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
c
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] Dial("SIP/2002-00
I have a SIP trunk in between a Cisco CallManager 8.6 and Asterisk 1.8.5.0.
I can call from the Cisco to the Asterisk but cannot when i call from the
asterisk to the Cisco.
I get the following error message:
Got SIP Response 400 "Bad Request - 'Malformed/Missing URL' " back from
172.123.27.20:50
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 y
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 Callm
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 great
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 join
nt-Length: 0
>
> Message: 8
> Date: Tue, 1 Mar 2011 10:04:14 -0600
> From: Terry Wilson
> Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy.
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Message-ID:
> Content-Type: text/plain; c
> 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,
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
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