similar to: Unable to find a path

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

Hi! I have a strange problem. In an AGI I tell Asterisk to playback a number, for example 31. I then use the AGI SAY NUMBER command and I only hear "thirty" and then get: -- Playing 'digits/30' (language 'de') Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist in any format Sep 29 11:47:40 WARNING[3401]: file.c:787 ast_streamfile: Unable

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

Hello All, I downloaded the ChanSpy patch from Mantis and updated to the latest asterisk source from cvs. Everything seems to have installed fine and everything works as it had before, but I can't get ChanSpy to work. I added a line to extensions, as a test: exten => *53,1,ChanSpy(scan) When I dial this extension from a SIP phone, and then make a call (which I am trying to monitor) from

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

Dear all, I have spent some time tying to get the system attendant patch to work; http://bugs.digium.com/bug_view_page.php?bug_id=0000214 I get no errors patching the system and the function runs, but I keep getting the following error; queue: Nexus1, options: (null), url: (null), announce: (null), timeout: 0 -- Started music on hold, class 'default', on SIP/phone10-a3f0 --

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Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

Any chance of say digits being extended to recognise "*" & "# " ?? Heck these are digits on a normal keypad :-) Gary .

> Well that didn't work....I now get this error > > > Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to > create > channel of type 'SIP' > == Everyone is busy/congested at this time > -- Executing VoiceMail("IAX2/iaxfwd@65.39.205.121:4569/5", "b") in > new > stackJan 12 16:56:21 WARNING[4989]:

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was relesed. now I am having troubles with my dialing plan and voice mail. As part of the upgrade I re-built the machine so there was a blank slate however after installing 0.7.1 I had no mail box creation script and could not figure out how to go about creating a mailbox, any suggestions would be usefull. I have looked at

Hi *, I have set up a very simple asterisk configuration where I intend to be redirected to the voicemail whenever I dial 100 with my kphone SIP client. The problem is that asterisk can not find the stream 'vm-theperson'. I have made a non-standard installation (since I am just testing), and that file is located in /mnt/tr2/fake_root/installed/usr/local/var/lib/asterisk/sounds. 1. How

Hi, I have a problem that I just can't figure out how to solve. I start *, dial it using a ISDN phone over PSTM, to a Hisax card installed in * I get the demo-greeting, listen for a few seconds and hang up. I dial it again, but this time the first second is sound from where the previous call ended, then the greeting starts as it should. Right now I have removed all codecs but codec_gsm.so

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-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, I'm now the proud owner of an X100P and am struggling to set up a CVS-compiled Asterisk to do my bidding. I checked zaptel/zapata/asterisk out today and pretty much did a straight make install on all packages. So far the only consistent trick I can make it perform is calling from one SIP phone to another. Could I get a bit of

Specs: Fedora Core 3. Dual P3 600 (Dell PEdge 1300) SCSI Disks 1x X100P (channel 1) 1x TDM20 (channels 2+3) 1x Knockoff X100P (channel 4) I am looking to have all local and all toll free calls go outbound through the Copper line, and all long-distance and international to go out through the Vonage line. This way I can eliminate LD on my home line, and pay minimal LD charges through

I think there is a memory bug in `substr` that is triggered by a UTF-8 corner case: an incomplete UTF-8 byte sequence at the end of a string.? With a valgrind level 2 instrumented build of R-devel I get: > string <- "abc\xEE"??? # \xEE indicates the start of a 3 byte UTF-8 sequence > Encoding(string) <- "UTF-8" > substr(string, 1, 10) ==15375== Invalid read of

When I pin is being matched to ASTCC database I get this message and the call is dropeed. Oct 2 15:22:20 WARNING[327699]: file.c:475 ast_openstream: File your does not exist in any format Oct 2 15:22:20 WARNING[327699]: res_agi.c:435 handle_streamfile: Unable to open your == Spawn extension (from-sip, 77, 2) exited non-zero on 'SIP/2000-4d5b'