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 when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ?dummy? extension.
sip.conf
register => username:password@provider.ie/2093
[provider-in]
type=peer
host=sip.provider.ie
context=onecontext
[2092]
type=peer
other stuff
context=onecontext
So the dummy extension here is ?2093? and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
?onecontext? context.
Now in my extensions.conf ?onecontext? includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ?onecontext? is to allow outgoing access to the
PSTN.
[onecontext]
include => createmenu //creating an IVR menu
include => createconf //creating a conf call
etc
include => default //used for voicemail
[createmenu]
;does something
[createconf]
;does something
;outgoing calls ? main purpose of onecontext
exten => _X.,1,Dial(SIP/${EXTEN}@provider-out)
exten => _X.,2,Hangup
[default]
;mailbox for 2092 and other users
Now this is where the problems start! For incoming calls I tried to
do ?include => incomingpstn? in ?onecontext? which I thought would
call a new context called ?incomingpstn? which would have an entry
for the dummy user. i.e.
[incomingpstn]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1) //directs to another
context called Internal Extension
I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn?t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ?incomingpstn? context
to the default context i.e.
[default]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1) //directs to another
context called Internal Extension
With this I can hear the MainMenu when I dial my DDI but I can?t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don?t have access to the other features in Asterisk.
The point is that I?m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.
This is what comes up on my asterisk console:
-- Executing Dial (?SIP/2092-2829?, ?SIP/021123456@provider-out?) in
new stack
-- Called 021123456@provider-out
-- Playing ?MainMenu? (language ?en?)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
?SIP/2092-5837?
== Spawn extension (default, 2093, 2) exited non zero etc etc
I?m very stuck on this and can?t figure it out.
Any help appreciated.
Many thanks,
Aisling.
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls
Is Exist "InternalExtension" context ? and 2093 exten ?
2006/1/5, Aisling < ashling.odriscoll@cit.ie>:
Hi all,
I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
My provider told me to change my sip.conf as follows
register => username:password@sip.blueface.ie/2093
; To receive incoming calls specify this block and replace
"yourcontext" for your dial plan.
[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 it to work is to copy the code I had in the
'incomingpstn' context of my extension.conf to the 'default'
context.
i.e.
[default]
exten => 2093,1,Wait(1)
exten => 2093,n,Background(MainMenu)
exten => 1,1,Goto(InternalExtension,2093,1)
Then the file would play. First of all I don't get why this is
It
doesn't even seem to refer to the code in my sip.conf
I don't get it.
Secondly whilst moving this code to the default context means I can
hear my initial welcome menu, when I press '1' to interrupt the menu
and move to menu option 1 (another sound file) it won't let me
interrupt and I eventually get the error "Timeout but no rule 't'
in
context 'default".
Does anyone have any ides where the problem might be?
Many thanks,
Aisling.
-------------------Legal Disclaimer---------------------------------------
The above electronic mail transmission is confidential and intended only for the
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sender at the above quoted email address. Any unauthorised form of reproduction
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I had a similar problem , and then used GoTo instead of include Iqbal Aisling O'Driscoll wrote:>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 when I create an >entry for them in sip.conf I set only one context. Also to allow for >incoming calls from my provider it seems I must direct the calls >firstly to a ?dummy? extension. > >sip.conf > >register => username:password@provider.ie/2093 > >[provider-in] >type=peer >host=sip.provider.ie >context=onecontext > >[2092] >type=peer >other stuff >context=onecontext > >So the dummy extension here is ?2093? and 2092 is a phone who >registers with SER and when SER redirects to Asterisk uses the >?onecontext? context. > >Now in my extensions.conf ?onecontext? includes other contexts. This >is how I get access to conference calls, creating IVR menus etc. Also >the main purpose of ?onecontext? is to allow outgoing access to the >PSTN. > >[onecontext] >include => createmenu //creating an IVR menu >include => createconf //creating a conf call >etc >include => default //used for voicemail > >[createmenu] >;does something > >[createconf] >;does something > >;outgoing calls ? main purpose of onecontext >exten => _X.,1,Dial(SIP/${EXTEN}@provider-out) >exten => _X.,2,Hangup > >[default] > >;mailbox for 2092 and other users > > >Now this is where the problems start! For incoming calls I tried to >do ?include => incomingpstn? in ?onecontext? which I thought would >call a new context called ?incomingpstn? which would have an entry >for the dummy user. i.e. > >[incomingpstn] > >exten => 2093,1,Wait(1) >exten => 2093,n,Background(MainMenu) >exten => 1,1,Goto(InternalExtension,2093,1) //directs to another >context called Internal Extension > >I also changed the [provider-in] for context=incomingpstn in my >sip.conf. However this didn?t work and I kept getting directed to the >voicemail of my pstn provider. The ONLY way I could get the incoming >calls working was to add the contents of the ?incomingpstn? context >to the default context i.e. > >[default] > >exten => 2093,1,Wait(1) >exten => 2093,n,Background(MainMenu) >exten => 1,1,Goto(InternalExtension,2093,1) //directs to another >context called Internal Extension > >With this I can hear the MainMenu when I dial my DDI but I can?t seem >to interrupt to divert to another submenu. In the testing that I have >done the user that is making the call is 2092 registered with SER. If >I change the context of 2092 directly in sip.conf to incomingpstn, >then I can hear the menu and interrupt to go to the submenu. But >obviously then I don?t have access to the other features in Asterisk. >The point is that I?m stumped as to why it only works in the default >context and if this is the case how do I get it to call the submenu. > >This is what comes up on my asterisk console: >-- Executing Dial (?SIP/2092-2829?, ?SIP/021123456@provider-out?) in >new stack >-- Called 021123456@provider-out >-- Playing ?MainMenu? (language ?en?) >-- other messages (not relevant I think) >== Spawn extension (outgoing, 021123456, 1) exited non-zero on >?SIP/2092-5837? >== Spawn extension (default, 2093, 2) exited non zero etc etc > >I?m very stuck on this and can?t figure it out. >Any help appreciated. > >Many thanks, >Aisling. > >-----Original Message----- >From: asterisk-users-bounces@lists.digium.com >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of >Giovanni Miano >Sent: 05 January 2006 21:09 >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [Asterisk-Users] Incoming PSTN Calls > >Is Exist "InternalExtension" context ? and 2093 exten ? >2006/1/5, Aisling < ashling.odriscoll@cit.ie>: >Hi all, > >I am having difficulty getting incoming PSTN calls working. I have >set up an account with a third party provider. In my system, the user >register with SER and use Asterisk for PSTN access, voicemail etc > >My provider told me to change my sip.conf as follows > >register => username:password@sip.blueface.ie/2093 > >; To receive incoming calls specify this block and replace >"yourcontext" for your dial plan. >[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 it to work is to copy the code I had in the >'incomingpstn' context of my extension.conf to the 'default' context. >i.e. > >[default] >exten => 2093,1,Wait(1) >exten => 2093,n,Background(MainMenu) >exten => 1,1,Goto(InternalExtension,2093,1) > >Then the file would play. First of all I don't get why this is?It >doesn't even seem to refer to the code in my sip.conf?I don't get it. >Secondly whilst moving this code to the default context means I can >hear my initial welcome menu, when I press '1' to interrupt the menu >and move to menu option 1 (another sound file) it won't let me >interrupt and I eventually get the error "Timeout but no rule 't' in >context 'default". > >Does anyone have any ides where the problem might be? > >Many thanks, >Aisling. > > > >-------------------Legal Disclaimer--------------------------------------- > >The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >. > > >
Hi,
The hostname that you used in your register directive ('provider.ie')
must have a corresponding section in sip.conf. In your example, you used
'[provider-in]'. If that is what you actually used, then this might
explain why your incoming goes to the default context because it
couldn't find its own "section". Try renaming
'[provider-in]' to
'[provider.ie]'.
-kokmeng.
Aisling O'Driscoll wrote:
>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 when I create an
>entry for them in sip.conf I set only one context. Also to allow for
>incoming calls from my provider it seems I must direct the calls
>firstly to a ?dummy? extension.
