similar to: Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-)

Easy questions for you guys probably, I'd like to serve 10 parallell incoming calls at the same time, so I bought a lot of Zap-channel cards for analog phone lines. But I want all users to be able to use the same phone number to dial in, but I want the number to be switched to an avaiable zap-channel. Do I need some kind of switch for this? It sounds reasonable, but I'm not sure. :) Am

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

Hi, I'm using SIP MESSAGE to asterisk V10 and it fails to be received. I'm not super sure of the reason but I'm making this guess: Due to I'm using non ipaddress in the to field, which contains sip:mobil1.xyz.com, Asterisk makes the mistake to try matching this name "mobil1.testserver.com" in extensions.conf and no extension/peer is found in the sip-message context

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile

I see there are still some floating point operations left in the codec init(ialization) code. Changing that code to fixed point is not only difficult (due to the trigonometric functions etc) but may also degrade the precision. Here is an idea whereby we can easily precompute (record) all those values on a powerful processor and then use (replay) them on an embedded processor / DSP. The only

Hi I have a question about the AEC and preprocessor. I have seen that when near end is talking for a longer time (about 10 s) without being interrupted by the far end the residual echo (or rather leak_estimate) increases making the preprocessor suppress the near end talk when it shouldn't. Is there a way to make the leakage estimate only update when far-end is present or similar in order to

part of extensions.conf: exten => 11,1,Answer() exten => 11,n,NoOp(CallerID : ${CALLERID(all)}) exten => 11,n,Playback(/tmp/welkom-tcs.alaw) exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1) ; wordt doorgerouteerd naar context open, maar indien gesloten : exten => 11,n,NoOp(Oproep tijdens winkel gesloten) exten => 11,n,Playback(/tmp/winkel-gesloten.alaw) exten =>

Hello What soundfile format, is the one that uses least transcoding during playback? As I can see, I can choose wav or gsm. What sucks least cpu power, during playback to example a Zap channel? I would guess wav, but is this correct? Kind Regards Jon Leren Sch?pzinsky -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.408 / Virus Database: 268.13.11/496 -

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

Hello list, is it possible with the monitor-command to record conversations and place the soundfile in a pre-defined directory per user ?! So when extension 200 presses '*#' to record the conversation, the resulting sound file is written to his home directory on the Samba-server. This way each user has his own directory with its recordings that no one else can access (as default rights

I'm running a small Linux server with samba installed on it I want to access this server from an XP client which is a member of a ad-domain but when trying to logon XP putts in ad-domain-name\username as logon name to the samba server, how can I work around this?

Hi Guys, I want to playback a sound file stored in mysql database in my perl script............pls can anyone help with an idea? response would be greatly appreciated Rgds _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

Hello. I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

Hello, I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to CVS-HEAD, and realtime. Compiled no problem and now running, with realtime extensions and sip users in postgres (ODBC connection) database, trunking also works. I have looked on google, wiki, and this mailing list, along with talking to some peers, but to no avail. My problem revolves around voicemail. I have looked

Hello, Which port Smb4K uses? I am not able to use Smb4K on my server for the local network. Thanks Varun

Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while