similar to: Log level of 500 Server Internal Error.

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

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

# 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

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

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

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]

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

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

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

Hello, Has anybody seen that Audiocodes gateway is replying with "486 Busy here" when it's actually not (last call ended ~15 seconds ago). I see this quite often. From other logs i see that previous call ends at 11:13:01, then app_queue tries to dial at 11:13:14 and fails numerous times, before succeeding at 11:14:02 I have attached sample SIP debug log: Any ideas what i could

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 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