similar to: SetAccount in extensions.conf

> I'm using Asterisk 1.4 and I'm wanting to set an > account code for incoming calls. In the > extensions.conf file I have the following: > > exten => s,1,SetAccount(1234) > exten => s,n,Dial(SIP/1234) > > Then when I dial the extension the following error > message pops up in the CLI: > > [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 >

Hi Randy, One thing I would note is that speex is designed for 8kHz, 16kHz or 32kHz. 160 samples is equal to 20ms of 8kHz audio. Have you tried resampling from 11.025kHz to 8kHz and then using the speex 8kHz mode? (nb, or narrow band). Or, if you want to preserve the higher quality of your 11.025kHz sample rate, resample to 16kHz and use the wideband speex encoder. I believe there is new

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hi Jean-Marc, Just some initial feedback, I've just tried to build svn head using the Visual Studio 2005 compiler, and had the following issues: 1. Missing definition of M_PI if it's undefined (and it is on this platform) 2. In speex_resampler_process_int, the compliler can't determine the value of *in_len and *out_len at compile time and thus determine the size of the x and y

> I believe there is new resampling functionality in the speex svn head, > although I haven't tested it yet. You might also want to check out > 'Secret Rabbit Code' for your resampling. Yes, I've just been working on a resampler recently. Its changing a lot, but it's now usable. I'd actually be quite happy to have some feedback on it. Jean-Marc > Hope this

Hello all, I gave this a stab a few months ago but gave up and just set up a dual boot. I'm running low on disk space now and tired of messing with the muliple partitions, so I'm giving it another go to get Guild Wars running in Linux. I'm runing Wine 0.9.3 on Kubuntu 6.06 (Dapper Drake). I'm able to go through the instillation without issue, but when it actually tries to

Hi - I am currently developing a RTSP/RTP/SDP solution to stream Speex encoded data. Using my current source, I have successfully streamed u-law and PCM encoded audio but have been unsuccessful thus far with Speex. Because of some constraints of my system, I am encoding audio at 11.025kHz. I am still using the 160 samples per frame which makes my frame size 28 bytes. I have successfully

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

Hi, I have problem with setting up a conferences. When I dial the reserved conference number from xlite the line gets hunged up and on a console there is a following message: WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe' for extension (internal, 1234, 3) exten => 1234,1,Answer() exten => 1234,4,MeetMe(1234|Md) exten => 1234,101,HangUp()

Hi all, i need help to implement a voicemail scenario. What i am trying to do is the following. user X dials a direct access for user Y voicemail and is asked to enter a number (e.g 12345678) and then leaves a message. Then asterisk sends a notification with attachement. The problem is that i need the number entered (e.g 12345678) in the subject. Is that possible. thx in advance.

Hi All, I have an NT 4 domain with multiple samba servers. One of my samba fileservers stopped allowing domain login requests. While it can enumerate the domain users with wbinfo -u, and the domain groups with wbinfo -g, getent passwd does not list the domain users. All the other servers in the domain are fine. Any suggestions for how to track down this error? Sincerely, Donald

Hi Josh (and everyone else on the list), Here's a quick fix for libFLAC.m4 and libFLAC++.m4 for a problem that crops up if you use the macros in an environment where LD_LIBRARY_PATH is set. The macros save LD_LIBRARY_PATH as ac_save_LDPATH, but restore it from ac_save_LD_LIBRARY_PATH. This patch changes ac_save_LDPATH to ac_save_LD_LIBRARY_PATH. Without this, LD_LIBRARY_PATH effectively gets

Folks: The single reference to a Mac version of flac software that I can find seems to be way out of date. In addition, when I DL the file and attempt to install, I am greeting with an impressive list of "missing" software items that I believed, incorrectly, it seems, was suppose to be included in the file that I did DL. Any suggestions?? BTW, I run Mac OS X 10.4.8. Avi P.

Hi all, inside my * is a E400P. The machine is a PII 400Mhz with 256MB Ram. OS is Debian woody. * is the newest cvs co. I have written a little callgen script which make outgoing calls through my *: #! /bin/sh set -e n=$1 # Nummer anz=$2 # Anzhal der Versuche anz2=$3 # Kan?le sle=$4 # Timeout bis zum n?chsten Versuch if [ -z $4 ]; then sle=0 fi s=1

2023-11-08 18:14:13] ERROR[571246][C-000017e2] : Got 19 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() # 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref() # 2: [0x58e660] asterisk stasis_cache.c:824 update_create() # 3: [0x58efed] asterisk stasis_cache.c:903 caching_topic_exec() # 4: [0x586b90] asterisk stasis.c:1380 dispatch_message() # 5: [inlined] asterisk

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Does VoicemailMan have to be installed ? Why not available. I have setup a mailbox in voicemail.conf and I can leave a voicemail - just cannot pickup up using *97. My *97 code in extensions.conf: exten => *97,1,Answer exten => *97,2,VoicemailMain(${CALLERIDNUM}@default) exten => *97,3,Hangup asterisk console: Verbosity was 8 and is now 12 -- Executing

Both of our production asterisk servers are dumping core when making writes to our cdr tables. Here is a backtrace of the problems we are having: #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at util.c:347 347 if (tds_ctx && tds_ctx->err_handler) { (gdb) bt #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at

This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on

pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in extension 'PHONE NUMBER' in context 'phones' [Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto: Priority 'outgoing|PHONE NUMBER' must be a number > 0, or valid label PHONE NUMBER = the number I called. This dialplan worked fine in version 1.4. Michael