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Is there a way to limit outbound calls when feeding files to the outgoing directory in asterisk? I several thousand files i need to feed asterisk, hoping to copy it to the outgoing directory all at 1 time.

Anyone know if the MySQL() application has a configurable timeout? If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I'd like to cut it down to a few seconds. Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part --------------

I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office).

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like "number 12345 not available" I was only hearing "345 not available". Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

what method is preferred: haylafax and Iaxmodem or spnadsp for faxing. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071228/e829632a/attachment.htm

Polycom Provisioning Tool Updated. I made a bug fix that was reported, which was causing the directory creator to not work when there was an invalid character in the filename of the csv. I have also posted an FAQ: http://www.wintrisk.com/ppt.html#FAQ Download the new one, and tell me what you think! It's free, and mildly useful! http://www.wintrisk.com/ppt.html Yours, Michael Munger,

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

Hello, I have another little problem with my ZAPs channels, in fact, when I received a call, I heard no sound while in the CLI, sound is played: -- Starting simple switch on 'Zap/4-1' -- Executing [s at from-zaptel:1] Answer("Zap/4-1", "") in new stack -- Executing [s at from-zaptel:2] BackGround("Zap/4-1", "hello-world") in new stack --

>>Your membership in the mailing list asterisk-users has been disabled >>due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Another question, Is anybody using the Tellabs 2572 EC with a PRI. When we moved from our analog lines to a PRI, I thought it would be simple stuff moving the EC to the PRI. Changed signaling, made sure that channel 24 wasn't being ECd and everything came up. But, I was getting complaints of random echo on the PRI. Local echo. Also, we weren?t able to do any kind of modem dial-outs (Adit

List, Im a Cisco certified Network guy with little telecom experience (BRI/PRI at the time) so please forgive my terminology. I am showing interest after the Network World SHSU October 4 article. We have 3 offices (Hub-Spoke T1 Frame relay to the remote offices(Data & voice on separate T)). Each office currently does their own thing for telecom. Our Main(HUB) office currently has 14 channels

Hi all, I am trying to set up DND in my asterisk, I am using the following context: [app-naoperturbe] exten => *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten => *11,2,GotoIf($["${DND}" = "YES"]?*11,3:*11,101)exten => *11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten => *11,4,Playback(beep)exten => *11,5,Hangup()exten =>

I have a TDM2402E card. Occasionally I have noticed that a number I call that gives and IVR the DTMF keys are not detected. All other times the DTMF works fine. /proc/interrupts is: CPU0 CPU1 0: 500798020 500749281 IO-APIC-edge timer 8: 4 9 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 15: 4505912 4502264 IO-APIC-edge

Does anyone have any application examples for Confbridge in Asterisk 10? I'm just looking for simple ad-hoc functionality similar to meetme in 1.8. Thank you in advance.

Hai, Does anybody have problems with a wrong Connected Line ID with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej

hello list, I am trying to arm an ivr for schedule of office and outside of office [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no [in] include =>scheduleofservice|08:00-18:00|mon-fri|*|* include =>outsideofschedule|18:00-23:59|*|*|* include =>outsideofschedule|00:00-07:59|*|*|* include =>outsideofschedule|*|sat-sun|*|* [scheduleofservice] exten