? to : from, u ? user:peer
}
break;
case INC_CALL_LIMIT:
if (*call_limit 0 ) {
if (*inuse = *call_limit) {
ast_log(LOG_ERROR, Call %s %s '%s' rejected due
to usage limit of %d\n, outgoing ? to : from, u ? u
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return -1;
}
}
(*inuse)++;
if (option_debug 1 || sipdebug) {
ast_log(LOG_DEBUG, Call %s %s '%s' is %d out of
%d\n, outgoing ? to : from, u ? user:peer, name, *in
}
break;
default:
ast_log(LOG_ERROR, update_call_counter(%s, %d) called
with no event!\n, name, event);
}
// paradise dove
if (p)
ASTOBJ_UNREF(p,sip_destroy_peer);
else if (u)
ASTOBJ_UNREF(u,sip_destroy_user);
return 0;
}
Paradise Dove
On 12/2/05, Alvaro Parres [EMAIL PROTECTED] wrote:
Could you send it patch please.
On 11/30/05, Paradise Dove [EMAIL PROTECTED] wrote:
btw, i've patched this part of code and now its working fine for me.
i'm going to upload it.
Paradise Dove
On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Paradise Dove wrote:
Yes with version 1.2. I have tried already with call-limit and the
same.
i agree with you, it seems to be a bug which i've submited before (bug
#5281) but it's now closed by bug marshals!
It's not closed. It's suspended waiting input from you:
Closing until the appropriate debug/trace output can be provided.
On 10/30 you said you were still trying to get the debug output.
Cheers,
Kevin
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
Yes with version 1.2. I have tried already with call-limit and the same.
i agree with you, it seems to be a bug which i've submited before (bug
#5281) but it's now closed by bug marshals!
by reading chan_sip.c u will find out that in function
update_call_counter() it first tries to update the call counter
(call-limit) prop. of user strcuture of the extension.
u = find_user(name, 1);
if (u) {
inuse = u-inUse;
call_limit = u-call_limit;
p = NULL;
} else {
/* Try to find peer */
if (!p)
p = find_peer(fup-peername, NULL, 1);
...
and then in function sip_devicestate() it changes the state of the
extension according to peer (call-limit) prop. of the extension.
if ((p = find_peer(host, NULL, 1))) {
if (p-addr.sin_addr.s_addr || p-defaddr.sin_addr.s_addr) {
/* we have an address for the peer */
/* if qualify is turned on, check the status */
if (p-maxms (p-lastms p-maxms)) {
res = AST_DEVICE_UNAVAILABLE;
} else {
/* qualify is not on, or the peer is responding
properly */
/* check call limit */
if (p-call_limit (p-inUse ==
p-call_limit))
res = AST_DEVICE_BUSY;
else if (p-call_limit p-inUse)
res = AST_DEVICE_INUSE;
else if (p-call_limit)
res = AST_DEVICE_NOT_INUSE;
else
res = AST_DEVICE_UNKNOWN;
}
So, if your sip user is defined as friend it will be detected as user
in function update_call_counter and also detected as peer in function
sip_devicestate which doesn't sense the call-limit of friends. if
change the type of your sip user to peer you will see that hints
works fine.
another way to find this bug is to run the command sip show inuse on
CLI when some sip extensions are in a call. you will see that just the
user counter of sip friends are updated.
Paradise Dove
On 11/29/05, Alvaro Parres [EMAIL PROTECTED] wrote:
On 11/28/05, Kevin Hanson [EMAIL PROTECTED] wrote:
Alvaro Parres wrote:
Hi list...
I have been testing the hint extension. And i detect
that when i have in the sip.fg of the extension the
incominiglimit=X (any number) the hint doesn't work all the
time show the extesion as idle.
If this is a bug or not ??
Thanks.
___
What version of Asterisk? 1.2 deprecated incominglimit in favor of
call-limit.
Cheers,
Kevin
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Re: [Asterisk-Users] Is a BUG ? Hints and incominglimit
btw, i've patched this part of code and now its working fine for me. i'm going to upload it. Paradise Dove On 11/30/05, Kevin Hanson [EMAIL PROTECTED] wrote: Paradise Dove wrote: Yes with version 1.2. I have tried already with call-limit and the same. i agree with you, it seems to be a bug which i've submited before (bug #5281) but it's now closed by bug marshals! It's not closed. It's suspended waiting input from you: Closing until the appropriate debug/trace output can be provided. On 10/30 you said you were still trying to get the debug output. Cheers, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP! on disconnecting stale calls.
hi, how can i hangup such calls without restarting asterisk? the Zap channel on this case is busy for more than 7 hours some logs are followed. thanks, Paradise Dove - Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25788 seconds Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25788 seconds Nov 23 16:59:50 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25789 seconds Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25790 seconds Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25790 seconds Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25791 seconds Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25791 seconds Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25792 seconds Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25792 seconds - Channel Location State Application(Data) Zap/15-1 [EMAIL PROTECTED]:1 Up Bridged Call(SIP/2378-740f) 1 active channel 1 active call ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP! on disconnecting stale calls.
