[Asterisk-Users] EICON Drivers and CAPI
Hi, Does anyone use EICON Diva Server Card ISDN and CAPI ? I need some help for drivers installation. I use RED HAT 8.0 (kernel 2.4.18-14) I want to know which drivers should I use and how can I install it. Thanks Rattana
Re: [Asterisk-Users] low-cost * (newbie question)
Subject: Re: [Asterisk-Users] low-cost * (newbie question) From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Organization: Date: 01 Apr 2003 09:59:51 -0600 Reply-To: [EMAIL PROTECTED] On Tue, 2003-04-01 at 09:31, Grzegorz Nosek wrote: [snip] thanks for your reply No analog modems. If your ISDN adapter is supported properly, you can place that straight into a asterisk box. The analog line would need a X100P. so to sum things up: one x100p for every analog line and one isdn4linux adapter for every isdn line, right? i'd like to test it on an experimental net with no isdn, so i'd like to use a fake channel to provide something the client can talk to. will a soundcard+gnophone on the * box do? or is there another channel type, dedicated to testing, with no hardware required? in what ways is functionality of the oss channel limited in comparison to, say, isdn? gnophone supports dialing from URLs, If your application could generate web pages to feed URL's the people could click on, then gnophone can accept them from the external app. Otherwise, you may want to watch the the whole app has a web interface so i'd probably go this way activity and place sample.call files in the queue as your employee needs a new call to service. -- Steven Critchfield [EMAIL PROTECTED] thanks again, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] low-cost * (newbie question)
On Wed, 2003-04-02 at 05:14, duncan wrote: gnophone on your * server will require X on your * server, and that would not be recomended. On my 1ghz AMD chip the screensaver could cause a severe degradation on my VoIP channels. Put gnophone on your other asterisk boxes and call each other. agreed, this is good advice, but you can run asterisk on a machine also running X without _too_ much of a performance hit, just make sure you have a blank screensaver set - rather than one that activally involves processing to reder itself. likewise its a bad idea to install seti or other programs like it on a server. I used this as a good reason to buy a new desktop for home use so the 1ghz machine was solely for my home asterisk machine. It doesn't run X at all now. Same for my asterisk server at work, it never had X installed on it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi compil error
Hi; I Use RED HAT 8 and chan_capi 0.1.0 I have errors in compilation. it can't find capi20.h and i have subscripted value is neither array nor pointer. help me please. before i have red hat 7.2 it work so i don't why there're errors. Rattana
Re: [Asterisk-Users] will this machine handle it
I am running that same level of box ( with more RAM ) as my test box and it worked nearly acceptably with the T100P but has severe sound quality issues with the T400P running more than 2 calls into VM or playing announcements. It will be marginal, try and find a faster box, even a P-II 450 would be much better. Bill Jeff McClure wrote: Hi folks, Right now I'm running * along with a lot of other apps on my firewall box, which is a P-II 400 with 192MB of RAM. I have a single T100P card connected to a channel bank that's using one FXO and two FXS ports. I want to move * off to another computer (mostly because I think the other apps on the current box are causing enough of a load to affect the sound quality a bit). I'm looking for a computer to put it on, and I've found someone with a P-II 350 with 64MB of RAM (I could steal another 64MB from the firewall if I have to). So, I need an opinion from some more experienced users. Given the same number of ports and assuming I don't run any other apps on the box, is that P-II 350 beefy enough to handle my * setup comfortably? Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA186: Call/Leg Transaction Doesn't Existon local call
Mark Spencer wrote:But if I try to call from one of them to the other, the remote end rings just fine in both cases, but then as soon as asterisk bridges the two channels, the remote end sends a Call/Leg Transaction Doesn't Exist error and hangs up the line. Apparently it doesn't like our reinvites for some reason. Any SIP experts out there able to figure out why the ATA doesn't like them? the Pingtel seems to be okay with them. I set canreinvite=no in the sip.conf for both instruments. That doesn't seem to have made any difference. Does anyone know of a fix for this? I have talked ATA-ATA when they're both on different networks; that's the oddest part. . . B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA186: Call/Leg Transaction Doesn't Existon local call
I set canreinvite=no in the sip.conf for both instruments. That doesn't seem to have made any difference. Does anyone know of a fix for this? I have talked ATA-ATA when they're both on different networks; that's the oddest part. . . Presumably, something still has to be wrong with the utilization of the reinvites. You can use sip debug to watch and confirm that indeed a reinvite is being sent. If not, at least send me a trace *off-list* to look at. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] will this machine handle it
