[asterisk-users] Digium Phone D40 plus dialing

of the digit map I don’t see any place where I can do a translation and prepend the + to the outgoing string.   The question is how do I dial using plus dialing on the Digit phone?  I’ve tried 00 or holding onto 0 with no success. Any ideas? Thanks Mitch --  Mitch Johnson Sent with Airmail

Re: [asterisk-users] asterisk-users Digest, Vol 99, Issue 37

Chris, Thanks for answering my message. I'm currently using version 10.5.1. I included the error message on the dial plan to show what errors I was displaying. The call goes through after that error message is displayed. As soon as I hear the phone ring, it drops my call on the calling phone

Re: [asterisk-users] One way calling on asterisk to cisco

I did duplicate cucm as cucm2. I was a bit confused as to what changed. However, it was the same results. I commented out the cucm1 instances so it was forced to use cucm2. however I still get the same results: == Using SIP RTP CoS mark 5 -- Executing [8000@myphones:1]

[asterisk-users] One way calling on asterisk to cisco call manager integration

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic

Re: [asterisk-users] TLS/SRTP calls go to circuit busy.

Once again, thanks for your reply. I had done some research already but forget to include it in my previous email. I did find a bug that is remarkably similar to the issues that I'm having. The bug number is 18674. Thanks, Mitch Johnson Message: 8 Date: Fri, 04 Mar 2011 00:34:45 -0600

Re: [asterisk-users] TLS/SRTP calls go to circuit busy.

@lists.digium.com Message-ID: b401c9b4-0721-43b4-9762-c3f02483b...@digium.com Content-Type: text/plain; charset=us-ascii On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote: I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test

[asterisk-users] TLS/SRTP calls go to circuit busy.

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would

[asterisk-users] Question about how traffic passes from phones

Hopefully this is a simple question. How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I like to think that the secure call terminates on the Asterisk and the non-secure call is somehow

[asterisk-users] One way dialing over a SIP trunk

I have a SIP trunk built between a Cisco CallManager version 8. I can dial the phones registered to the Asterisk PBX from a phone registered to the Call Manager. I've tried to keep the config as small as possible to help the troubleshooting process. Attached is he most recent debug. My