as i said before, i've ran soft hangup on both sip and zap channels on this call several times but no success. by exploring the code in chan_sip.c it shows that * also attempts to run softhangup on this call. is this probably be a bug? thanks, paradise dove On 11/25/05, tracinet [EMAIL PROTECTED] wrote: Have you tried the soft hangup command? On 11/24/05, Paradise Dove [EMAIL PROTECTED] wrote: hi, how can i hangup such calls without restarting asterisk? the Zap channel on this case is busy for more than 7 hours some logs are followed. thanks, Paradise Dove - Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25788 seconds Nov 23 16:59:49 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25788 seconds Nov 23 16:59:50 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25789 seconds Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25790 seconds Nov 23 16:59:51 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25790 seconds Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25791 seconds Nov 23 16:59:52 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25791 seconds Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25792 seconds Nov 23 16:59:53 NOTICE[3752] chan_sip.c: Disconnecting call 'SIP/2378-740f' for lack of RTP activity in 25792 seconds - Channel Location State Application(Data) Zap/15-1 [EMAIL PROTECTED]:1 Up Bridged Call(SIP/2378-740f) 1 active channel 1 active call ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] faxdetect on voicemail
hi, is there anyway to just enable faxdetection in voicemail? thanks, paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CallerID detection problem
hi, is there anyway to make * to detect callerid before first ring. i know that it seems silly; but here i have a case that Telco sends the caller-id before first ring. this issue is detected by installing a callerid detection device on the line. it shows callerid just before the first ring. so * can't detect the callerid. thanks, paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 1.2.0-beta1 and Hints (Re: CVS HEAD and Hints)
me too! i had hints working for months before upgrading to CVS HEAD. i've also submitted a bugs: http://bugs.digium.com/view.php?id=5281 my question is that is there anybody who is using CVS HEAD and hints works for him? btw, thanks, Paradise Dove On 10/7/05, Stefan Tichy [EMAIL PROTECTED] wrote: On Thu, Oct 06, 2005 at 10:35:46PM +0200, Olle E. Johansson wrote: incominglimit is replaced by call-limit. Please read sip.conf.sample. Outgoinglimit has not worked for ages, so we removed it. One limit works for both incoming and outgoing calls now. sip.conf.sample available in 1.2.0-beta1 lists incominglimit and outgoinglimit, but it is different in the current CVS head. I appreciate that hint, but it does not help me in getting dialplan hint working. -- Stefan Tichy [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS HEAD and Hints
Hi, i was just wondering that is there anybody who has any success with hints on CVS HEAD? a sample configuration of sip.conf and extensions.conf is pleased. Paradise Dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error on loading zaptel module
i get this error on dmesg: zaptel: Unknown symbol __stack_smash_handler zaptel: Unknown symbol __guard paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: E1 ChannelBank and UniCall
has anybody succeeded in connecting an E1 CB to asterisk using R2 Digital signalling and Unicall? any help will be appreciated, Paradise Dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: Valiant E1 CB and UniCall
Is there any success in connecting Valiant E1 CB with Unicall to asterisk? any help will be appreciated, Paradise Dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hints not working on CVS HEAD
i've tried it on both snom190 and eyeBeam none of them work. nothing is changed in configs. is there any success in making snom LEDs work on CVS HEAD? thanks, paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Return
does * support call return? i want when the operator transfers a call if the transferee is busy or doesn't answer the call the call return back to operator again... this feature may be called: call return on busy call return on no answer Paradise Dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to disconnect a bridged channel
Hi, i've just faced with some bridged calls which could not be hungup just killing the asterisk process solved the problem: Zap/63-1 (incoming s1 ) Up Bridged Call SIP/2035-e9cb logs say: Jul 22 14:54:12 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6785 seconds Jul 22 14:54:13 NOTICE[17161] chan_sip.c: Disconnecting call 'SIP/2035-e9cb' for lack of RTP activity in 6786 seconds ... warning:Jul 22 14:54:36 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! warning:Jul 22 14:54:36 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! warning:Jul 22 14:54:37 WARNING[26237] channel.c: Avoided initial deadlock for '0xb7c861b8', 10 retries! ... tones of these messages... I'm using latest CVS HEAD. thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] all zap channels get RING signal when starting *
hi all, when i start * all zap channels get ring signal so i get a huge number of incoming dummy calls when starting *. i'm using TE105P with 4 TA750 full fxo with latest CVS HEAD: zaptel.conf: span=1,0,0,esf,b8zs fxsks=1-24 span=2,0,0,esf,b8zs fxsks=25-48 span=3,0,0,esf,b8zs fxsks=49-72 span=4,0,0,esf,b8zs fxsks=73-96 loadzone=us defaultzone=us zapata.conf: [channels] context=incoming callerid=asreceived busydetect=yes busycount=7 faxdetect=no signalling=fxs_ks overlapdial=no usecallerid=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 channel = 1-96 thanks. Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no active channel but one active call???
hi, what does this mean?: www*CLI show channels Channel (ContextExtensionPri ) State Appl. Data 0 active channels 1 active call after some searchs got this: www*CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Format Last Msg 172.22.22.27 239920830697669 00101/3343865 ulaw Rx: ACK 1 active SIP channel(s) logs say: Jul 22 20:28:59 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:00 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8106 seconds Jul 22 20:29:00 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:01 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8107 seconds Jul 22 20:29:01 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:02 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8108 seconds Jul 22 20:29:02 WARNING[26237] channel.c: Exceptionally long queue length queuing to SIP/2399-27f7 Jul 22 20:29:02 NOTICE[26237] chan_sip.c: Disconnecting call 'SIP/2399-27f7' for lack of RTP activity in 8108 seconds thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Force SIP peers to Re-Autheticate
hi all, is there any way to force all sip peers to re-authenticate themselves? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm400p not working after cvs-head update
I have the same problem. seems that tdm400b is not working on CVS HEAD On 6/18/05, Steve Totaro [EMAIL PROTECTED] wrote: did you udate first? - Original Message - From: David Romero To: Asterisk-Users@lists.digium.com Sent: Friday, June 17, 2005 9:36 AM Subject: [Asterisk-Users] tdm400p not working after cvs-head update I hava tdm400p card on [EMAIL PROTECTED] box, this card works fine for weeks, today i did a CVS update to the latest head files and the card is not working. Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) HELP!. thanks David Romero ## ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zap to zap bridging not hanging up
i have the same problem.
it seems to be a bug.
