similar to: no sound. "Failed to write frame" (2nd post)

Hi all, Couldn't find a place to search the list archives... I'm having issues in getting any sound using a fresh asterisk install and a SJPhone to connect to it. I went by the instructions pointed at the "10 minute guide", located here: http://www.voip-info.org/tiki-index.php?page=Asterisk+quickstart. I installed it on a Slackware 10.1, by using no more than "make

Hey all, is there a way to wildcard field searches? As in: - a document like {:title => ''foo'', :description1 => ''bar'', :description2 => ''bar2''} I''d search: index.search("description*: search query") I understand the example above is silly, but it''s enough to make the question understandable :-)

Does anyone knows a working FOSS rating application for Asterisk? I tried CDRTool, but since it's free for non commercial use, it won't suit me. Also I gave a shot at Trabas (the one everybody says it sucks), but I couldn't understand how does it pull records from Asterisk's CDR table. The rate-engine addon (http://www.voip-info.org/tiki-index.php?page=Asterisk+addon+rate-engine)

Hey all, I''m distributing requests to a small farm of Ferret servers across the network using DRb. In a specific part of my program, I''m trying to find an entry across servers, and for that, I''m using index[''example_doc_id''].load as the return value of the function in question. This returns a Ferret::Index::LazyDoc, which is all fine and dandy,

Hi all, I''ve been doing some work on the Ajax Scaffold (http://www.ajaxscaffold.com/), to make it work with file uploads (more specifically, the file_upload plugin). The first logical step was to get rid of form_remote_tag, since it''s impossible to send files via XMLHTTPRequests (to my best knowledge). So I referred to the iframe hack, similar to the upload progress bar here

Hey all, I went through the docs in Ferret''s page, plus a quick search through the email list (thread titles), and I couldn''t find any info on how to have Ferret storing it''s data using UTF-8. In the scenario I would use it, nothing''s being stored outside (like external databases). So it''s just how Ferret would do it that I''m interesting in

Hi all, maybe not a Ferret question, but I assume here might have came across that already. I wrote a simple CGI app that adds docs into a Ferret index. The idea is testing asian languages input and searching. The script that does the input seems to be OK. As David mentioned in a question I made a little while ago, Ferret''s index is agnostic, in the sense that you can store anything in

Hi list, I have been looking around for a while now, and cant seem to get to the bottom of my problem. My setup is that I have a separate SIP server that servers my SIP subscribers, and I want to use Asterisk purely for voicemail for now. So I set up a common SIP extension at my SIP server, and made Asterisk register it, so that normal users can forward calls to that common extension, and

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

Hey all, I''m getting some really weird results when searching documents. It *seems* to be somehow related to the document format I''m using. I wrote a small script to replicate it: ################ #!/usr/bin/ruby require ''rubygems'' require ''ferret'' include Ferret index = Index::Index.new(:path => ''/tmp/fooindex'', :key

All, After installing asterisk on Linux, I run "asterisk -vvvc". But I got the following warning message: chan_oss.so] => (OSS Console Channel Driver) May 26 00:37:58 WARNING[-1084845952]: chan_oss.c:980 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing

Hi all Asterisk 1.8.11.0 on Centos 6.5 My VOIP phones are using G729 to a G729 trunk from a vendor (Centracom, South Africa). Unlicensed G729 codec version on server. 75% of my .gsm files from MixMonitor are coming up corrupted about 3 minutes into the recording. The server has been up for 7 months beforehand with no problems with recordings to .gsm format files. I noted

Hi I have a cisco 7960 Phone that connects to my Asterisk server without a problem. But when I call the echotest it just hangs up, echotests from other VoIP providers works just fine. I have tried a softphone and it works just fine. The error I get when the 7960 calls is this: -- Executing Playback("SIP/2000-180c", "demo-echotest") in new stack -- Playing

hi i'm a newbie in asterisk...i installed asterisk but when i tried to dial 1000 for the first time i got the following error messages and i don't hear anything... May 29 20:46:03 WARNING[262160]: chan_oss.c:413 soundcard_setinput: Unable to re-open DSP device: Device or resource busy May 29 20:46:03 WARNING[262160]: chan_oss.c:572 oss_write: Unable to set device to input mode May 29

After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very strange problem. There is no sound with Playback() or Background() commands. Even though, Asterisk console shows the file is being played when I call the extension (i.e. echo test), I can't hear anything. My echo test extension looks like this: exten => 600,1,Answer exten => 600,2,Playback(demo-echotest) exten

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile:

Hi, I've been messing with getting SIP working for days now, with limited success. I've got Asterisk set up on a remote server with the echo test. Please try it out to verify I've got the server working right: sip:robot@nixon.butchwax.com Running FC1, ThinkPad T22, headset thru the soundcard. Asterisk is asterisk-1.0_RC1. No NAT. The phones I've tried so far are as

I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and