search for: rockynet

Hi, How can I have A*k convert a call from +441793xxxxxx to Dial 00441793xxxxxx instead? With the "_+." Below I can "catch" the call, but EXTEN doesn't get set as expected.. and then I need to figure out how to pass the call onto the outgoing-pstn context. Not sure if a Goto would work here... [outgoing-pstn-international] exten => _+.,1,Set(EXTEN=00${EXTEN:+1}) exten

Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 voip at rockynet.com

Has anyone seen this before? These phones are behind an edgewater. SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bKaf87f1c9f;received=xxx.xxx.xxx.xxx. From: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=23943befc9dc103 To: 7408 <sip:5D03C49B-7408@dpl.voip.rockynet.com>;tag=as2c0b7dcd Call-ID: 723559d029d27c820c8dae4b01e45c77@192.168.50.31 This phone is correctly in sip.conf and registered for over a year until today, every single phone behind two different edgewaters have this...

Hello , Thursday, June 14, 2007, 4:00:37 AM, you wrote: > Message: 2 > Date: Wed, 13 Jun 2007 09:40:08 -0600 > From: Anthony Francis <anthonyf@rockynet.com> > Subject: [asterisk-users] Weird sip registration problem > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <46700FD8.1080109@rockynet.com> > Content-Type: text/plain; charset=ISO-8859-1; forma...

Hi, Is there a list of all the extension states as sent by the manager interface? (I know I could look them up in the source but that involves some "backtracing".) The ones I know are: -1: no hint for the extension 0: registered && idle 1: busy 4: unreachable, not registered 8: ringing I've recently seen 16 (== hold?) but can't find that value documented anywhere.

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."

how can I fix this just started ...... Jul 14 14:32:35 NOTICE[4983]: chan_zap.c:6223 ss_thread: Got event 18 (Ring Begin)... == Starting Zap/1-1 at bell,s,1 failed so falling back to exten 's' == Starting Zap/1-1 at bell,s,1 still failed so falling back to context 'default' Jul 14 14:32:35 WARNING[4983]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid

If it is posible for a imcoming call to ring both the Polycom desk phone and my cell phone at the same time, if I dont answer fall back to my voice mail box. I would like to hire someone to cofigure that for me. Bob -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains -------------- next part -------------- An HTML attachment

Hi all! I have an issue with TDM400P FXO card. When a call enter into my IVR and select the proper option, the person that ansswer the call say your "thanks for contact us ..." but the caller cant hear this words because a delay between asterisk and caller part or between asterisk and the ATA device. What is the item on zapata.conf that can affect this delays. Thanks for any help

Would like wish to ALL a Marry Christmas and a happy new year, full of peace, love, happinesses and much success. That let us have one excellent year of 2008. Best Regards Josue Conti.

Hello, A user who uses my Asterisk made me part of a worry about listening to his voicemails. He has received 4 voicemails on January 3, respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at these times. On first listen to his messages, at 8.00 pm, Asterisk has announced two new voicemails(15H00 and 15H36). He has erased thos voicemails. At 8.30pm ,

Hi, folks. I've been on the Asterisk Announce list for a while now, and it seems to me that the release versions of Asterisk are a bit bleeding-edge. They qualify as stable, but I wouldn't call them "production stable" since half the time a new one comes out, a fix for it comes out the next day. So... that said, what's a good version to linger on? I don't *need*

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

...y. I am also registered with FWD on IAX2 and they were always up. Any tech support calls to Teliax would take more than 2 days to get a response. Only when I threatened to leave would someone suddenly pop up and answer my concerns. They claim to have been changing bandwidth providers (away from rockynet, or at least companies that peer with Cogent), but traceroutes show they are still with them. So far, when I've actually gotten ahold of a tech support person, they have told me to try different addresses for the server. They've changed recently from voip.teliax.com to ast01.teliax.com to...

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

Hi all, I just bought the 7970G phone. It's a beautiful phone. In trying to make it work with Asterisk, I've read many posts on the net. However, all of them make reference to having to install the SIP firmware on the phone. Where can I get it? Thanks

Dear which Linux version would be fine for asterisk CentOS 5.0 or Debian 4.0 or RHEL 4.0 Regards Satish patel --------------------------------- Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

Hello Asterisk People. Asterisk have a really annoying bug, i use frequently the manager status command and when asterisk decide not to show the "statuscomplete" event, it really don't show the "statuscomplete" string, in fact none of the "AgentsComplete", "QueuesComplete' are shown.... I use it for monitoring a queue, but this is really getting me

On Thu, Nov 6, 2008 at 7:50 PM, Anthony Francis <anthonyf at rockynet.com> wrote: > http://en.wikipedia.org/wiki/Jacque_Fresco > > A resource based economy. > > Greg Woods wrote: >> On Thu, 2008-11-06 at 09:46 -0700, Anthony Francis wrote: >> >>> Gotta love this list being farmed for spammers now. I am sure they call >>&gt...