On 6/5/05, Master Abi [EMAIL PROTECTED] wrote:
Hi
I am trying to develop a night divert. Caller dials in after hours on
Zap and it gets divert to a mobile number via a second Zap. The call
bridges but will not hangup the channels when the parties finish.
Is there something I am missing or an dial option that I should be
using. I am using latest CVS.
[night]
exten = s,1,Answer
exten = s,2,Wait,1
exten = s,3,Set(TIMEOUT(digit)=3)
exten = s,4,Set(TIMEOUT(response)=6)
exten = s,5,Set(dvt=${DB(DIVERT/MOBILE)})
exten = s,6,Gotoif($[${dvt} != ]?s|7:s|103)
exten = s,7,Dial,${PSTNTRUNK}/${dvt}|30|tr
exten = s,8,Hangup
[default]
include = melton-night|17:31-8:59|mon-fri|*|*
Thanks
master
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[Asterisk-Users] Detecting DeadLocks
Is there any way to detect * deadlocks automatically? i.e with a running program in background. Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lots of RTP checksum errors
Hi all, i'm getting about 1-3 NOTICE[1915]: rtp.c:453 ast_rtp_read: RTP: Received packet with bad UDP checksum message per call on CVS HEAD from 31 Mar. which seems some changes regarding rtpchecksums is made at that time. setting rtpchecksums to no or yes in rtp.conf doesn't make any sense. now i'm using latest CVS Head. any ideas? Thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lots of RTP checksum error
Hi all, i'm getting about 1-3 NOTICE[1915]: rtp.c:453 ast_rtp_read: RTP: Received packet with bad UDP checksum message per call on CVS HEAD from 31 Mar. which seems some changes regarding rtpchecksums is made at that time. setting rtpchecksums to no or yes in rtp.conf doesn't make any sense. now i'm using latest CVS Head. any ideas? Thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and CAS
what about CAS 3 Bit? does * support it? thanks, Paradise Dove On 4/8/05, Steve Underwood [EMAIL PROTECTED] wrote: David Hajek wrote: Hi, is it possible to use Asterisk with T110P and CAS (channel associated signalling)? There are hundreds of CAS protocols. Quite a few currently work with the T110P. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Presence:
also add snom-190 and snom-360 to your list PolyCom 500 and 600 have the same feature too. On 4/15/05, Brian Leyton [EMAIL PROTECTED] wrote: Or Flash Operator Panel. http://www.asternic.org Brian Leyton IT Manager Commercial Petroleum Equipment -Original Message- From: Henry Devito [mailto:[EMAIL PROTECTED] Sent: Thursday, April 14, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Presence: The Snom 220 and Side cars you can have up to 3 side cars on a 220 there are 20 buttons on each side car. - Original Message - From: Sean Kennedy [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, April 14, 2005 1:42 PM Subject: [Asterisk-Users] Line Presence: Hi all With the recent thread on line presence in asterisk, can anybody tell me if there is a phone out there that supports this? Say I have 20 extensions: Is there any way, hardware based, for me to see the activity on those lines. And for a bonus, is there any way for me to interact with them? Thank you. Sean ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Xten-lite for linux
i have the same plroblem. no link on xten site! On Thu, 31 Mar 2005 14:49:11 -0300, Carlos Gabriel Drach [EMAIL PROTECTED] wrote: Kris Edwards wrote: This is the best linux sip phone I've used so far. Audio quality has been perfect and it seems really stable, so hopefully it will be out of beta soon. I might actually pay for the full version! (not counting console games, that would be the second piece of software I've purchaced since 1987). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Where can i get that version? Not found any link on xten site... Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback on busy
consider this scenario: A Calls B B transfers A to C C (is busy or does not answer) so A backs to B On Tue, 1 Mar 2005 23:07:17 +0330, Paradise Dove [EMAIL PROTECTED] wrote: consider this scenario: A Calls B B transfers A to C C (is busy or does not answer) so B backs to A On Tue, 1 Mar 2005 14:25:33 -0500, C F [EMAIL PROTECTED] wrote: use retrydial. in the cli type show application retrydial have fun. On Tue, 1 Mar 2005 22:17:35 +0330, Paradise Dove [EMAIL PROTECTED] wrote: hi, is there anyway to implement callback on busy and callback on no answer on asterisk? has anybody done this before? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.6 music-on-hold
upgrade to latest CVS Stable. it's solved there! On Wed, 2 Mar 2005 22:02:58 -0600, Eric Rees [EMAIL PROTECTED] wrote: I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom Hold button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callback on busy
hi, is there anyway to implement callback on busy and callback on no answer on asterisk? has anybody done this before? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callback on busy
consider this scenario: A Calls B B transfers A to C C (is busy or does not answer) so B backs to A On Tue, 1 Mar 2005 14:25:33 -0500, C F [EMAIL PROTECTED] wrote: use retrydial. in the cli type show application retrydial have fun. On Tue, 1 Mar 2005 22:17:35 +0330, Paradise Dove [EMAIL PROTECTED] wrote: hi, is there anyway to implement callback on busy and callback on no answer on asterisk? has anybody done this before? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ericsson MD-110 and Dig-410
does MD 110 support SIP? On Thu, 24 Feb 2005 15:08:49 +, Niksa Baldun [EMAIL PROTECTED] wrote: Your span definition should be fine (except there should be commas instead of dots, but that is probably just a typo). You need to play with various parameters on the MD-110 side, those in RODAI command, as well as SIG parameter in ROCAI command. I don't know how well is QSIG implemented in libpri, but interconnecting should be possible even without it. It is a pain in the butt, but I am afraid that trial-and-error is the only way to go. Theodoros Georgiou wrote: Hello All I am wondering is someone knows how to configure the * to work with an Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the right configuration in the zaptel.conf ? I currently have it configured as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I connect it to the PBX and no activity can be seen either in the logs or in the asterisk console. The port is responding when I connect it to the external PRI. Can anybody help ? Anyone who has seen that before ? Thanks Theo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip wifi phone?