You can try using the -p option to Asterisk. Mark On Wed, 2 Apr 2003, Jeff McClure wrote: Good points. This system currently does not use any SIP or IAX channels (or any other form of VoIP) and only deals with 1 call at a time (the single FXO channel is the only link to the outside). At some point, I may add an IAX link to a friend's * box, but I really don't see this setup ever having to deal with more than 2 concurrent calls (one over IAX, possibly with GSM, the other using just the Zaptel channels with no compression at all). Oh...and voicemail uses GSM, of course. The current sound quality is pretty good, but what I do hear are tiny little hiccups during GSM playback in voicemail. Again, I suspect what I'm hearing is the effect of the load on box spiking due to other processes. Does that sound reasonable? Maybe that extra level of detail can help some folks form opinions about required CPU horsepower. --On Wednesday, April 02, 2003 6:43 AM + WipeOut . [EMAIL PROTECTED] wrote: Hi Jeff.. What you are asking is a little bit of a grey area because there are a number of factors that will affect how well you system will perform.. things like the average number of concurrent calls?, are you using VoIP?, what codecs are you using for the SIP of IAX channels? and no doubt a few others.. But here is my experience.. I am using a PII-400 and with 2 concurrent VoIP calls using G.711 codec the processor barely registers anything.. So I should thing that this system should handle 10-15 concurrent calls... If I used the GSM codec for example I an sure this number would drop significantly.. Hope that helps.. Hi folks, Right now I'm running * along with a lot of other apps on my firewall box, which is a P-II 400 with 192MB of RAM. I have a single T100P card connected to a channel bank that's using one FXO and two FXS ports. I want to move * off to another computer (mostly because I think the other apps on the current box are causing enough of a load to affect the sound quality a bit). I'm looking for a computer to put it on, and I've found someone with a P-II 350 with 64MB of RAM (I could steal another 64MB from the firewall if I have to). So, I need an opinion from some more experienced users. Given the same number of ports and assuming I don't run any other apps on the box, is that P-II 350 beefy enough to handle my * setup comfortably? Thanks, Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Jeff McClure [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] will this machine handle it
a) do you have a preemtive kernel and low-latency scheduling I'm running a stock 2.4.20 kernel. You could call this my production box, so I'm hesitant to run development/customized kernels on it. If I'm able to split * out onto its own box, I will be much more willing to try optimizations like this (I'm thinking about using Gentoo so I can compile the entire system from the ground up with optimizations). Do you have a suggestion for a good kernel version to use for this? b) have you tried (temporarily) removing any apps that may load the system and still seen if the hiccups or other such symptoms are present Since this box provides so many services to me, it's inconvenient to strip it down like that, so I have not tried it yet. However, it's a good idea, so I'll try to create a maintenance window to try it out soon. c) i've run a single X100P on a P1-100 and not had problems but it was a while back *nod* I don't know what the load looks like for a single X100P versus my T100P with 3 active channels. I do know that the wct1xxp module itself seems to create a load that shows up in top as a system percentage of anything from next to 0% to occasional spikes of about 20%. Quoting [EMAIL PROTECTED]: On Wed, 2 Apr 2003, Jeff McClure wrote: The current sound quality is pretty good, but what I do hear are tiny little hiccups during GSM playback in voicemail. Again, I suspect what I'm hearing is the effect of the load on box spiking due to other processes. Does that sound reasonable? a) do you have a preemtive kernel and low-latency scheduling b) have you tried (temporarily) removing any apps that may load the system and still seen if the hiccups or other such symptoms are present c) i've run a single X100P on a P1-100 and not had problems but it was a while back -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems VOX: +92(51)282-0628 x7400 | FAX: +92(51)282-0621 | IAX: (700)282-0628 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] will this machine handle it
