similar to: How to get asterisk to take a dump?

Hi All; I do not know if anyone faced such case in dealing with open vpn (as we need it for fring to be used from the mobile: Which mobile can be used to install the open vpn client on it, so we can use it to do a vpn channel with the server that has open vpn server? Regards Bilal

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good

Hi, My Asterisk server process irregularly segfaults, ie. it usually works fine (is stable) when there's low traffic but repeatedly crashes during morning hours when there are more calls. I gdb'ed the core dump files and found that the culprit may be format_mp3. So I disabled MOH today and will see if that's the cause. I know that mp3 files are known to cause * crashes but what I

Hi, Am writing scripts to manage configuration management and Asterisk. I would like to be able to point the asterisk binary at config directory with an asterisk.conf in it, and for asterisk to run a pre- flight check. A bit like a pint check in php, 'apachectl configtest' and lots of other tools. asterisk will then exit with 0 on a safe config, and 1 on a bad config. I can

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hi i have problem with AddQueueMember logic. I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) and now i want to call to this Agent: exten => _1XX,1,Dial(Agent/${EXTEN:1}) call to 113 and asterisk should call to Agent => 13 on interface SIP/ekiga. This doesn't work, How can i do this on Asterisk 1.4(not

My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I

Hi guys I've just read this about the upcoming release of * 1.6: ?Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing.? That

Hi all, I found that there will be a memory leak if asterisk running day by day without restart. Is it good to restart asterisk service daily? What is the better way to restart it daily like apache? ango

Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

I create my sip users using a common macro in 1.4: [internal] exten => 200,1,Macro(phones|200|SIP/200) [macro-phones] exten => s,1,Dial(${ARG2}|45|Tt) etc... But now in 1.6 this fails: -- Executing [200 at handsets:1] Macro("SIP/201-0942b530", "phones|200|SIP/200") in new stack [Nov 20 08:55:55] WARNING[5958]: app_macro.c:201 _macro_exec: No such context

Hi all! I've installed a new Asterisk 1.6.0.1 with addons and dahdi drivers and tools but my calls aren't logged. I'd enabled mysql log and noticed that asterisk send a 'DESC cdr' so connection is working between asterisk and mysql and I am able to call other phones so Asterisk is working as well. No error messages on startup though. Any idea why is it happen? As I realized

Hi, I'm having a question with asterisk queue system, is it a fifo or a lifo or random ? Sometimes when we have people waiting in the queue and new agents are connected to handle the load the first call that is handled is not the one which is already waiting for 4min, but the new one which has just arrived. However this doesn't happens everytimes Is it normal ? regards, benoit

Dear all, I'm trying to configure Asterisk to be able to ask the caller to enter a given password in order to continue dialplan execution. I've tested this feature using the Authenticate application like this: exten => _X./5219,1,Answer exten => _X./5219,2,Authenticate(1234,a) exten => _X./5219,3,Playback(pin-number-accepted) exten => _X./5219,4,Dial(SIP/${EXTEN},120)

Hi all, I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070924/b2eeeca9/attachment.htm