Displaying 9 results from an estimated 9 matches for "internalextension".
2006 Jan 06
2
Incoming PSTN Calls - Stumped
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that?s happening (and I?m very stumped
with this)
.I think my contexts are definately the reason that I
can?t interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my sip proxy (SER). Therefore w...
2006 Jan 05
1
Incoming PSTN Calls
...[blueface-in]
type=peer
host=sip.blueface.ie
context=incomingpstn
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
[incomingpstn]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1) //press 1
for internal extensions.
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and pointing it
to the appropriate context but that didn't work either. The only way I
can get...
2006 Jan 11
0
Incoming PSTN Calls - Can't interrupt Main Menu
...assword@sip.provider.ie/
>
>In my extensions.conf I then have
>
>[incomingpstn]
>exten => s,1,Wait(1)
>exten => s,n,Background(MainMenu)
>exten => 1,1,Goto(internalExt,s,1)
>exten => 2,1,Goto(mainconfmenu,s,1)
>
>[internalExt]
>exten => s,n,Background(InternalExtension)
>
>[mainconfmenu]
>exten => s,n,Background(MainConfMenu)
>
>I can hear the MainMenu sound file being played. What's strange is that
>when I press '1' to interrupt, which in my logic should invoke the
>internalExt context, nothing happens. The MainMenu sound file...
2004 Jun 21
0
dialplan help!-RESOLVED
...=> s,1,Answer
> exten => s,2,Wait(1)
> exten => s,3,ResponseTimeout(2)
> exten => s,4,DigitTimeout(6)
> exten => s,5,Background(bgw-ThanksForCallingBgw)
> exten => s,6,Background(bgw-IfYouKnowExt)
> exten => s,7,Background(bgw-OtherwiseHold)
> include => internalextensions
>
> exten => t,1,Dial(ZAP/3,15)
> exten => t,2,Background(bgw-ThankYou)
> exten => t,3,Hangup
>
> exten => i,1,Background(bgw-SorryInvalidEntry)
> exten => i,2,Goto(s,5)
>
> exten => h,1,Hangup
>
> [fromzap]
> ignorepat => 9
> include =&g...
2006 Dec 20
1
Incoming Lines Confusion
First off, please, for the love of God, don't cremate me, if I should
already know the answer to this!
I've installed a small setup for an office who wanted to be able to talk to
each other instead of having to rely on MSN to communicate. Weird request, I
know, but hey, we do what we need to do to get paid. I installed soft
phones, gave everyone an extension, and bingo, they can call and
2006 Feb 09
0
Sip One way audio
...; Address to bind to
context = incomingcall ; Default for incoming calls
tos=lowdelay
disallow=all
allow=alaw
allow=gsm
allow=ulaw
[104]
accountcode=vsllc
type=friend
context=employee
username=104
secret=**redacted**
host=dynamic
qualify=yes
reinvite=no
canreinvite=no
mailbox=104@internalextensions,750@internalextensions
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
;nat=no
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2005 Feb 23
2
Creating extension groups
Hi
I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server.
Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please
2005 Mar 14
2
asterisk outbound to SIP provider problems
Hi
I am having alot of difficulty connecting to SIP providers (I am trying 3)
and can't seem to find anything similar in the wiki or on the lists.....I
can receive inbound calls fine however on placing an outbound call the
calling phone never gets 'connected' but 2 way audio is passed for about
20secs before some sort of timeout.
Anything suggestions as to what I could try
2005 Jun 23
0
Voicemail recording cutoff when silent for 1 second
...Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
pbx*CLI>
Sip read:
0 headers, 0 lines
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/internalextensions/230/INBOX/msg0008
format: wav49, 0x8144680
Jun 24 00:56:31 WARNING[7507]: app.c:619 ast_play_and_record: No audio
available on SIP/233-a3ba??
-- User hung up
-- Executing GotoIf("SIP/233-a3ba", "1?menuinternal|t|2") in new
stack
-- Goto (menuinternal,t,2)
-- Exe...