[Asterisk-Users] FreeBSD Digium g.729 codec seg faults on rev 30652
Greetings- Was running the Digium FreeBSD g.729 codec until recently when the latest Asterisk bits were obtained via svn: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk This produced: Checked out revision 30652 This on FreeBSD 6.1-RELEASE Attempting to start asterisk it returns: == Registered custom function URIENCODE [codec_g729a.so]May 27 13:29:59 WARNING[71884]: loader.c:728 __load_resource: missing mod_data for codec_g729a.so Segmentation fault (core dumped) This codec was licensed through Digium and, while listed as 'unsupported', worked very well. Any way Digium will release a G.729 codec for FreeBSD which is useable with rev 30652 ? regards -kim -- w8hdkim er.. gmail ___ --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] Help on x101p disconnect when called party answers
On Mon, November 21, 2005 2:31 pm, MZ said: hi, using a multimeter i have verified that the analog line we have actually supports polarity reversal when the remote party answers and another reversal on hangup. with this i assume that i can use the kewlstart signalling so that the x101p can automatically disconnect. my problem is -- as soon as the called party answers, the call is disconnected. This is a problem here too.. Is there any way to get the x101p/TDM400p FXO ports to not disconnect when there is a polarity reversal or momentary interruption of CO battery voltage on the PSTN line ? -kim -- [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
[Asterisk-Users] Compile broken on FreeBSD ?
I'm seeing this trying to compile on FreeBSD with source via cvs from cvs.digium.com at ~1000 UTC 6-30: func_enum.c: In function `function_enum':func_enum.c:126: error: too many arguments to function `ast_get_enum'gmake[1]: *** [func_enum.o] Error 1 Also, the cvsup server on cvs.digium.com has been refusing connections for some time. Is cvsup no longer available on this server ? regards -kim -- w8hdkim er gmail.com ___ --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] Music on Hold Quality
On Wed, September 28, 2005 5:41 pm, Matt said: I have heard this issue when on hold with Cisco and Vonage... Idon't think it's an asterisk problem I htink it's a G711 problem... orgsm problem. Basically they are made for voice, and I think the music goes outside their encoding ranges... sound logical?Rolling off the high-end of the audio range above 4 KHz helps. Try madplay instead of mpg123 and also have the playback gain reduced~12 db with this musiconhold.conf line:default =custom:/usr/local/lib/asterisk/mohmp3/,/usr/local/bin/madplay--mono --sample-rate=8000 --attenuate=-12 --output=raw:- -kim ___ --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] Call waiting?
On Fri, May 27, 2005 9:48 am, Adam Collard wrote: Is call waiting supported on an analog incoming line? I have a customer that has a line with call waiting that wants to go to Asterisk, but wants to keep the call waiting. If it is, how would I set it up in asterisk. We have callwaiting=yes in sections of zapata.conf for both fxo and fxs interfaces. With a call in progress on the pstn, a second call [waiting] results in an immediate disconnect of the first pstn call. On the asterisk end you hear a loud glitch and a burst of caller ID data, then it disconnects. This is with the zapata-bsd drivers, maybe with the linux drivers it would be different. -kim -- [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
[Asterisk-Users] Call waiting on TDM-400 FXO
Is pstn call waiting working on a Digium TDM-400 with FXO ? Configuration in zapata.conf: callwaiting=yes callwaitingcallerid=yes callprogress=yes If an incoming call happens while the FXO channel has a call in progress, and the call is routed to a FXS channel (which has callwaiting=yes in zapata.conf) the call on the FXO is interrupted. Is anyone else seeing this ? -kim -- [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] BSD Compatability
On 5/2/05, trixter http://www.0xdecafbad.com [EMAIL PROTECTED] wrote: On Mon, 2005-05-02 at 00:06 -0400, skamp wrote: asterisk runs great on BSD if you follow the sirections, and the card i believe does work That is debateable :) First which bsd? [snip] Here is the translation table from a dell poweredge 2650/3.06 mhz running fbsd 5.4-stable. g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - -- -- - - - - - gsm - - 2 23 214 823 12 ulaw - 3- 1 3 214 823 12 alaw - 3 1 - 3 214 823 12 g726 - 4 3 3 - 32 5 924 13 adpcm - 3 2 2 3 - 1 4 823 12 slin- 2 1 1 2 1 - 3 722 11 lpc10- 32 2 3 21 - 8 23 12 g729 - 3 2 2 3 2 14 - 2312 speex- 3 2 2 3 2 14 8- 12 ilbc - 3 2 2 3 2 14 8 23- Here is the translation table from a home brew P-4 3.6 GHz running fbsd 5.4-stable: Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 --- ---- - - - - gsm --2 2 2 2 13 - - 13 ulaw -2 - 1 2 2 13 - - 13 alaw -2 1 -2 2 13 - - 13 g726 -2 2 2 -2 13 - - 13 adpcm - 2 2 2 2 -13 - - 13 slin - 1 1 1 1 1- 2 - - 12 lpc10 - 2 22 2 2 1 - - - 13 g729 - - - ---- - - - - speex - -- - - -- - - - - ilbc - 3 33 33 2 4 - - - Lets see how badly the text formatting get mangled ;) -kim ___ 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: RE: [Asterisk-Users] Problems with TDM400P card
