This may be a dumb question, but I have never done menus, how do I link
the below up to my phone number? For example, right now I route calls
to the SIP phone like so:
[ipcomms]
include = default
exten = 211212, 1, Dial(SIP/6000,20,tr)
where ipcomms is my context from sip.conf for
Before I attempt to program a system like this, I wanted to see if: A)
its possible, and B) its too insanely difficult for a perl developer.
My office building has a dialer on the front door so people can call me
and gain access. The dialer on the door has a full keypad, and
basically just
I posted an email a few days regarding a problem with hearing the
voicemail greeting on my sip phones.
It turns out to be a phone/stun/linksys issue - not an asterisk issue.
Which brings up a couple of questions
I always assumed that you can have multiple SIP phones behind a Linksys
, Shane D wrote:
Very odd. Could you try taking the mailbox line out of sip.conf and
see what happens?
On 1/31/08, John Von Essen [EMAIL PROTECTED] wrote:
Here are my configs:
sip.conf:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
[6000]
type
]: app_voicemail.c:6281 vm_authenticate:
Couldn't read username
Really destroying SIP dialog '[EMAIL PROTECTED]' Method:
BYE
So it plays the greetings, and is working, I just cant hear it.
-john
On Jan 31, 2008, at 3:00 AM, Andrew Joakimsen wrote:
On Jan 31, 2008 12:30 AM, John Von Essen [EMAIL
Ok, I have spent all night trying to figure this out, and hopefully
somebody has a similar experience.
I have a very basic asterisk config. Sample configs, with the only
addition being by SIP phone, and my incoming voip. Last week I got
everything setup, calls were working, etc.,.
I cam
Tried it, but no change.
A few updates. Even though I dont hear anything, if I hit a keys on the
phone and then hang up, message log says:
[Jan 30 21:26:57] WARNING[7917] app_voicemail.c: Unable to read password
I enabled logging of everything, and the below is the snippet for when
my