On Wednesday 02 Apr 2003 15:47, Jeff McClure shaped the electrons to say: a) do you have a preemtive kernel and low-latency scheduling I'm running a stock 2.4.20 kernel. You could call this my production box, so I'm hesitant to run development/customized kernels on it. If I'm able to split * out onto its own box, I will be much more willing to try optimizations like this (I'm thinking about using Gentoo so I can compile the entire system from the ground up with optimizations). Do you have a suggestion for a good kernel version to use for this? b) have you tried (temporarily) removing any apps that may load the system and still seen if the hiccups or other such symptoms are present Since this box provides so many services to me, it's inconvenient to strip it down like that, so I have not tried it yet. However, it's a good idea, so I'll try to create a maintenance window to try it out soon. preemptive and low latency kills zaptel on shuttle machines. it really fdepends on your mobo if it will work with this patches and the zaptel drivers. We are using gentoo all over the place but some boxes need vanilla kernels :) c) i've run a single X100P on a P1-100 and not had problems but it was a while back *nod* I don't know what the load looks like for a single X100P versus my T100P with 3 active channels. I do know that the wct1xxp module itself seems to create a load that shows up in top as a system percentage of anything from next to 0% to occasional spikes of about 20%. Quoting [EMAIL PROTECTED]: On Wed, 2 Apr 2003, Jeff McClure wrote: The current sound quality is pretty good, but what I do hear are tiny little hiccups during GSM playback in voicemail. Again, I suspect what I'm hearing is the effect of the load on box spiking due to other processes. Does that sound reasonable? a) do you have a preemtive kernel and low-latency scheduling b) have you tried (temporarily) removing any apps that may load the system and still seen if the hiccups or other such symptoms are present c) i've run a single X100P on a P1-100 and not had problems but it was a while back -- Mirza Wasim Baig | Principal Consultant | Convergence Business Systems VOX: +92(51)282-0628 x7400 | FAX: +92(51)282-0621 | IAX: (700)282-0628 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this correspondence in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or rely on any part of this correspondence if you are not the intended recipient. Any opinions expressed in this message are those of the individual sender. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H.323 support
Julio Tommasi wrote: Have any body succesfully compiled the files in asterisk-oh323-0.2.tar.gz ? This is a very, very old version. Try the latest one (0.5.1) from http://www.inaccessnetworks.com/projects/asterisk-oh323 Michael. I have the following errors: +for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG -I/usr/include -I/usr/include/crypto -I/root/pwlib/include/ptlib/unix -I/root/pwlib/include -I/root/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o wrapper.o cc1plus: warning: changing search order for system directory /usr/include cc1plus: warning: as it has already been specified as a non-system directory wrapper.cxx: In constructor `WrapH323Connection::WrapH323Connection(WrapH323EndPoint, unsigned int, int, int, short unsigned int)': wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this function) wrapper.cxx:563: (Each undeclared identifier is reported only once for each function it appears in.) wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 I have the latest versions of PWlib and openh323. May be I need the same versions that appear in the README file, but I can't get them. Does any body have these versions ? Thanks Julio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Buying channel banks online?
Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Thnaks -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying channel banks online?
On Wed, 2003-04-02 at 10:43, WipeOut . wrote: Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Not off hand. Are you having problems with ebay, or people unwilling to ship to where you are? If it is an unwillingingness to ship to you, I could possibly help you out there. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation fault
Title: segmentation fault Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. Segmentation fault. No coredump found. In case 1 there was a significant memory growth: Top at the startup: 15986 root 9 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk 15987 root 8 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk Top in several hours: 15986 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk 15987 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk Top after a day: 27441 root 9 0 45980 44M 2156 S 0.0 4.5 0:00 asterisk 27442 root 8 0 45980 44M 2156 S 0.0 4.5 0:16 asterisk Actually, I saw it over 50. There were some warning messages on the way. For example: Apr 1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 103, but we have nothing -- resetting! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 30, but we have nothing -- resetting! Apr 1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 87 failed: Unknown error 500 Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Question: What do I do to give you more info? Should I issue 'ulimit -c unlimited' to get a coredump? Are there any flags/modes to set? Thank you. Alex Zarubin
Re: [Asterisk-Users] segmentation fault
