similar to: chan_zap "Cannot handle frames in 2 format"

I've just finished updating my Asterisk to the CVS version. Unfortunately, chan_zap won't load anymore. The hardware has not changed and the config files have not changed. I can re-install the two packages back and forth. The old one will still work. The new one won't. I tried updating to a brand-new zaptel and wcfxo modules, with no difference. This has got to be the most

Any ideas on this? This has always worked in the past until we got this new customer and his new fax machine. Faxing from my machine works fine. But faxing from his gives this error. I can't imagine why his fax machine would be causing frame errors. -- Redirecting Zap/4-1 to fax extension -- Executing Macro("Zap/4-1", "fax-handle|7133950500|2815692740|Y") in

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

Hello everyone, In working with Asterisk on FPU starved hardware like the Soekris Net4801, it has become clear that it would be very nice to have the standard Allison sounds that are distributed with Asterisk in multiple formats. With AstLinux taking 27mb, that leaves a lot of space on a 256mb CF card for sound files! However, the last thing that I want to do is convert from one lossy

Thanks. sox was exactly what I was looking for. -----Original Message----- From: volsung@asu.edu [mailto:volsung@asu.edu] Sent: Thursday, February 08, 2001 2:06 PM To: vorbis-dev@xiph.org Subject: Re: [vorbis-dev] Conversion API for computer telephony systems (Dialogic Mu-law wa v format to PCM encoded wav format) On Thu, 8 Feb 2001, Giovanni Sanfelici wrote: > I am working on a project

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

I am working on a project involving the conversion of a Mu-law sound format (Dialogic Mu-law wav format) file into a standard PCM encoded wav file format. Could somebody tell me if this feature is supported in the Vorbis software and if there is any source code available that performs this task. If not, does anybody know of any resources that might provide me with this tools or information.

I'm using Asterisk to do audio as well as H.263 video over SIP. Actually the video works pretty well but I have trouble with the audio. I'm wondering if someone can suggest codec/jitter settings or other tweaks. The system looks like this: Softphone <---ulaw----> Asterisk #1 <------IAX (usually GSM)---------> Asterisk #2 <-------- IAX (usually GSM) -------->

Hi all, I am running asterisk 1.2.18, zaptel 1.2.18, libpri 1.2.4. on a suse 10.2, running kernel 2.6.18.2-34-default. The zaptel drivers are loaded on boot via /etc/init.d/zaptel, but Asterisk is unable to start. It ends with the following message: [chan_zap.so] => (Zapata Telephony w/PRI) Jun 13 13:05:41 VERBOSE[3356] logger.c: == Parsing '/etc/asterisk/zapata.conf': Found

Hi all. chan_zap not compiling, yes yes I know this sounds trivial but here me out... This morning I decided to upgrade to Asterisk 1.4.2 and Zaptel 1.4.1. I successfully installed zaptel 1.4.1 and the card is picked up and correctly configured. ztcfg shows the following: [root@zion root]# ztcfg -vv Zaptel Version: 1.4.1 Echo Canceller: MG2 Configuration ====================== Channel

Suddenly after recompiling my 2.4.22 kernel I can no longer load chan_zap: Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 5145 (mkintf): Unable to get span status: Inappropriate ioctl for device Sep 22 21:25:08 ERROR[16384]: File chan_zap.c, Line 6638 (load_module): Unable to register channel '1' Sep 22 21:25:08 WARNING[16384]: File loader.c, Line 301 (ast_load_resource):

Hi friends, Where can I get some information to understand messages like the following ones? *NOTICE[6455] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1* *NOTICE[6455] chan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1* * ERROR[6455] chan_zap.c: !! Got S-frame while link down* *ERROR[6458] chan_zap.c: !! Got reject for frame 27, retransmitting

Trying to use: Asterisk 1.4.2 Zaptel 1.4.0 chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap

It's a display bug in PuTTY+Epsilon, but a subtle one! I upgraded PuTTY from 0.65 on one PC to 0.73 (latest) and the problem disappeared. I still see it on another PC running PuTTY 0.70. I'm still trying to characterize it but the output is correct and it's because I'm using Epsilon to view the output file that I'm not seeing the missing characters. The issue is related

Hello, in order to vectorize task (of plotting barplots), I use, as probably all R-folks, function apply() instead of for() loop. <R> testarr<-matrix(1:30, nrow=5) rownames(testarr)<-letters[1:5] colnames(testarr)<-LETTERS[1:6] apply(testarr,1,function(x) {x11(); barplot(x)}) #the funcion used is actually more compex, but it doesn't matter </R> Now, how I can propagate

Hi, I've a little problem with my asterisk server. I have managed an asterisk server for a few months one. Today, I wanted to restart it and when I did it, my asterisk server didn't want to start again. I looked at the various messages in /var/log and the same thing appears. " [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found Sep 30

What am I doing wrong here? Why is this happening? libpri is version 1.0.7-1 (debian package) asterisk is version 1.0.7.dfsg.1-2 (debian package) zaptel is version 1.0.9.2 -- Executing Dial("SIP/739-5935", "Zap/g1/0916000739") in new stack -- Called g1/0916000739 -- Channel 0/1, span 1 got hangup Oct 10 13:14:45 WARNING[7544]: app_dial.c:412 wait_for_answer:

Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'

Hi, I have Asterisk 1.4.0 using Queue App. I use PRI connecting my Asterisk with Siemens EWSD This was working OK but since two days I have this error: [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22 10:53:09] WARNING[8050] chan_zap.c: Ring requested on channel 0/3 already in use on span 1. Hanging up owner. [Jun 22

Hello I've been googling for this message, but can't find why Asterisk sends a warning. The configuration files look similar to http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample It's a TDM card with just one FXO module on it, and I connected an RJ11 cable to it from the wall plug: # cat /etc/zaptel.conf fxsks=1 loadzone=fr defaultzone=fr # ztcfg -vv