similar to: Play sound on hangup

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

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

Hi, I'm wondering why zttest shows Best: 99.976 -- Worst: 99.967 -- Average: 99.971469, Difference: 99.971469 Shouldn't it be 100% as timing is hardware and comes from PRI? Am I missing some kernel config? Regards, Atis My /etc/zaptel.conf is span=1,4,0,esf,b8zs span=2,3,0,esf,b8zs span=3,2,0,esf,b8zs span=4,1,0,esf,b8zs #lspci 07:03.0 Communication controller: Digium, Inc. Wildcard

Hi, I'm wondering - why SIP device state doesn't get updated to anything else, except Not In Use. For queue call (with Local channel) i get: app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use) app_queue.c: The device state of this queue member, Agent/21168, is still 'Not in

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

Hi, I was trying to compiles addons 1.4 and res_config_mysql doesn't compile. is res_config_mysql still supported and is it still posible to use mysql with asterisk RealTime?? Bests Tomasz

Greetings everyone, today I spent the last part of my day trying to find a parse error inside this snip: http://pastebin.ca/740081 If there's anyone who can shed some light on why my GosubIf condition is throwing a parse error, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Regards, -Michael

Hi, I have an simple queue and agents defines with memeber => SIP/123. If for example Agent "SIP/123" has an call, the queue didnt care and tries to send additional calls to this agents. So Iam loosing time. SIP/123 (In use) has taken no calls yet How to stop this, especially when the device is not able to send an BUSY back. Use LOCAL channels and parse 'show queues' or

We have a customer that has Asterisk 1.4.12.1, Zaptel 1.4.5.1, Asterisk-Addons 1.4.3. running on a Dell Poweredge 1900 server (Dual Core Xeon, 4gb RAM, 500gb Raid 5). Until a month ago they had two TE120P cards and everything was working fine. Since they needed to add a third E1 line we decided to change one of the TE120P cards with a TE210P. After the change we had a couple of crashes (server

On 12/17/07, Jared Smith <jsmith at digium.com> wrote: > On Mon, 2007-12-17 at 12:00 -0800, shadowym wrote: > > I do wish Digium or whoever tests this stuff had a more reliable way of > > testing software releases rather than relying on feedback from the > > community. Fonality, for example use what they call a "hammer" which sounds > > to me like a

Hi, I just had my production box deadlocked - no calls could go trough, CLI didn't load. Last lines in log were: [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Executing [28901 at local_dial:40] GotoIf("SIP/204.11.200.152-c0070ed0", "1?41:57") in new stack [Jan 11 09:15:43] VERBOSE[7265] logger.c: -- Goto (local_dial,28901,41) [Jan 11 09:15:43] VERBOSE[7265]

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

# moving to -users as this belongs there. It is a nice idea to run several Asterisk processes simultenously, it will defineately help with multithreading. However I would suggest trying less instances - that would perhaps give greater benefit, as Asterisk has it's own threading. For example 8 instances of Asterisk / 4 instances.. However, in this case - if You go for splitting everything up,

On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis <anthonyf at rockynet.com> wrote: > http://en.wikipedia.org/wiki/Jacque_Fresco > > A resource based economy. > > Greg Woods wrote: >> On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: >> >>> Gotta love this list being farmed for spammers now. I am sure they call >>> it targeted delivery or

patents, but it's full of legal terms. Maybe anyone can comment? http://www.europarl.europa.eu/commonpositions/2005/pdf/c6-0058-05_en.pdf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835

Hello, I'm trying to find simple MP3 player in flash, to integrate it with call recordings. My requirements would be: * simple UI * buffering (would be nice) * slider * volume control * support of 8kHz stereo mp3 * javascript access to seek/position * free for any use (GPL, MPL, MIT, BSD) So far I've found that JWplayer[1] does great with my recordings. However it's not small in

Hello, I am trying to alter the outbound callerID for extensions within a context I have created. I wrote the following: exten => _9.,2,ExecIf($[$["${REALCALLERIDNUM}" = "360"] | $["$ {REALCALLERIDNUM}" = "670"]]|Set|CALLERID(num)=581560) exten => _9.,3,ExecIf($[$["${REALCALLERIDNUM}" = "361"] | $["$

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

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