similar to: MusicOnHold. not getting it.-GOT IT!!

G'Day All; I am a little unsure on how to get Music On Hold to work. Please critique my extensions.conf. ????? Thanks ; SIP 5001 exten => 5001,1,Dial(SIP/5001) exten => 5001,2,Voicemail(u${EXTEN}) exten => 5001,3,Hangup exten => 5001,102,Voicemail(b${EXTEN}) exten => 5001,103,Hangup Thanks -------------- next part -------------- An HTML attachment was

How "real time" is realtime? If the extensions.conf is stored in the database, does * query it row by row or is it "cached"? In other words, given the following exerpt: exten => 5001,1,Dial(IAX2/test@test/s,30,g) exten => 5001,2,Voicemail(u5001) exten => 5001,102,Voicemail(b5001) exten => 5001,103,Hangup exten => 5002,1,Dial(IAX2/test2@test2/s,30) exten =>

Before you all reply that its available via Cisco, I'm not qualified to be a tech member according to Cisco. I just bought 4 7960's with which to use with * and I want to load up the SIP image into them. Does anyone have it that they can make available to me please? Thanks -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com/

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

So I go into a new Apple store on Sat to buy some stuff for my Mini, and I notice some Snom 360's on the sales counter. Venturing a question, I ask, are they using Asterisk? Guys says yes. Cool! I said: What kind of box are you using. He points to a Mini sitting on the counter! 2 X cool! He's using a SIP-FX0 converter. Plug: http://www.mymacdealer.com great store in Alberta. Anyone else

Mark, Thanks a lot for the feedback. It's reassuring to say the least Mike Message: 18 Date: Sat, 21 Jan 2006 15:36:18 -0500 From: Mark Phillips <g7ltt@g7ltt.com> Subject: Re: [Asterisk-Users] SIP and NAT - best practices? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <43D29B42.3060705@g7ltt.com> Content-Type:

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

Hi Yawl, After about 6 months of prattting about I've convinced my boss that we should be installing * into our currently under constuction Data Center in Somerset NJ. There will be 10 permanent people and DR space for another 50. My plan is as follows; ATAComm dual XEON server with quad T1 board. A handfull of ATA's for fax machines, job lot of X-Pro softphones for the DR bit, Polycom

That's great.....it's a virus I tell you * is everywhere :) Viva la asterisk. Cheers, Dean > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Mark Phillips > Sent: Friday, 3 June 2005 6:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject:

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

Interestingly enough I had this same problem today.... 1. I created the directory and permissions for the directory " /var/lib/asterisk/festivalcache/ " (per the comment in the festival.conf file) 2. I had to comment out some things in the festival.conf file: the "host" line, the "port" line, and the "festivalcommand" line. I have also noticed the

Hi Folks, I'm trying to get DIAX to work as an IAX soft phone and I'm having some issues. Outbound calls from the phone to the rest of the world are fine. It's just that I can't get the inbound to work. I have a stanza in my extensions.conf that reads; exten=>_3410,1,Dial(IAX2/3444@default|20) And a stanza in my iax.conf that reads [3444] type=friend context=default

Do you think this should be "Bug Reported"? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Samaritoni Sent: Wednesday, October 06, 2004 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Festival, getting gethostbynamefailed error

I've been banging my head with this all day. I today switched from a very old CVS build to AAH1.3 and so far everything has been easy. However I cannot accept calls from a previously working IAX trunk. I've set up an trunk with all the same credentials as before and can call the folks at the other pbx. However whenever they call me I tell them that I don't have an

Hi Folks, I have an IAX trunk link to a collegues house. I'm using AAH and he's got the latest CVS as of last Tuesday. Problem we're having is this; when I dial his extension 7201 (Pulver WiSIP phone) his * box sends me 1 ring and then Alison's busy message. If I call his 7202 extension (X-Ten Pro on a Win2K laptop) I get through but with only 1 way audio (me to him). Until

I downloaded the latest CVS a few days ago. It all compiled nicely on my new AAH platform. However, it won't start up. Investigation of my log files produces this; Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 VERBOSE[31473] logger.c: [pbx_wilcalu.so] Jul 26 22:59:18 WARNING[31473] loader.c: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined symbol:

I just built it and now can no longer get incoming or outgoing service. It was working with CVS Head prior to my "downgrade". Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

I think I've asked this before and think that Matt had said something about this. Is there an LCDproc client for Asterisk available and if so how can I get a copy please. Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very