what about senao SI-7800H? this is the link: http://www.senao.com.tw/english/product/product_wireless01_outdoor_1.asp?pgtl=Wirelesstp1id=02tp2id=06proid=000131 On Mon, 21 Feb 2005 23:42:30 -0600, Kristian Kielhofner [EMAIL PROTECTED] wrote: Kurt Fankhauser wrote: Sounds like I'm going to have to wait and hope some new phones are released. Kurt, Check out my message from October: http://lists.digium.com/pipermail/asterisk-users/2004-October/067769.html and here is a link to the Broadcom page: http://www.broadcom.com/products/product.php?product_id=BCM1160category_id=45 I really, really wish someone, anyone, would start cranking out some devices based off of these chips. Linksys is really into Broadcom. Why not them? (As long as they don't have a blue plastic case!) -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] snom soft phone
what is the password for Administrator in the softphone? On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke [EMAIL PROTECTED] wrote: Go to the web page, in Preferences there are two pull down menus for Audio Input and Autio Output. CS -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Juan J. Sierralta P. Sent: Tuesday, February 08, 2005 2:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] snom soft phone Hi, How do I change the default audio device ? I have one of those USB headset (which actually is another soundcard) but the simulation insist in using my Soundblaster Live card :( -- Juanjo sin .sig :( ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] not sharing IRQ's
but when i remove uhci_hcd module i will fall in a big trouble, look: the problem will solve when i load uhci_hcd again!! i've a TE405P card installed and modules loaded. Feb 6 08:11:16 WARNING[2907]: Failed to create new channel thread Feb 6 08:11:16 WARNING[2907]: Failed to start PBX :( Feb 6 08:13:57 WARNING[2907]: Failed to create update thread! Feb 6 08:14:12 WARNING[2907]: Failed to create new channel thread Feb 6 08:14:12 WARNING[2907]: Failed to start PBX :( Feb 6 08:14:12 WARNING[2907]: Failed to create update thread! Feb 6 08:14:18 WARNING[2907]: Maximum retries exceeded on call Feb 6 08:16:26 WARNING[2907]: Failed to create update thread! Feb 6 08:16:26 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:26 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:26 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:27 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create new channel thread Feb 6 08:16:27 WARNING[2907]: Failed to start PBX :( Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:16:27 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:17:35 WARNING[2907]: Failed to create update thread! Feb 6 08:18:13 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:13 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:13 WARNING[2907]: Failed to create update thread! Feb 6 08:18:13 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:13 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:15 WARNING[2907]: Failed to create update thread! Feb 6 08:18:15 WARNING[2907]: Failed to create update thread! Feb 6 08:18:16 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:16 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:17 WARNING[2907]: Unable to start simple switch thread on channel 34 Feb 6 08:18:17 WARNING[2907]: Cannot kill myself Feb 6 08:18:17 WARNING[2907]: Failed to create update thread! Feb 6 08:18:20 WARNING[2907]: Failed to create update thread! Feb 6 08:18:20 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:20 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:21 WARNING[2907]: Cannot kill myself Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! Feb 6 08:18:26 WARNING[2907]: Unable to start simple switch thread on channel 34 Feb 6 08:18:26 WARNING[2907]: Cannot kill myself Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! Feb 6 08:18:26 WARNING[2907]: Failed to create new channel thread Feb 6 08:18:26 WARNING[2907]: Failed to start PBX :( Feb 6 08:18:26 WARNING[2907]: Failed to create update thread! On Wed, 12 Jan 2005 20:08:15 -0500, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: just to make sure: when i have zaptel devices on my box and i also use meetme and iax2, do i need to have USB device enabled and it's modules loaded? No your zaptel device will provide the needed hardware timer the USB timer hack is for when you don't have any digium card ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Reproducible crash with CVS stable (from about 5 days ago...) - but only from iax clients
submit a bug in bug tracker at http://bugs.digium.com
On Wed, 2 Feb 2005 21:55:16 +0100, Robert Rozman [EMAIL PROTECTED] wrote:
Hi,
I've spotted weird crash of Asterisk cvs Stable. I have defined queue in
queues.conf :
[prodaja]
music = default
announce = queue-markq
strategy = ringall
context = from-pstn
timeout = 15
retry = 5
maxlen = 0
announce-holdtime = no
announce-frequency = 30
announce-holdtime = yes
monitor-format = gsm|wav|wav49
monitor-join = yes
eventwhencalled = yes
member = Agent/1000
---
then I have in dialplan :
exten = 51,1,Queue(prodaja)
Now when I call 51 from BT100, everything is OK, musiconhold plays, I hear
announcements...:
Asterisk CVS-v1-0-01/28/05-15:21:35, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-v1-0-01/28/05-15:21:35 currently running on
centrala (pid = 28749)
-- Remote UNIX connection
Verbosity was 3 and is now 11
centrala*CLI MANAGER LOGIN MD5 127.0.0.1, admin, amp111
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- Executing Queue(SIP/201-ec33, prodaja) in new stack
-- Started music on hold, class 'default', on SIP/201-ec33
-- Stopped music on hold on SIP/201-ec33
-- Playing 'queue-youarenext' (language 'si')
-- Told SIP/201-ec33 in prodaja their queue position (which was 1)
-- Playing 'queue-thankyou' (language 'si')
-- Started music on hold, class 'default', on SIP/201-ec33
-- Saved useragent Grandstream BT100 1.0.5.18 for peer 201
-- Stopped music on hold on SIP/201-ec33
-- User disconnected when they almost made it
== Spawn extension (from-internal, 51, 1) exited non-zero on
'SIP/201-ec33'
-- Executing Macro(SIP/201-ec33, hangupcall) in new stack
-- Executing ResetCDR(SIP/201-ec33, w) in new stack
-- Executing NoCDR(SIP/201-ec33, ) in new stack
-- Executing Wait(SIP/201-ec33, 5) in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'SIP/201-ec33' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-ec33'
centrala*CLI
centrala:~ #
-
But asterisk constantly crashes with IAX clients calling this number (same
with Firefly, iaxcommclient, Iaxphone), I get:
centrala:~ # asterisk -vvvrgc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk CVS-v1-0-01/28/05-15:21:35, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-v1-0-01/28/05-15:21:35 currently running on
centrala (pid = 27854)
Verbosity was 3 and is now 11
-- Accepting AUTHENTICATED call from 192.168.0.101, requested format =
1024, actual format = 8
-- Executing Queue(IAX2/[EMAIL PROTECTED]/2, prodaja) in new stack
centrala*CLI
Disconnected from Asterisk server
/usr/sbin/safe_asterisk: line 40: 27854 Floating point exception(core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
Executing last minute cleanups
Asterisk ended with exit status 136
Asterisk exited on signal 8.
