search for: vvvgc

hello all, i'm using TE110P and all configuration prosess is done. but after i do the 'modprobe' for zaptel and wcte11xp, follow by ztcfg -vv and asterisk -vvvgc i have this error. *CLI> May 27 14:44:33 WARNING[3059]: chan_zap.c:8534 pri_dchannel: PRI Error: We think we're the network, but they think they're the network, too. May 27 14:44:34 WARNING[3059]: chan_zap.c:8534 pri_dchannel: PRI Error: We think we're the network, but they think th...

I'm running Redhat version 9 with a TDM400P and AVM B4 ISDN2 card. I am assuming the first step to get * working is to get a tone from the fxo ports, having looked and played with both the zapata and zaptel files, I still can't get anything.on the ports. When starting asterisk -vvvgc it does not appear to be loading Zapata file. Any guidance for a newbie would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030604/908528d3/attachment.htm

...arge) with the Asterisk 1.0.5 package. 2) Redhat 9 with Asterisk CVS 1.0.7+ I can't get the AGI "STREAM FILE" command to work with a simple bash script. I can get other AGI commands to work like "SAY NUMBER 123" etc. I've set AGI DEBUG and have started Asterisk with -vvvgc. No error messages, returns 0. Just doesn't play anything. Any help would be greatly appreciated. Bill

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try again... Cannot allocate channels... Process Stopped! Error -10 WARNING[1074420448]: File codec_g729b.c, Line 511 (load_module): Unable to i...

Hi, I'm running asterisk-1.0.2-2mdk. When I tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe # ps aux | grep mpg123 root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3...

...n Digium Wildcard and a single X100m FXO interface connected to a POTS analog line. Build and install of both work ok - I'm using Suse 8 on a dual Pentium box. I load the driver with "modprobe wctdm" and the LED on the wildcard lights up. Then I start Asterisk with "asterisk -vvvgc" and asterisk fails to start. The messages look pretty good till the last few lines.. . . . Aug 11 08:29:42 WARNING[6888]: chan_zap.c:9651 setup_zap: Ignoring switchtype Aug 11 08:29:42 WARNING[6888]: chan_zap.c:778 zt_open: Unable to specify channel 1: No such device or address Aug 11 08:29:...

I've been trying to get Asterisk 1.4.X running under Solaris 10 x86 with limited success. I can build Asterisk and get it started but have run in to a problem with a segmentation fault with the "help" command in the CLI. When I start Asterisk: # ./asterisk -vvvgc Asterisk 1.4.9, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License...

...fg, and it's reporting that 1 FXO channel and 1 FXS channel have been configured. I've set up in my zapata.conf file two channels: one for the analog phone (fxo_ks), and one for the POTS line (fxs_ks) that is coming into the office. The problem is when I test my setup with "asterisk -vvvgc", asterisk pukes and reports the following: [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found May 13 17:19:49 DEBUG[-1220444032]: chan_zap.c:1077 update_conf: Updated conferencing on 1, with 0 conference users -- Registered channel 1, FXO Kewls...

start it with asterisk -vvvgc bkw ----- Original Message ----- From: "Arjan" <arjan@inventionz.org> To: <asterisk-users@lists.digium.com> Sent: Sunday, July 11, 2004 12:27 PM Subject: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies > Hi All, > > I'm pretty green to Asterisk. I'm trying...

...ts and pieces are here, and I've got the Digium hardware installed and (I think) loaded properly (RedHat 9, Compaq 1850R, T100P, TDM400P 2 port FXS). I'm still waiting on my Cisco 7940/7960 handsets (ordered with SIP). CVS'ed down all the components, sample configs, and "asterisk -vvvgc" starts clean. The only change made so far is to noload chan_oss. Anyway, I'd like to get comfortable with * prior to replacing our Siemens Hicom system. The PSTN is PRI for BellSouth, and we're going to use SIP phones internally. Now for the stupid questions.... 1) How do I figure...

When run asterisk ?vvvgc IT show me this error Asterisk Ready. *CLI> Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'phone@192.168.0.6' timed out, trying again Apr 11 08:59:27 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration from '<sip:phone@192.168.0.6&gt...

...: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers. ---------------------------------------------------- Error in asterisk -vvvgc -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1", "Zap/g2/088008") in new stack -- Called g2/088008 -- Zap/4-1 answered Zap/1-1 -- Attempting native...

So far i've grasped that to use a card in NT mode it should have layermask=3 as module option. Is it the only thing that sets TE or NT mode for card? Perhaps there are settings in misdn.conf ? I can only get the card to work in TE mode and even then when asterisk is ran as asterisk -vvvgc it exits right after chan_misdn is loaded with theese messages: [chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri)) == Parsing '/etc/asterisk/misdn.conf': Found UnLocking config_mutex == Registered channel type 'mISDN' (This driver enables the asterisk to use ha...

Hi. I use Asterisk-CVS HEAD version. When I type a command 'restart gracefully', then asterisk just stop. Messages are : [root ]# asterisk -vvvgc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) >>>>>> snip messages. Asterisk Ready. *CLI> restart gracefully Waiting for inactivity to perform restart... Executing last minute...

...el 16: D-channel (Default) (Slaves: 16) Channel 17: Individual Clear channel (Default) (Slaves: 17) . . . Channel 31: Individual Clear channel (Default) (Slaves: 31) 31 channels configured. ZT_CHANCONFIG failed on channel 25: No such device or address (6) And when I lunch this command asterisk -vvvgc I got these: . . . -- Registered channel 23, PRI Signalling signalling Jun 14 13:50:58 ERROR[6493]: chan_zap.c:6710 mkintf: Channel 24 is reserved for D-channel. Jun 14 13:50:58 ERROR[6493]: chan_zap.c:9910 setup_zap: Unable to register channel '1-15' Jun 14 13:50:58 WARNING[6493]: load...

...files in the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 then copying the chan_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 After building and installing asterisk When I run 'asterisk -vvvgc' command it doesnot find dtmf_put during linking time. So, I commented the dtmf_put function in chan_unicall.c. Than every thing worked fine... Can any body tell me about how to configure unicall.conf for incoming and outgoing call As asterisk is running on one end and E1 card runni...

...n the follwing sequence spandsp libsupertone libunicall libmfcr2-0.0.3 then copying the chan_unicall.c and channels_Makefile.patch to channels subdirectory of asterisk-1.2.24 After building and installing asterisk When I run 'asterisk -vvvgc' command it doesnot find dtmf_put during linking time. So, I commented the dtmf_put function in chan_unicall.c. Than every thing worked fine... Can any body tell me about how to configure unicall.conf for incoming and outgoing call As asterisk is running on one end and E1 card...

...here a way to adjust the tone detection, make it more sensitive? Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers. ---------------------------------------------------- Error in asterisk -vvvgc -- Starting simple switch on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1", "Zap/g2/088008") in new stack -- Called g2/088008 -- Zap/4-1 answered Zap/1-1 -- Attempting native...

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from CVS, Asterisk crashes on startup with an apparent MySQL (res_config_register) error: # asterisk -vvvgc > asterisk_startup_error1.log asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: un defined symbol: ast_cust_config_register The log is shown below. I've seen the posts from 1/25/05 and several more recent ones regarding this same issue or a similar...