similar to: * crashed

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

Hello, While upgrading * from 1.0.9 to 1.2.5, I have installed chan-capi-head and I can't start asterisk under asterisk group "asterisk -vvvvgc -U asterisk " and "asterisk -vvvvgc -U asterisk -G dialout" work well but "asterisk -vvvvgc -U asterisk -G asterisk" fail. I am thinking about a group permission configuration but I have exactly the same one

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

Hi, I try to use oh323 package from inaccessnetworks for asterisk, but after make and make install that package, I have this WARNING message hwen a try to launch asterisk from shell command line...asterisk -vvvc... [liboh323wrap.so]WARNING[1024]: File loader.c, Line 235 (ast_load_resource): No load_module in module /usr/lib/asterisk/modules/liboh323wrap.so

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After upgrading the kernel on an Asterisk box, asterisk segfaults on startup. It seems like it's the g729 codec that causes this: #0 0x4015acad in memset () from /lib/libc.so.6 #1 0x4022686a in load_module () at codec_g729b.c:416 #2 0x08054794 in ast_load_resource (resource_name=0x80d1068 "codec_g729b.so") at loader.c:298 #3

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

Hello, list a have a problem i can start asterisk, i get the fowlling error: [chan_capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Feb 11 13:50:36 NOTICE[2535]: chan_capi.c:2636 load_module: CAPI not installed! Feb 11 13:50:36 WARNING[2535]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Feb 11 13:50:36

I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.

Hello, Does anyone know what driver to use with an "Eicon Diva Server 4BRI" card? My modem.conf: -------------- [interfaces] context=remote driver=i4l msn=16453 device => /dev/ttyds01 -------------- I get the following error: -------------------------- == Loading modem driver chan_modem_i4l.so => (ISDN4Linux Emulated Modem Driver) WARNING[8192]: File chan_modem.c, Line 383

Please, try exten => 99,1,Wait,1 exten => 99,2,Record,/tmp/pcmfile:pcm exten => 99,3,Wait,1 exten => 99,4,Playback,/tmp/pcmfile exten => 99,5,Wait,1 exten => 99,6,Record,/tmp/voxfile:vox exten => 99,7,Wait,1 exten => 99,8,Playback,/tmp/voxfile (put your own extension). Pcm recording is OK, playback is OK. Adpcm recording is noticeably worse. Adpcm playback is very

when i issue asterisk -vvv command i get this error please help regards Barbra [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7,

hi just after talking to Jeremy, I downloaded the source he's using and compiled. core dump! This was without gatekeeper: (gdb) bt #0 0x40141118 in mallopt () from /lib/libc.so.6 #1 0x401400fc in malloc () from /lib/libc.so.6 #2 0x405e981b in PAbstractArray::SetSize(int) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x405ed006 in PString::SetSize(int) () from

I have a T1 PRI from the telco and have been trying to get it to work with the wildcard T400P. I know framing and coding is esf and b8zs. what else should I be putting into the zaptel.conf and zapata.conf. Dave Anderson

Hi, I have two WCT100P cards installed on a suse 9.0 box. Installation for Zaptel complains of some unresolved dependencies. The zaptel and wct1xxp modules load without any errors. ztcfg give no problems. The problem is when I start asterisk I get the following error and asterisk shuts down

it just came to my mind, and i haven't done any researches yet if somebody tried this one with asterisk.. :) well just in case somebody or someone on the list aware, i appreciate any advise. in telco world, there's like 64kbps per channel and voice can be carried on a 16kbps channel. is it possible to configure asterisk to make 4 extensions (ATAs example), to call out using single FXO

can somebody that has these hardwares(X101P and S400P) working on his asterisk system please assist.................. you can send the solution to austino@skannet.com..... the error message is what i have below. ---------- Forwarded message ---------- Date: Thu, 1 May 2003 21:01:21 +0100 (WAT) From: austino@skannet.com To: asterisk-user@lists.digium.com Subject: Error working with X101P and

Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel => 1-15,17-31 ;group=2 channel =>32-46,48-62 ;group=3 channel => 63-77,79-93 ;group=4 channel => 94-108,110-124 And here my zaptel.conf: zaptel.conf [----] 0 L:[ 1+ 0

Hello, can i use different audio codecs when i calling between sip devices ( snom phones ) and different when i making call from isdn to sip or from sip to isdn ? best regards Marian -- SUNTEQ s. r. o. Bojnicka cesta 35 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A