On Mon, May 2, 2005 1:12 am, Geoffrey Sachs said: Thanks for the info. What hard drives are you using ide or serial ata. Does it make a difference. Thanks There have been some references recently regarding disk drive types relating to tdm400 noise problems. Has anyone established there is a correlation between drive hardware and noise? If this is the case, it may be indicative of marginal interrupt timng performance on that hardware. FWIW the system described earlier has a single sata drive attached to the Intel 925XCV mb on-board controller, from dmesg: atapci1: Intel ICH6 SATA150 controller The drive is a Seagate ST3250823: ad4: 238475MB ST3250823AS/3.01 -kim -- [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: RE: [Asterisk-Users] Problems with TDM400P card
On Mon, May 2, 2005 8:24 am, Rich Adamson said: To help identify the source of the delays, I built a new system this weekend from scratch. When that is complete, I'll use it to compare the differences in motherboards, OS distro's, and maybe kernel versions. Very good Rich, the results of that work will be very interesting. Realtime scheduling modifications for Linux and FreeBSD are discussed on Mantis at: http://bugs.digium.com/bug_view_page.php?bug_id=0003203 Should you decide to evaluate Asterisk on FreeBSD, you might want to take a look at Staffan Ulfberg's excelllent contribution on the above site, with links to patches near the bottom of the page. The system described in my recent post was built from * CVS Head of 4-22-05 with these patches applied, with the exception of the changes which cause Asterisk to lower it's priority to normal after forking. The zaptel drivers for FreeBSD are from 4-26-05, downloaded from the Subversion repostory as described at: http://www.voip-info.org/wiki-FreeBSD+zaptel Patches to the zaptel drivers are described on the Mantis link above. -kim ___ 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: RE: [Asterisk-Users] Problems with TDM400P card -correction to last post
On Mon, May 2, 2005 9:01 am, Kim Culhan said: Patches to the zaptel drivers are described on the Mantis link above. El wrongo kimster, they're described in this post to the asterisk-bsd list: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000719.html The patches are in this post: http://lists.digium.com/pipermail/asterisk-bsd/2005-March/000722.html -kc ___ 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: RE: [Asterisk-Users] Problems with TDM400P card
On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said: I would also be interested in alternatives to the Tdm400p. I have had endless problems with a tdm400p card not being able to get the zttest numbers above 99.975 and as a result not being able eliminate an intermitent but consistent echo. I have tried to date 4 different motherboard and hardware combinations as well as different linux versions to no avial.I would welcome some feedback on this. Since there appear to be several combinations of hardware and operating system which don't work well, here is a combination which appears to work fairly well: Intel 925XCV mb P-4 560 (3.6 gHz) wcfxs0: Wildcard TDM400P REV E/F FreeBSD 5.4-STABLE zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% ^C --- Results after 10 passes --- Best: 100.00 -- Worst: 100.00 -- Average: 100.00 hope this helps -kim -- [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: RE: [Asterisk-Users] Problems with TDM400P card
On Sat, April 30, 2005 10:52 am, Rich Adamson said: On Fri, April 29, 2005 4:58 pm, GEOFFREY SACHS said: I would also be interested in alternatives to the Tdm400p. I have had endless problems with a tdm400p card not being able to get the zttest numbers above 99.975 and as a result not being able eliminate an intermitent but consistent echo. Kim, that is helpful. I'm not a FreeBSD user, but does it have a vmstat utility? Has vmstat, you might like FreeBSD.. :) If so, what do see if you run 'vmstat 1' and let it run for about twenty seconds? Do you see the cpu utilization going to about 100% every five or six seconds? Negative: vmstat 1 procs memory page disk faultscpu r b w avm freflt re pi po fr sr ad4 in sy cs us sy id 1 2 0 61684 9662607 0 0 0 6 0 0 1326 392 482 0 0 99 0 2 0 61684 9662601 0 0 0 1 0 0 1337 501 494 0 1 99 0 2 0 61684 9662600 0 0 0 0 0 0 1345 486 490 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 6 1350 492 509 0 2 98 0 2 0 61684 9662600 0 0 0 0 0 0 1344 488 490 1 0 99 0 2 0 61684 9662600 0 0 0 0 0 0 1344 492 489 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1345 494 488 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 492 493 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 488 490 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 492 490 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1345 486 487 0 1 99 0 2 0 61684 9662600 0 0 0 0 0 0 1344 513 494 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 494 494 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1345 492 492 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 486 487 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 492 490 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 496 491 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1345 492 491 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 1 1345 486 491 0 0 100 0 2 0 61684 9662600 0 0 0 0 0 0 1344 492 490 0 0 100 ^C -kim -- [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
[Asterisk-Users] Dialout handler with/without leading 1
If this handles the case where 10 digits are required: exten = _9NX,1,StripMSD,1 exten = _NX,2,Dial,Zap/4/BYEXTENSION How do you create a handler which works for either this or the case with a leading '1' plus 10 digits? tnx -kim -- [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
