similar to: bugs.digium.com

>>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."

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."

Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1. Everything is working well, until I noticed that Caller ID between facilities are showing properly, on the phone display, until the handset is picked up, then it's displaying NAME NOT FOUND. I do database lookup against extension number (Of remote PBX) and use that as the MySQL key to pull name. Snippet below: -- Executing

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

Hey everybody, I've, within the last 3 weeks, moved over to a PRI from SBC/AT&T. I've received several complaints about dropped calls. Reviewing the archives on PRI and dropped calls shows that I should set the resetinterval=never in the zapata.conf and restart. This hasn't helped. The dropped calls have to date only been on outbound calls. Usually within 2 to 3 minutes

I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26: https://issues.asterisk.org/view.php?id=8824 I've been running 1.4.21.1 for a while and have the need to update one of our Asterisk Conference/Faxserver/switches and wanted to see if the patch available would apply. It failed in several places. This keeps me from updating my other installs. Doug -- Ben

That can't be correct. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Hey everybody, I wanted to know what other may be doing to stem the flood of inbound junk faxes? We currently using Asterisk/iaxmodem/Hylafax for fax services and get a number of junk faxes daily. Most (If not all) of them have caller-id blocked and have a TSI of "". I was hoping that, since we are using a PRI, there would be other information coming across that I could use to

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).

Is it just me? After the mail list server upgrade, the average delivery time for messages to the users list is between 4 and 5 days. The Dev list seems fine! Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec. I'm also moving those installs from Mandriva 10.0 to Debian 6.06 (Squeeze). In my testing, the TE220P PCIe cards that I have, the timing was awful on both slots, so I compiled Kernel 3.6.9 from kernel.org. Timing jumped to

Ah, yes. One of the many differences between the US and the rest of the world. >>> support@drdos.info 4/3/2007 2:52:16 PM >>> john beaman wrote: > I too was curious about this, so I copied the text into Babel Fish, and this is the result: > > I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle

Everybody, I have a fall though context that, among other things, tests to see if someone it trying to pick up a non-existent parked call (Defined from 90 to 99). I have the following: [not-in-service] exten => _X.,1,Wait(1) exten => _X.,n,ResetCDR() ; ************************************************** ; Check to see if the mis-dialed number was a parking ; slot. If so, jump to the

I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug

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

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.

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Has anybody has issues with the new Native MOH and IAX trunking when placing a call on hold? My scenario, Call is placed on a Definity G3 via PRI to Asterisk. Gets trunked over to another Asterisk system via IAX2. Call is answered by operator and placed on hold. At that point, audio is very broken and feedback pulsing is heard. Bad enough that the caller hangs up. If the call is

We've moved within the last two months to Asterisk 1.4.x All remote facilities are connected via highspeed (9mbit) connections (Over OpenVPN) to a central Asterisk box, acting as a voice router, that funnels all calls into our Avaya Definity G3R via PRI. When corporate employees visit the remote facilities and try to call the G3R's voice mail system(Audix), DTMF is not recognized unless

I've replied to two emails in the last two days and haven't seen them yet. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."