search for: anthonyf

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

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

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...

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

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...

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

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

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."

Hi all, I have a strange situation on a Asterisk 1.2.17 with FreePBX 2.2.1 Doing a select in the CDR table I noticed there are some calls with billsec greater than duration, duration is always 0 in those calls. How can this happens ? Am I missing something ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l.

Anybody care to muse on Wi-SIP vs. SIP-DECT? My limited research indicates that none of the WiSip phones will ever be able to match the performance of DECT phones. Maybe I'm wrong but a Wi-SIP phone seems like a DIESEL sports car. There is nothing wrong with the technology, but it seems like a shoe-horned fit into the requirements of a wireless endpoint. DECT uses a wireless radio layer

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