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 originat
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 ring
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
firewa
00 PM, 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]
>> con
-- Playing 'vm-login' (language 'en')
[Jan 31 06:42:49] WARNING[8513]: 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
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 SIP/6001
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 across