asterisk -vvvcg (use g option to generate the coredump file) than gdb asterisk core.pid bt Also you might send a log of pri intense debug span number regards Martin On Wed, 2 Apr 2003, Alex Zarubin wrote: Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. Segmentation fault. No coredump found. In case 1 there was a significant memory growth: Top at the startup: 15986 root 9 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk 15987 root 8 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk Top in several hours: 15986 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk 15987 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk Top after a day: 27441 root 9 0 45980 44M 2156 S 0.0 4.5 0:00 asterisk 27442 root 8 0 45980 44M 2156 S 0.0 4.5 0:16 asterisk Actually, I saw it over 50. There were some warning messages on the way. For example: Apr 1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 103, but we have nothing -- resetting! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 30, but we have nothing -- resetting! Apr 1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 87 failed: Unknown error 500 Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Question: What do I do to give you more info? Should I issue 'ulimit -c unlimited' to get a coredump? Are there any flags/modes to set? Thank you. Alex Zarubin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZHONE Fix !! (long)
Everyone - thought I would pass on a useful piece of information. Finally got a solution to my phantom ringing problem. Problem - the zhone is triggered into detecting ringing by the Automatic Line Insulation Tests (ALIT or LIT) run nightly automatically by the telco. Here it is twice between 8pm and 9pm on my particular lines. My first approach before I knew specifically the buzzword for the testing, was to call up repair and describe the problem, and they denied knowledge of any such tests, sent a tech out to test the line - surprise he found nothing since there is nothing physically wrong with the line. I got frustrated and gave up at the time. A few months of annoying ringing go on, I get pissed off again and start searching and find out indeed there is automated testing going on all the time. Now granted the real problem is the zhone itself being oversensitive, but I take the view that you can always fix shit hardware with the right software. After writing off the telco (Bell Canada) as morons, I attempt to find a solution in asterisk - ring debounce looks right so I investigate the zaptel libs. Mark chimes in about how could this be possible yada yada. After investigation I find the debounce set to 500ms, but basically ignored and treated as an on/off value - implementation of which still seems pretty strange, but I am convinced it is not doing much, if any, debouncing of the RBS on the t1. I decide to give up on that since it looks like I would have to redo a large part of the driver to build in the hysterisis I need into the signalling bits. Why this problem is so vexing is I have one line that rings through to a phone, and then transfers to vm, since the line never actually rung, there is no disconnect supervision on the hangup either, so I get an infinitely long voicemail that fills up the whole hd. (yeah I know there are solutions to that but might as well fix the real issue right ?) So I get on the phone to the telco again today with the attitude I usually do when I am pissed off at them, that I will not hang up till they do what I want. Procedure - get the person's name, then insist they solve your problem for you. Be prepared for a long time on hold (3hrs is my record). They will not just hang up on you once you have their name. So I start in today saying I want the LIT cancelled for the 4 lines I have and I am not taking no for an answer since it is so annoying. 45min of hold and transferring later, I am talking to a guy that knows exactly what I am talking about and helps me out. In this area you can even be more specific and say you want an NLT record added to your line. I suspect this is related to the software for the DMS-100 switch so it is likely the same in most places in North America. If this helps even one more person with a zhone and this annoying problem, it was worth writing it all down. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] segmentation fault
Title: RE: [Asterisk-Users] segmentation fault OK, here it is. On a flow of shorter calls it lasted about an hour. [EMAIL PROTECTED] asterisk]# gdb asterisk core.12348 GNU gdb Red Hat Linux 7.x (5.0rh-15) (MI_OUT) Copyright 2001 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i386-redhat-linux... Core was generated by `asterisk -dgvvvc'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/i686/libpthread.so.0...done. warning: Unable to set global thread event mask: generic error [New Thread 1024 (LWP 12342)] Error while reading shared library symbols: Can't attach LWP 12342: No such process Reading symbols from /usr/lib/libncurses.so.5...done. Loaded symbols for /usr/lib/libncurses.so.5 Reading symbols from /lib/i686/libm.so.6...done. Loaded symbols for /lib/i686/libm.so.6 Reading symbols from /lib/i686/libc.so.6...done. Loaded symbols for /lib/i686/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/lib/asterisk/modules/chan_modem.so...done. . . . . Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libnsl.so.1...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /usr/lib/asterisk/modules/format_pcm_alaw.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_pcm_alaw.so #0 0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1) at channel.c:354 354 cur = chan-pvt-readq; (gdb) bt #0 0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1) at channel.c:354 #1 0x0805a9a0 in ast_queue_hangup (chan=0x81cd398, lock=1) at channel.c:391 #2 0x42412855 in handle_request (p=0x82b6350, req=0x42c1b25c, sin=0x42c1b24c) at chan_sip.c:3762 #3 0x42412e6d in sipsock_read (id=0x80d7850, fd=10, events=1, ignore=0x0) at chan_sip.c:3840 #4 0x08050d9e in ast_io_wait (ioc=0x80d9018, howlong=1000) at io.c:268 #5 0x424131f5 in do_monitor (data="" at chan_sip.c:3928 #6 0x4003ec6f in pthread_start_thread (arg=0x42c1bbe0) at manager.c:284 (gdb) -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED]] Sent: Wednesday, April 02, 2003 11:38 AM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] segmentation fault asterisk -vvvcg (use g option to generate the coredump file) than gdb asterisk core.pid bt Also you might send a log of pri intense debug span number regards Martin On Wed, 2 Apr 2003, Alex Zarubin wrote: Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. Segmentation fault. No coredump found. In case 1 there was a significant memory growth: Top at the startup: 15986 root 9 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk 15987 root 8 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk Top in several hours: 15986 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk 15987 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk Top after a day: 27441 root 9 0 45980 44M 2156 S 0.0 4.5 0:00 asterisk 27442 root 8 0 45980 44M 2156 S 0.0 4.5 0:16 asterisk Actually, I saw it over 50. There were some warning messages on the way. For example: Apr 1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 103, but we have nothing -- resetting! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 30, but we have nothing -- resetting! Apr 1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 87 failed: Unknown error 500 Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Question: What do I do to give you more info? Should I issue 'ulimit -c unlimited' to get a coredump? Are there any flags/modes to set? Thank you. Alex Zarubin ___ Asterisk-Users mailing list [EMAIL
Re: [Asterisk-Users] Sip Transfer
cvs update -r 1.x channels/chan_sip.c make install where 'x' is from 1 to 30 version 1.30 is dated 2003-04-02 if not sure check rcs2log -v |more regards Martin On Tue, 1 Apr 2003, Russ Beaupre, P.E. wrote: A while ago SIP transfer via the # key on a call to a cell phone via iconnect was working. I updated to the current CVS tonight and now that functionality is gone. Any ideas as to how to enable it again? Thanks in advance -russ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: ipDialog Ethernet SIP Phone $199
Title: ipDialog SipTone Ethernet Phone Here is a SIP phone I havent seen before. Does anyone have any experience with this one? -Original Message- From: George Richardson [mailto:[EMAIL PROTECTED]] Sent: Wednesday, April 02, 2003 4:56 PM To: [EMAIL PROTECTED] Subject: ipDialog Ethernet SIP Phone $199 $199.00 The ipDialog SipTone Ethernet phone is a cost effective Linux based VoIP telephone designed to provide high quality voice communications over IP networks. The best value that we have found. Check out the specs on this phone before purchasing anything else! The ipDialog SipTone has the major features found in more expensive phones at amuch better price. With this phone, you do not have to sacrifice quality to get something affordable. ipDialog SipTone Features ipDialog SipTone Tech Specs ipDialog SipTone Admin/User Manual Use your ipDialog SipTone with Free World Dialup IPDSipTone $199.00 =r Email: [EMAIL PROTECTED]
Re: [Asterisk-Users] Buying channel banks online?
www.channelbanks.com I have one unit. It works. -alex On Wed, 2 Apr 2003, WipeOut . wrote: Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Thnaks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying channel banks online?
I think it's difficult for me to accept the price. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 02, 2003 6:12 PM Subject: Re: [Asterisk-Users] Buying channel banks online? www.channelbanks.com I have one unit. It works. -alex On Wed, 2 Apr 2003, WipeOut . wrote: Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Thnaks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying channel banks online?
The web site doesn't mention if it can do CPD, and it says it only supports loopstart analog lines. On Wed, 2003-04-02 at 20:12, [EMAIL PROTECTED] wrote: www.channelbanks.com I have one unit. It works. -alex On Wed, 2 Apr 2003, WipeOut . wrote: Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Thnaks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Buying channel banks online?