Automatically restarting Asterisk.
Asterisk cleanly ending (0).
--
Also to notice, that moh plays ok in other situations from iax clients
(hold, transfer, conference)... :
sterisk CVS-v1-0-01/28/05-15:21:35, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
Connected to Asterisk CVS-v1-0-01/28/05-15:21:35 currently running on
centrala (pid = 28749)
Verbosity is at least 11
-- Accepting AUTHENTICATED call from 192.168.0.101, requested format =
1024, actual format = 4
-- Executing MeetMe(IAX2/[EMAIL PROTECTED]/1, 81|pMs) in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '81'
-- Playing 'conf-onlyperson' (language 'si')
-- Started music on hold, class 'default', on IAX2/[EMAIL PROTECTED]/1
-
Any hint, any advice ?
Regards,
Rob.
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[Asterisk-Users] Unable to create channel of type 'Zap' (cause 0)
what is the meaning of (cause 0). i know that in * code it indicates an undefined cause but that's not enough. i have many of this message in my logs. what would be the posiible causes for this message? i have also the same message with SIP channels... thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoided deadlock
but still the main question mark remains:
what are the possible causes which make this warning appear
thanks!
On Tue, 1 Feb 2005 18:10:03 +0330, Paradise Dove [EMAIL PROTECTED] wrote:
but still the main question mark remains:
what are the possible causes which make this warning appear
thanks!
On Tue, 25 Jan 2005 16:36:29 -0500, Paul Rodan [EMAIL PROTECTED] wrote:
This started showing up a few upgrades ago. It always avoids a deadlock, for
SIP and IAX, it's friggin annoying but I don't see any actual issues,
everything works as normal.
The most common device it avoids a deadlock on is my Cisco 3640 router. I
have 6 Voice T1/PRI's plugged into it, it converts it to/from SIP. My
Asterisk server will go to push a call to the Cisco via SIP and I get that
message quite often, but the call goes through perfectly.
Seems like a waste to log it if nothing is actually wrong and everything
works normal. I can't seem to find out if it's giving me a warning about
something that should not occur. I've rebooted the Cisco 3640 and my
Asterisk server to no Avail.
I'm using a Super Micro Celeron 2.8ghz w/ 512mb of RAM and a 40gb IDE hard
drive, running Gentoo 2004.3; No digium hardware, just zaptel/zaprtc w/
rtcsetup running in the background.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise Dove
Sent: Tuesday, January 25, 2005 3:10 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Avoided deadlock
it would help to know all the possible causes for this warning,
something like:
- kernel
- hardware latency (MB, cpu, ...)
- buggy sip device
- lack of resource
- ...
just let us know if anybody knows.
thanks,
Paradise Dove
On Mon, 24 Jan 2005 11:28:34 -0600, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Brian West wrote:
Repeat after me... WARNING != ERROR. This is just letting you know that
it
walked the channel list and did avoid a dead lock by not trying to grab a
lock on a channel that's already locked.
if (ast_mutex_trylock(l-lock)) {
if (retries 10)
ast_log(LOG_DEBUG, Avoiding initial deadlock for
'%s'\n, l-name);
else
ast_log(LOG_WARNING, Avoided initial deadlock for
'%s',
%d retries!\n, l-name, retries);
Read the code it tells you... channel.c
bkw
I'd suggest posting a bug if you haven't already and if you have
purchased
any Digium products I would recommend calling them as well. The
ast_channel_walk_locked error is a rare and hard to diagnose problem and
the
bug trackers and Digium would be the best people to help you.
