similar to: Asterisk 1.2.4 and IAX MOH

I'd like to use native moh instead of with mpg123... for some reason the processes never bloody die. For native moh to not spawn an external player, I'd need to convert the default supplied moh sound files in /var/lib/asterisk/mohmp3 to ulaw and g729 format. Anyone know of a free, easy way to convert them? Thanks, Doug.

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."

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

>>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."

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

/etc/asterisk/musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/mohmp3 application=/usr/local/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s -- Executing Answer("Zap/1-1", "") in new stack -- Executing MusicOnHold("Zap/1-1", "") in new stack -- Started music on hold, class 'default', on channel 'Zap/1-1' --

Hi all, I've got an AGI script that launches the conference bridge with a line like: "$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)" The $conf variable contains the room number. I'm trying to configure it so that when only one person is in the conference, they hear moh. My /etc/asterisk/confbridge.conf looks like:

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

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

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

Hi, After following the suggestions on the mailing lists and the wiki I'm still experiencing choppy moh. The song plays but with frequent noise parts. - I'm using asterisk 1.2.4 on our production server and 1.2.7 on the test server. - native moh with .gsm and .pcm formats (according to http://astrecipes.net/?n=152) - compiled ztdummy as a timing source any pointers on how to dig deeper

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

Dears -How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway. Even I can see in the CLI debugging the status is ringing -my idea is to add music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing line In which script this line located? -- Executing [97130245612 at default:1]

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

Tracker seems to be down. -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

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."

>> Has anyone benchmarked Asterisk on a dedicated single versus dual >> processor machine? > > http://www.astertest.com/ > > Cheers, Philipp The test results that Philipp pointed out show some protocol comparisons that include "iax2 trunking / alaw" and "iax2 / alaw" and concludes that "IAX2 trunking is more than twice as fast as non trunking

I am playing around with IAX encryption and have had good success. I read somewhere, that trunked packets are not encrypted. Does anybody know if this means the trunk packets themselves are not encrypted but the voice frames in them are encrypted or does this mean that if you are using trunking then encryption of the voice frames will not occur. I have used Wireshark to sniff the packets and it

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

I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.