Via email, we do, our website is www.sharkdata.net - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 02, 2003 11:12 PM Subject: Re: [Asterisk-Users] Buying channel banks online? The web site doesn't mention if it can do CPD, and it says it only supports loopstart analog lines. On Wed, 2003-04-02 at 20:12, [EMAIL PROTECTED] wrote: www.channelbanks.com I have one unit. It works. -alex On Wed, 2 Apr 2003, WipeOut . wrote: Hi, Anyone know of any sites selling channel banks online (apart from ebay).. Preferably with international shipping.. Thnaks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] segmentation fault
Does running it under valgrind produce more useful output? Mark On Wed, 2 Apr 2003, Alex Zarubin wrote: OK, here it is. On a flow of shorter calls it lasted about an hour. [EMAIL PROTECTED] asterisk]# gdb asterisk core.12348 GNU gdb Red Hat Linux 7.x (5.0rh-15) (MI_OUT) Copyright 2001 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i386-redhat-linux... Core was generated by `asterisk -dgvvvc'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libdl.so.2...done. Loaded symbols for /lib/libdl.so.2 Reading symbols from /lib/i686/libpthread.so.0...done. warning: Unable to set global thread event mask: generic error [New Thread 1024 (LWP 12342)] Error while reading shared library symbols: Can't attach LWP 12342: No such process Reading symbols from /usr/lib/libncurses.so.5...done. Loaded symbols for /usr/lib/libncurses.so.5 Reading symbols from /lib/i686/libm.so.6...done. Loaded symbols for /lib/i686/libm.so.6 Reading symbols from /lib/i686/libc.so.6...done. Loaded symbols for /lib/i686/libc.so.6 Reading symbols from /lib/ld-linux.so.2...done. Loaded symbols for /lib/ld-linux.so.2 Reading symbols from /usr/lib/asterisk/modules/chan_modem.so...done. . . . . Loaded symbols for /lib/libcrypt.so.1 Reading symbols from /lib/libnsl.so.1...done. Loaded symbols for /lib/libnsl.so.1 Reading symbols from /usr/lib/asterisk/modules/format_pcm_alaw.so...done. Loaded symbols for /usr/lib/asterisk/modules/format_pcm_alaw.so #0 0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1) at channel.c:354 354 cur = chan-pvt-readq; (gdb) bt #0 0x08055d86 in ast_queue_frame (chan=0x81cd398, fin=0x42c1991c, lock=1) at channel.c:354 #1 0x0805a9a0 in ast_queue_hangup (chan=0x81cd398, lock=1) at channel.c:391 #2 0x42412855 in handle_request (p=0x82b6350, req=0x42c1b25c, sin=0x42c1b24c) at chan_sip.c:3762 #3 0x42412e6d in sipsock_read (id=0x80d7850, fd=10, events=1, ignore=0x0) at chan_sip.c:3840 #4 0x08050d9e in ast_io_wait (ioc=0x80d9018, howlong=1000) at io.c:268 #5 0x424131f5 in do_monitor (data=0x0) at chan_sip.c:3928 #6 0x4003ec6f in pthread_start_thread (arg=0x42c1bbe0) at manager.c:284 (gdb) -Original Message- From: Martin Pycko [mailto:[EMAIL PROTECTED] Sent: Wednesday, April 02, 2003 11:38 AM To: '[EMAIL PROTECTED]' Subject: Re: [Asterisk-Users] segmentation fault asterisk -vvvcg (use g option to generate the coredump file) than gdb asterisk core.pid bt Also you might send a log of pri intense debug span number regards Martin On Wed, 2 Apr 2003, Alex Zarubin wrote: Configuration: Linux wpbx 2.4.9-13 #1 Tue Oct 30 20:11:04 EST 2001 i686 unknown P4 2.5 GHz, 1 GB RAM T400P with 3 T1s plugged in. A flow of 46 calls (spread out over 3 T1s). Each call gets transferred (Dial) to the SIP platform and stays for 5 min. Case 1. Asterisk built out of CVS Mar. 19. Test was running for 3 days. Segmentation fault. Case 2. Asterisk built out of CVS Apr. 1. Test was running for 12 hours. Segmentation fault. No coredump found. In case 1 there was a significant memory growth: Top at the startup: 15986 root 9 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk 15987 root 8 0 6440 6436 2144 S 0.0 0.6 0:00 asterisk Top in several hours: 15986 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk 15987 root 9 0 9192 9188 2148 S 0.0 0.9 0:00 asterisk Top after a day: 27441 root 9 0 45980 44M 2156 S 0.0 4.5 0:00 asterisk 27442 root 8 0 45980 44M 2156 S 0.0 4.5 0:16 asterisk Actually, I saw it over 50. There were some warning messages on the way. For example: Apr 1 23:22:33 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 86 failed: Unknown error 500 Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 102, retransmitting frame 102 now, updating n_r! Apr 1 23:24:54 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 103, but we have nothing -- resetting! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 29, retransmitting frame 29 now, updating n_r! Apr 1 23:28:49 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: !! Got reject for frame 30, but we have nothing -- resetting! Apr 1 23:30:00 WARNING[11276]: File chan_zap.c, Line 5248 (zt_pri_error): PRI: Read on 87 failed: Unknown error 500 Apr 1 23:30:24 WARNING[10251]: File chan_zap.c, Line 5248 (zt_pri_error):