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Re: [Asterisk-Users] Single or Dual Processor? High volume MeetMe
so you mean that it depends on the type of motherboard and the chipset which is using. am i right? if yes, which mainboards and chipsets is recommended for a large scale * box? On Mon, 31 Jan 2005 12:19:50 -0800, William Boehlke [EMAIL PROTECTED] wrote: On Intel it is our experience that the constraint is the PC bus. Throughput tops out at somewhere between 50 and 100 calls depending on disk speed, without ever using a meaningful part of one processor. William Boehlke Signate -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Spencer Nassar Sent: Sunday, January 30, 2005 11:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Single or Dual Processor? High volume MeetMe Has anyone benchmarked Asterisk on a dedicated single versus dual processor machine? Or could any Asterisk developers comment on whether it is architected in such a way that threads could run on multiple CPUs (especially MeetMe2)? At a higher level, can I host more simultaneous lines and/or conferences for MeetMe if I use a dual processor machine versus single? Also, any info on memory use with high numbers of conference users (100, 1000)? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRTP support
hi all, just want to know, if there is any workaround to add SRTP support to *. as i know there is an open source library (libsrtp http://srtp.sourceforge.net/srtp.html) which makes it more possible to be done. any idea? thanks, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Crash
hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? () #1 0x080d425d in _IO_stdin_used () #2 0x in ?? () ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Crash
this is what i've typed to get the crash info: gdb /usr/sbin/asterisk --core=/core.3673 is it wrong? On Sun, 30 Jan 2005 03:11:24 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote: hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc Seg faults can be faulty memory, overheated CPU, but usually it is an error in programming. #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? () #1 0x080d425d in _IO_stdin_used () #2 0x in ?? () Next time provide the asterisk binary along with the core file to gdb so you can get symbol names and line numbers. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Crash
the same result! On Sun, 30 Jan 2005 03:24:38 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Sun, 2005-01-30 at 12:46 +0330, Paradise Dove wrote: this is what i've typed to get the crash info: gdb /usr/sbin/asterisk --core=/core.3673 Not sure if that is wrong, but I also see from the gdb man page that you should be able to start it by gdb /usr/sbin/asterisk /core.3673 On Sun, 30 Jan 2005 03:11:24 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote: hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc Seg faults can be faulty memory, overheated CPU, but usually it is an error in programming. #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? () #1 0x080d425d in _IO_stdin_used () #2 0x in ?? () Next time provide the asterisk binary along with the core file to gdb so you can get symbol names and line numbers. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.
i have the same problem... i've also added a feature request to bug tracker (http://bugs.digium.com/bug_view_page.php?bug_id=0002612) regarding this issue. On Sun, 30 Jan 2005 13:40:06 -0600, Jon Gabrielson [EMAIL PROTECTED] wrote: Can't asterisk look for a dialtone? Even a $5 modem can detect whether or not there is a dialtone. Thanks, Jon. On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote: When I place a call with asterisk, asterisk will try to dial out on the first line even if the first line is already being used by someone else. Any ideas on what I'm doing wrong? My question would be, how would asterisk know the line is in use if it isn't controlling it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk tries to dial out on lines already in use.
On Sun, 30 Jan 2005 18:14:38 -0600, Jon Gabrielson [EMAIL PROTECTED] wrote: Asterisk should be able to do this, there are several cases when this is essential. The first is a shared/party line where asterisk cannot have guaranteed access for whatever reason. In our case, that reason happens to be because we also use our outgoing lines for faxing. The second is that without dialtone detection, if for some reason the line is down, asterisk needs to know so that it can try a different outgoing line. If the first line is down, asterisk shouldn't hang, it should wait a few seconds and try to dial out on the next line. this is the feature which other PBXs have. the ability to detect out of order lines (no dialtone - used by others) . btw, as i said before a feature request about this is submitted to bug tracker at http://bugs.digium.com/bug_view_page.php?bug_id=0002612. suppose that any followups could be done there. maybe setting bounty on this issue speedup the process! thanks, Paradise Dove Jon. On Sunday 30 January 2005 04:13 pm, Steven Critchfield wrote: On Sun, 2005-01-30 at 13:40 -0600, Jon Gabrielson wrote: Can't asterisk look for a dialtone? Even a $5 modem can detect whether or not there is a dialtone. Maybe you should just use your $5 modem and write your own software. Asterisk is a PBX. PBXs shouldn't have to deal with your bastardized setup that doesn't respect the normal way in which a PBX is set up. A PBX sits between the PSTN and ALL other access to the PSTN. In doing so, asterisk can know ahead of time that the line is available. If you wait for dialtone detection, then you have to also make code to understand all international dialtones as well. Then you have to delay dial till you are certain it is the tone you are expecting. On Sunday 30 January 2005 10:37 am, Wilson Pickett wrote: When I place a call with asterisk, asterisk will try to dial out on the first line even if the first line is already being used by someone else. Any ideas on what I'm doing wrong? My question would be, how would asterisk know the line is in use if it isn't controlling it? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoided deadlock
it would help to know all the possible causes for this warning,
something like:
- kernel
- hardware latency (MB, cpu, ...)
- buggy sip device
- lack of resource
- ...
just let us know if anybody knows.
thanks,
Paradise Dove
On Mon, 24 Jan 2005 11:28:34 -0600, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
Brian West wrote:
Repeat after me... WARNING != ERROR. This is just letting you know that it
walked the channel list and did avoid a dead lock by not trying to grab a
lock on a channel that's already locked.
if (ast_mutex_trylock(l-lock)) {
if (retries 10)
ast_log(LOG_DEBUG, Avoiding initial deadlock for
'%s'\n, l-name);
else
ast_log(LOG_WARNING, Avoided initial deadlock for '%s',
%d retries!\n, l-name, retries);
Read the code it tells you... channel.c
bkw
I'd suggest posting a bug if you haven't already and if you have purchased
any Digium products I would recommend calling them as well. The
ast_channel_walk_locked error is a rare and hard to diagnose problem and
the
bug trackers and Digium would be the best people to help you.
