similar to: Playing Audio before the Phone is Ready

Has anyone had successfully installed more then two digium wildcards in the same machine? I'm going for four. thanks Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200

I am hoping to replace my Nortel 8x24 switch with Asterisk. Right now my cabling comes from my outside phone box into my office and into a punchdown block and leaves the punchdown block as an amphenol connector which currently plugs into the Nortel swicth. A second amphenol connector them plugs into the switch and extends to another punchdown block that I believe carries the lines throughout the

Is there something special that needs to be done to allow a T100P/T400 to respond to a remote loop request? I am having some issues with a Point to point T1 line using zaptel ppp. This line gave us small problems when we had a pair of Cisco 2600's on either end but now with the zaptel ppp it is going down every couple of minutes for 15 intervals. The phone company says the line is good to

Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic

Does anyone know Zulty?s ZIP 2 ip phone is fully compatible with asterisk? Does the MWI and call transfer work correctly? Henry Devito Telephone Connection, Inc Network Design / Implementation Phone: 402.330.7510 Fax:??? 402.330.8586 ? Toshiba CTX/DK/Stratagy Certified Cisco Certified Internetwork Expert (CCIE) Voice ( VoIP) Cisco Certified Internetwork Expert (CCIE) Routing and Switching MCSE

Hi, I've just started trying allow_nets on one of my servers. I have auth_debug and auth_verbose both enabled and the output is as follows: Oct 28 13:05:48 mink dovecot: auth-worker(default): auth(user at domain.net,x.x.x.x): allow_nets: Matching for network 127.0.0.1/8 Oct 28 13:05:48 mink dovecot: auth-worker(default): auth(user at domain.net,x.x.x.x): allow_nets: Matching for network

Does asterisk support the intel 537/md3200 chipset? I don't want to start any flames here, I know all about using generic crap in asterisk,[*] which I really don't approve of other than for testing, but I have a customer demanding a generic chipset for his one backup analog line. He will not spend the money for a Digium card and says he will find another company if I can not provide a

I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application. Added the following [callingcard] ; CallingCard application exten => 777,1,Answer exten => 777,2,Wait,2 exten => 777,3,DeadAGI,a2billing.php exten => 777,4,Wait,2 exten => 777,5,Hangup I am using 777 as the calling card

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Hi Timo, Was getting the following error repeatedly in my mail client (squirremail) "error: connection dropped by IMAP server" Looked at the logs for this user and found the following: Nov 19 17:11:32 mink dovecot: IMAP(bobuser at maildom.com): broken sync positions in index file /virtual/store2/mail/s/bobuser_maildom.com/Maildir/dovecot.index Nov 19 17:11:32 mink dovecot:

If I want to send outgoing DTMF over a BRI interface, can I do it with 'isdn4linux' or must I use the 'capi' library? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ mje@posix.co.za - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 -------------- next part -------------- A non-text attachment was

Hi Everyone, I've been asked to determine which phones our organization should go with. And I've narrowed it down to the Polycom IP500 or the Cisco 7940. From my travels through google, it's hard to find a definitive comparison of the two phones. So I thought I would ask the people that have probably used both. From what I can tell, the only major benefit the Cisco has over the

Hi, I'm using postfix with Dovecot 1.0.5 and getting the following errors when a user moves a massive folders contents to trash (65000 or so messages): Apr 21 15:31:01 mink dovecot: IMAP(dev2 at aluminati.net): Timeout while waiting for lock for transaction log file /virtual/store1/mail/d/dev2_aluminati.net/Maildir/.Deleted Items/dovecot.index.log Is there anything I can do to compensate for

Hi, I've got two imap boxes, one is running 1.1.5 and hasn't exhibited the error below, but the one that's been upgraded to 1.1.6 is showing them. Nov 14 16:06:31 mink dovecot: imap-login: Disconnected (no auth attempts): rip=x.x.x.x, lip=x.x.x.x, TLS handshaking: SSL_accept() failed: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac If this isn't

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear all, bear with me, I know the SUBJECT sounds pretty unclear. I'll clarify in a minute. And please excuse that due to the keywords being unclear no usable help was found on google & Co... Assume there is a workstation, which connects to multiple machines, one of which is considered potentially unsafe. So, it would be nice to have agent

I tried to open a Trac ticket on this, but the menus are not working for some reason right now. I''ve noticed this in our puppetmasterd logs since upgrading: puppetmasterd[7943]: Could not call: nested *?+ in regexp: /^\/var\/www\/html\/download\/NSClient++-0.2.7.zip/ Seems like the data passed should have RE characters quoted (ala \Q...\E in Perl) prior to this test. Regards, Mark

The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:

Hi can you help me to build a asterisk manager command event to join a conference? i've seen that there is the event Event: MeetmeJoin Channel: <channel> Uniqueid: <uniqueid> Meetme: <meetme> Usernum: <usernum> Can you explain me how it works? Can i use it to join an existing conference? Thanks

hi I wonder if you would like to join my localplayer.com network? I operate on the same philosophy. All my artists are unsigned and I have their consent to stream their music. on my station. I envision a drop down menu that helps me navigate to the city of my choice. Raymond wrote: > Hello: > > I'm starting a radio station on the LPBN. The music will be from all > local Los