Guess I got where your confusiion lies.... you DON'T set up the ALL contexts
the users will have acces to in sip.conf, in this file you will only set 1
single context. All other contexts you want the users to have acces to, you will
have to add them in the extensions.conf
So what you have to do is the following:
-user 2092, set it the createmenu context in sip .conf
- in extensions.conf under the "createmenu" context add as many
"include" lines as you need the user to have acces to.
it should look something like:
; sip.conf
[2092]
type=friend
username=2092
canreinvite=no
context= createmenu
; extensions.conf
[createmenu]
...
...
include => outgoing
include => some other context
hope this helps
Alyed
----------------------------------------
Clean DocumentEmail Hi, Thanks for the reply. What happens is
that all users are registered with SER (a sip proxy). I have set SER up so when
a user dials '0' followed by a pstn number it will be forwarded to
asterisk which will forward the call to a third party pstn gateway. I also use
asterisk so that when a user who is registered with ser doesn't answer
(sending a 408 cancel response) or is busy (sending a 486 busy response) that
the call is forwarded to asterisk voicemail. So therefore at the moment I have a
user '2092' which registers with ser and uses the 'outgoing'
context in asterisk for pstn access and accesses their voicemail mailbox through
the default context. Now I also set it up so that if a user registered with
SER dials 20005 it should forwards to asterisk. This should call the context
'createmenu' which creates an IVR menu. What I'm confused about
is this. I created a user 20005 in sip.conf using context=createmenu. This
wasn't working. After reading your post I realized my mistake was that the
context that is being called is that of the caller i.e. 2092 as opposed to whom
the call is directed at i.e. 20005. Therefore when I changed the context of 2092
to 'createmenu' it worked. BUT how can I set up my sip.conf so that
2092 can use the default, outgoing and createmenu contexts depending on the
correct scenario? If someone who is also using SER has any comments, I'd
also really appreciate it. i.e. [300]
type=friend
username=300
canreinvite=no
context= WHAT GOES HERE?? //createmenu calls the IVR but then outgoing
pstn calls don't work, outgoing allows pstn calls but then I can't
create a menu etc etc
insecure=very
;callerid= "voicemail user 1" <300>
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=g723.1 allow= g729 Many thanks, Aisling. -----Original
Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alyed Tzompa
Sent: 04 January 2006 00:28
To: asterisk-users@lists.digium.com
Subject: re: [Asterisk-Users] confusion about contexts I'm a bit
confused on how you get your calls to Asterisk, what I mean is: are you phoning
into asterisk via a sip user? in this
case, which one?, if not is it iax or though a zap channel?
anyway, here some tips:
For your first problem it seems it has to do with what I pointed above, check
that the user which is dialing into asterisk has
the correct context (context=create-menu) with at least type= peer
also don't have to retype the allow=codec, disallow=codec, dtmfmode=x for
every user, just set it in the general context in
sip.conf
your second problem think it has to do once again with the firts thing above,
and regarding the retyping, I'm afaid I don't know
any other way than writing those lines again and again for everyuser. Maybe
someone else out there knows someting else that can help.
Don't set many "outgoing" context for every user in sip.conf!!!!!
just set one and point all users to that one. If you need your
user to have acces to other contexts just add
include => your_context
at the end of whatever context you want (btw can add more than one
inlcude's )
Alyed
-------------------------------------------
Hi,
Hope someone can help me-Asterisk isn't behaving as I would expect
and I think it's down to my contexts.
There are two things I can't fathom.
Firstly I want to record an IVR and so have created a user 20005 and
a context called createmenu. I am using SER in front of asterisk so I
changed the ser.cfg so that if the user dialled this number it
forwards to asterisk. This works fine. The problem is when the invite
reaches my asterisk box, asterisk uses the wrong context. It appears
to call the "outgoing" context which is the context used to route
calls to my pstn gateway provider. It then trys to execute a "Dial"
command for 20005 which isn't supposed to happen.
Secondly SER uses Asterisk for voicemail if a phone doesn't answer
after a certain period of time or is busy. This works fine but I have
to create an entry for every user in extensions.conf under the
[default] context. Can I create a generic entry which would also work
to shorten the config file?...Also if I change this and out all the
entries under a context "voicemail" it doesn't work..I have to
keep
it in default.This must obviously be something got to do with
Asterisk finding the contexts.
I am confused as to how you apply multiple contexts to one user. At
the moment nearly each user (besides 20005 and 1234) has a context of
'outgoing' in sip.conf. This is so that they can make outgoing pstn
calls.But what if I needed them to use another context in other
situations?...I'm just confused as to what context should be applied.
I have included the relevant parts of my sip.conf and extensions.conf
below. I would appreciate any advice as to why these issues are
occurring.
Many thanks,
Aisling.
;sip.conf
[general]
bindport=5064
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
srvlookup=yes
canreinvite=no;
autocreatepeer=yes
nat=yes
dtmfmode=info
insecure=very
registerattempts=0
register => username:password@sip.blueface.ie/1234
;To receive incoming calls specify this and replace
"yourcontext-pstn" for your dial plan
[blueface-in]
type=peer
host=sip.blueface.ie
context=pstn
[1234]
type=friend
username=1234
canreinvite=no
context=pstn
insecure=very
;callerid= "Ais" <1234>
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=1234
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
;added below line(s) for BLUEFACE conf
;To make outgoing calls specify this block
[blueface-out]
type=peer
host=sip.blueface.ie
username=username
secret=password
[20005]
type=friend
username=20005
canreinvite=no
context=createmenu
insecure=very
;callerid= "Ais" <20005>
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=20005
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
[300]
type=friend
username=300
canreinvite=no
context=outgoing
insecure=very
;callerid= "voicemail user 1" <300>
host=dynamic
nat=yes
dtmfmode=INFO
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=g723.1
allow=g729
extensions.conf
[general]
static=yes
writeprotect = yes
[createmenu]
exten => 20005,1,Wait(2)
exten => 20005,2,Record(/tmp/asterisk-recording:gsm)
exten => 20005,3,Wait(2)
exten => 20005,4,Playback(/tmp/asterisk-recording)
exten => 20005,5,wait92)
exten => 20005,6,Hangup
;specify context for receiving incoming calls
[pstn]
;Note this is just an example there are infinite different ways to
handle the incoming call.
;exten => 1234, 1,Wait(1)
;exten => 1234, 2,Playback(beep)
;exten => 1234, 3,Hangup
exten => 1234, 1, Dial
(SIP/2092@seraddress) ; 1234 is the contact extension, default
contact extension is "s"
;exten => 2092,1,Answer()
;exten => 2092,2,Playback(welcome)
;exten => 2092,3,Background(menu)
;exten => 1,1,Dial($316)
;exten => 2,1,Dial($314)
[outgoing]
; Dial the Blue Face Speaking Clock
exten => 300,1,Dial(SIP/300@blueface-out)
exten => 300,2,Hangup
;Send PSTN calls to Blue Face
exten => _X.,1,Dial(SIP/${EXTEN}@blueface-out)
exten => _X.,2,Hangup
[default]
exten => 300, 1,Dial(SIP/300,20)
exten => 300, 2,Voicemail(u300)
exten => 300, 102,Voicemail(b300)
exten => 300, 103,Hangup
exten => 301, 1,Dial(SIP/301,20)
exten => 301, 2,Voicemail(u301)
exten => 301, 102,Voicemail(b301)
exten => 301, 103,Hangup
etc etc
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