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Re: [Asterisk-Users] Which is better IP500/IP600 or /CP7960
polycom is better for the same quality and lower price. On Sun, 16 Jan 2005 17:27:20 -0800 (PST), Robert Augustyn [EMAIL PROTECTED] wrote: Any preferences? And why? Thanks in advance. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Return of experience : Asterisk more stable with 2.6 or 2.4
i have no problem with 2.6. On Sat, 15 Jan 2005 13:12:18 +, Jeremy SALMON [EMAIL PROTECTED] wrote: Hi, Just a question, For you, what is the more reliable kernel for an asterisk prod server... Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So many Asterisk Patches - Which do I choose and use?
type these 3 command inorder to get CVS HEAD. export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login cvs checkout zaptel libpri asterisk On Wed, 12 Jan 2005 16:20:06 +, John Middleton [EMAIL PROTECTED] wrote: When you say CVS HEAD is the the same as stable? where do you get it from and what params do you use? On Wed, 12 Jan 2005 17:03:34 +, Niksa Baldun [EMAIL PROTECTED] wrote: There is no easy answer to your question. If you ask me, I prefer not to use any patches, except that I am forced to use bristuff because I have quadBRI ISDN cards. Bristuff patches Zaptel in order to enable using quadBRI and octoBRI cards, and also adds some features to *. More info on www.junghanns.net. Like you said, really valuable patches will make it to the CVS sooner or later, so I prefer to wait because it makes installation and maintenance easier. I use Gentoo with 2.6 kernel. I am not sure whether you will get any benefits from upgrading, but I didn't have any problems with it (except that I had to migrate from devfs to udev, but that issue exists with 2.4 kernel too). Paul Rodan wrote: Ok, I usually use the latest stable CVS, with no patches or modifications. If figured if there was a worthwhile patch, Mark would have already included it. However, there was that neat patch about being able to press a certain key and it'd begin recording in mid-stream, that was an awesome feature and I patched my latest features.c file with that patch. But I keep seeing mentions of other patches, specifically something about the MOH patch, the BRISTUFFED patch, and now I'm hearing about a Super Parking Lot patch? For now I've been using the mpg123 method, it tends to work for me, but if I can save CPU/RAM and other troubles by using another format, which one do I go with? What is BRISTUFFED? And if I'm right, the super parking lot patch allows for call parking based on context, a way to break it apart, instead of making it universal across the whole system (where can I find this patch)? So I'm going to ask the question, if I were to install the latest CVS Stable tonight, which patches should I install on it before compiling? Also, I'm using Gentoo Linux, with the 2.4.26-r9 gentoo kernel. I've seen issues with people making Asterisk work perfectly with the 2.6 kernel so I've stayed clear of it, but I still see people fighting to make it work and such, I saw one post a while back about the benefits using Asterisk w/ the 2.6 kernel, can somebody please refresh my memory? What are the benefits of using Asterisk with the 2.6 kernel? I'm trying to get the most out of my system. Any help in making tonights compile/upgrade go perfect would be greatly appreciated. Thanks, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] not sharing IRQ's
just to make sure: when i have zaptel devices on my box and i also use meetme and iax2, do i need to have USB device enabled and it's modules loaded? On Wed, 12 Jan 2005 12:24:55 +, Bob Goddard [EMAIL PROTECTED] wrote: On Tuesday 11 January 2005 23:01, Warren Burstein wrote: Michael Welter wrote that I should be worried about the usb module. Would rmmod uhci_hcd be enough, or should I disable it in the BIOS like Shoval said? [...] 169:84365328405830 IO-APIC-level libata, wctdm [...] If it isn't broken, why fix it? B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analogue RAS Server
I don't think it's possible. Asterisk would have to emulate analog modem, does anybody know if there ia any works on emulating analog modems (not specially to work with asterisk). something like Steve's spandsp for fax. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Request to schedule in the past?!?!
it's clear that your processor is overloaded. recommend you to use rawplayer instead of mpg123 for moh by converting your mp3 files to raw using sox (with mp3 support) take a look at cvs head. On Mon, 10 Jan 2005 06:45:54 -0800 (PST), Jason Goecke [EMAIL PROTECTED] wrote: Hello, Ever since I started using Asterisk I always get this error: Jan 10 15:39:26 NOTICE[4501]: res_musiconhold.c:463 monmp3thread: Request to schedule in the past?!?! I have a dedicated system system that really runs only Asterisk: - Pentium III 500Mhz - 128MB of RAM - 10GB of Disk Space - SuSE v9.2 - MySQL - Apache (only for use with Asterisk) - NTP client for clock synch There is no X server, no other apps besides NFS (copying files back and forth) and some standard services. There is no real load on the system as it only runs in a SoHo environ. I notice this error on calls, as there is a short cut-out in the audio. Even if I run Asterisk in non-verbose/non-console mode/psuedo-realtime the problem persists. How do I resolve this once and for all? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] answer supervision for POTS FXO interfaces
the only way is to set callprogress=yes but it's very experimental and makes many wrong alarms. by the way this feature is really missing in *. On Sat, 08 Jan 2005 17:42:42 +0200, Gilad Ben-Yossef [EMAIL PROTECTED] wrote: Samudra E. Haque wrote: hello, using Asterisk, is there any clever way to provide answer supervision based upon the received audio only from the FXO interface (from a public PSTN switch that does not have battery reversal, or CPC). In zapata.conf use either busydetecgt=yes busycount=6 (it will take about 10 seconds to indetify the hangup or busy) If you're lucky, you can try the experimental callprogress=yes Cheers, Gilad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using native moh
i dont know how to use * native moh feature which is added recently to CVS HEAD each time i hold a call i will get this warning on cli: WARNING[24235]: res_musiconhold.c:837 local_ast_moh_start: No class: default Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does TE405P support 3Bit CAS?
