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Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

Hi all, I am Sahana. I am currently working on Libtheora for windows platform and very new to it. I am getting a lot of compiling errors. Since Libtheora depends on libogg and libvorbis i have added these as well to my workspace in VC++ 6. I have a doubt regarding the x86 and x86_vc folders described in the libtheora software.Since both these folders have the same set of files which one do we

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

...when i ''ve a large amount of data i run out of memory. File.open("#{RAILS_ROOT}/public/files/Neighborsville.TXT").each() do |line| @stringArray = line.split("|") @i += 1 puts @i @pid = @stringArray[0] @chain_id = @stringArray[1] @business = Business.find_by_pid_and_chain_id(@pid,@chain_id); #Check PID + CHAIN_ID @business.pid = @stringArray[0] @business.chain_id = @stringArray[1] @business.cityname = @stringArray[17] @business.state = @stringArray[18] @business.business = Business.find_by_pid_and_...

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

What we would like to see happen or emulated is that if someone makes a call via our SIP provider to a PSTN number that is actually busy that we get an actual BUSY tone at the telephone. In our test case this is a PAP2-NA SIP device It would appear that when we call the far end (PSTN phone number) that is busy we do not get any busy indication at the user end (originating telephone on our

Hi, Since upgrading to 2.4.14 with the ext3 patch, I've been getting the following on bootup: VFS: Mounted root (ext3 filesystem) readonly. change_root: old root has d_count=5 Trying to unmount old root ... invalidate: busy buffer invalidate: busy buffer invalidate: busy buffer invalidate: busy buffer invalidate: busy buffer okay Is the invalidate busy buffer bit anything to worry about?

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

Hi List, I'm using VoIPJet and NuFone as a fallback, and it seems that both of them are circuit busy! Also it seems that VoIPJet takes forever to return 'circuit busy' while NuFone does it instantly. At any rate, is there like a reliable third VoIP provider I can use for fallback when the two others are busy? Cheers, Jean-Michel.

I am having an issue of deciding where I should put the business code for example the more complex code coming off the controller. Right now, I am just thinking to put it in the ''lib'' directory and the do something along these lines: require_dependency "system_process" class SystemMessageController < ApplicationController i...

Hi! I am having difficultly in having users of various SIP devices obtain the correct behaviour when they call a busy number ie. only hearing the Congestion/Busy tone. I assume this might be because the SIP device itself generates the 'ring' tone? With my current setup in the dialplan extract (below) the user of the SIP device hears one 'ring' and then the busy tone if a number

When using a PRI on my asterisk system, and when calling a busy number I often get an "All Circuits Are Busy" message. I've trouble shot it and made absolutely sure all circuits are not busy... but the number being called is in fact the issue. How do I get Asterisk to give me a busy signal instead of "All Circuits Are Busy" when calling a busy number? Is this something

I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-9999) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any

Hello everybody, I would like to implement "callback" function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am

Hello, we've had an Asterisk solution working for quite awhile. Today our IAX2 carrier started reporting circuit-busy to all calls for approximately an hour, which is different than a busy condition. I'm guessing that this indicates the carrier was having an outage, or was handling too many concurrent IAX2 calls. Is there a way to change our dialplan to fail to PSTN in case Dial(*)

Using 1.4 svn, I want to implent the busy application. With the following dialplan: [inboundqueue] exten => _X.,1,Answer() exten => _X.,n,Goto(dropcall,1) ... exten => dropcall,1,Busy(10) exten => dropcall,n,hangup() If I call any number in the inboundqueue, I get the following: [Oct 1 12:06:44] -- Executing [444678 at isdnspan1:1] Answer("Zap/1-1", "")

We all know you can monitor multiple devices in one hint, and it shows busy if any device in the group is busy. This is good for a user with multiple devices, but not useful for teams where any person could take a call, like a customer service group. Does anyone know if it's possible to have a hint with multiple devices which only shows busy if every device is busy? -- Carlos Alvarez

Hello, I use google docs' Forms to conduct surveys online. Multiple choices questions are coded as comma separated values. For example, if the question is like: 1. What magazines do you currently subscribe to? (you can choose multiple choices) 1) Fast Company 2) Havard Business Review 3) Business Week 4) The Economist And if the subject chose 1) and 3), the data is coded as a cell in a spreadsheet as, "Fast Company, Business Week" I read the data with read.csv into R. To analyze the data, I have to change that string into something like flags(indicator variab...

Hello, On a Asterisk 1.6.1 powered system, I've just discovered that using Busy() application in dialplan was no enough to send a Busy signal on incoming Dahdi channel. On this specific install, adding an Answer()) and a Playtone() statement in dialplan triggered sending of busy tone but I'm still surprised by my findings. Should I expect public switch to send a Busy tone to caller