does TE405P support 3Bit CAS? what are the configuration tips? thanx, Paradise Dove ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Segmentation fault
I get seg. fault with my * box. at the crash time i had about 35 Bridged Channel. i have: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with gdb. this is the output of gdb: Program terminated with signal 11, Segmentation fault. #0 0xb7fbbce4 in ?? () ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Segmentation fault
I'm using FC2. but with a fresh 2.6.9 kernel downloaded from kernel.org. I've recently upgraded my Glibc to glibc-2.3.3-27.1. I'm also using ECC Reg Memory. and this is my Xeon CPU info: (HyperThreading is ON) processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Xeon(TM) CPU 3.20GHz stepping: 5 cpu MHz : 3199.895 cache size : 512 KB physical id : 0 siblings: 2 fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr bogomips: 6340.60 On Mon, 13 Dec 2004 13:58:33 +0100, Andreas Sikkema [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: - dual xeon box (3.2Ghz) - 2Gb of memory - E7501 chipset motherboard. - U320 scsi disks - intel Gb ethernet device. - i only use sip for clients (no fxs in box) - TE405P for fxo (with 4 atran TA750). - ulaw is used as codec and echo cancellationo is enabled. but the core dump file has nothing to show with gdb. this is the output of gdb: Program terminated with signal 11, Segmentation fault. #0 0xb7fbbce4 in ?? () ___ What Linux (assuming you're running Linux) distribution are you running? I have seen lots of this kind of problems before. We had lots of stability problems with GNUgk on Debian Woody. Once we moved to Sarge we had no problems at all, with uptime going from a couple of days to several months when we had no need for GNUgk anymore. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544f: +31 (0)10 2245540 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Segmentation fault
I got another crash... the core dumped file shows that the crash has been occurred at the same point as the previous crash. Program terminated with signal 11, Segmentation fault. #0 0xb7fbbce4 in ?? () ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Segmentation fault
I have seen lots of this kind of problems before. We had lots of stability problems with GNUgk on Debian Woody. is there any relation between * and GNUgk? thanks Paradise Dove ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange Segmentation fault
Hmm without knowing anything else about your specific situation: A signal 11 most often is caused by a hardware malfunction, for instance a rotten bit in your memory or something.Any chance you could do some heavy diagnostics on that machine ? I've just seen a new update of my M.B. Bios, does it (upgrading a bios) seem to fix the problem? thanks, Paradise Dove ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sangoma
I'm using an A101u and it seems to work fine connected to a Carrier Access Access Bank I (24 FXS). How did you get it working with asterisk? - Paradise Dove ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoided deadlock
what does this warning really mean? does it have any side effect on my * box? 'cose I've recently had random seg. faults on my box. I'm using latest CVS -r v1-0 Dec 1 12:08:42 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:43 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:45 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:48 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:48 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:50 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:55 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:56 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:57 WARNING[6189]: Avoided deadlock for 'SIP/2502-6303', 10 retries! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoided deadlock
Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this warning really mean? I have tones of them! does it have any side effect on my * box? 'cose I've recently had random seg. faults on my box. I'm using latest CVS -r v1-0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoided deadlock
at the same time I have also this notice log. this makes my problem more meaningful. i think it might be a bug inside *. (am i right?) Dec 1 12:44:46 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4794 seconds Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4795 seconds Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4795 seconds Paradise Dove Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this warning really mean? I have tones of them! does it have any side effect on my * box? 'cose I've recently had random seg. faults on my box. I'm using latest CVS -r v1-0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Avoided deadlock
but i have already an UltraWide 320 Scsi HardDisk installed on my * box. seems that this won't be the cause of my problem at least. i think that it should be something betweeen these two errors: - NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4794 seconds - WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! I have these two lines in my sip.conf rtptimeout=300 rtpholdtimeout=480 it seems that these options don't work as expected. Paradise Dove On Wed, 1 Dec 2004 07:11:53 -0500, mattf [EMAIL PROTECTED] wrote: Hello, I had this problem a few months ago on a machine that I did a lot of recording on. It was caused by slow disk access time. Asterisk would wait for something to write to disk and basically freeze everything. It would always eventually happen to the same machine no matter if I wiped it completely and did a full reinstall. I fixed it by buying 4 new 320-SCSI drives and a new 320SCSI RAID card, no problems since then(6 months). I did report this to Digium several times, they were even in my machine a few times to monitor it, they had no clue what was causing it, and the ast_channel_walk_locked bug has no documentation anywhere about it(not much help to look in the code either). Hope this helps. MATT--- -Original Message- From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 01, 2004 6:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Avoided deadlock at the same time I have also this notice log. this makes my problem more meaningful. i think it might be a bug inside *. (am i right?) Dec 1 12:44:46 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4794 seconds Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4795 seconds Dec 1 12:44:47 NOTICE[6189]: Disconnecting call 'SIP/2502-6303' for lack of RTP activity in 4795 seconds Paradise Dove Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! what does this warning really mean? I have tones of them! does it have any side effect on my * box? 'cose I've recently had random seg. faults on my box. I'm using latest CVS -r v1-0 I have the same problem as you. But so far did not find an asnwer for it. Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: random echo on TA750
all i have is random echo I have already 4 TA750 with full FXO echocancel=yes and echo training=800 - what should i do? - could it be solved with tweaking echo params on *? - is there any additional devices that can be added between Channel Bank and * to get rid off echo forever? if its the motherboard latency issue then no external device will help so is there any way to prevent motherboard latency? any tweaks? tunning tools? or is there any approved motherboard for running asterisk in a high load environment? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: random echo on TA750
all i have is random echo I have already 4 TA750 with full FXO echocancel=yes and echo training=800 - what should i do? - could it be solved with tweaking echo params on *? - is there any additional devices that can be added between Channel Bank and * to get rid off echo forever? any help would appreciated Paradise Dove ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7912g SIP firmware
how can i get a CCO account? or is there any other place for cisco downloadable stuff without user/pass? or a free to all CCO account!!!?? On Fri, 12 Nov 2004 10:50:18 -0600, Eric Wieling [EMAIL PROTECTED] wrote: You CANNOT download Cisco firmware without a CCO account AND support contract. Jerry Geis wrote: Did you search for 7912 sip software in the search tab? That is where I found mine. Jerry Hello I did register, but I only find manuals and guides. But no software. And when I go to Downloads-Voice Software, I only have - Gatekeeper Transaction Message Protocol - Cisco Voice Call Manager Hotfix Patches But I don't see any IP Phones. :-( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