>
>sip.conf
>
>register => username:password@provider.ie/2093
>
>[provider-in]
>type=peer
>host=sip.provider.ie
>context=onecontext
>
>[2092]
>type=peer
>other stuff
>context=onecontext
>
>So the dummy extension here is ?2093? and 2092 is a phone who
>registers with SER and when SER redirects to Asterisk uses the
>?onecontext? context.
>
>Now in my extensions.conf ?onecontext? includes other contexts. This
>is how I get access to conference calls, creating IVR menus etc. Also
>the main purpose of ?onecontext? is to allow outgoing access to the
>PSTN.
>
>[onecontext]
>include => createmenu //creating an IVR menu
>include => createconf //creating a conf call
>etc
>include => default //used for voicemail
>
>[createmenu]
>;does something
>
>[createconf]
>;does something
>
>;outgoing calls ? main purpose of onecontext
>exten => _X.,1,Dial(SIP/${EXTEN}@provider-out)
>exten => _X.,2,Hangup
>
>[default]
>
>;mailbox for 2092 and other users
>
>
>Now this is where the problems start! For incoming calls I tried to
>do ?include => incomingpstn? in ?onecontext? which I thought would
>call a new context called ?incomingpstn? which would have an entry
>for the dummy user. i.e.
>
>[incomingpstn]
>
>exten => 2093,1,Wait(1)
>exten => 2093,n,Background(MainMenu)
>exten => 1,1,Goto(InternalExtension,2093,1) //directs to another
>context called Internal Extension
>
>I also changed the [provider-in] for context=incomingpstn in my
>sip.conf. However this didn?t work and I kept getting directed to the
>voicemail of my pstn provider. The ONLY way I could get the incoming
>calls working was to add the contents of the ?incomingpstn? context
>to the default context i.e.
>
>[default]
>
>exten => 2093,1,Wait(1)
>exten => 2093,n,Background(MainMenu)
>exten => 1,1,Goto(InternalExtension,2093,1) //directs to another
>context called Internal Extension
>
>With this I can hear the MainMenu when I dial my DDI but I can?t seem
>to interrupt to divert to another submenu. In the testing that I have
>done the user that is making the call is 2092 registered with SER. If
>I change the context of 2092 directly in sip.conf to incomingpstn,
>then I can hear the menu and interrupt to go to the submenu. But
>obviously then I don?t have access to the other features in Asterisk.
>The point is that I?m stumped as to why it only works in the default
>context and if this is the case how do I get it to call the submenu.
>
>This is what comes up on my asterisk console:
>-- Executing Dial (?SIP/2092-2829?, ?SIP/021123456@provider-out?) in
>new stack
>-- Called 021123456@provider-out
>-- Playing ?MainMenu? (language ?en?)
>-- other messages (not relevant I think)
>== Spawn extension (outgoing, 021123456, 1) exited non-zero on
>?SIP/2092-5837?
>== Spawn extension (default, 2093, 2) exited non zero etc etc
>
>I?m very stuck on this and can?t figure it out.
>Any help appreciated.
>
>Many thanks,
>Aisling.
>
>-----Original Message-----
>From: asterisk-users-bounces@lists.digium.com
>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
>Giovanni Miano
>Sent: 05 January 2006 21:09
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] Incoming PSTN Calls
>
>Is Exist "InternalExtension" context ? and 2093 exten ?
>2006/1/5, Aisling < ashling.odriscoll@cit.ie>:
>Hi all,
>
>I am having difficulty getting incoming PSTN calls working. I have
>set up an account with a third party provider. In my system, the user
>register with SER and use Asterisk for PSTN access, voicemail etc
>
>My provider told me to change my sip.conf as follows
>
>register => username:password@sip.blueface.ie/2093
>
>; To receive incoming calls specify this block and replace
>"yourcontext" for your dial plan.
>[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 it to work is to copy the code I had in the
>'incomingpstn' context of my extension.conf to the 'default'
context.
>i.e.
>
>[default]
>exten => 2093,1,Wait(1)
>exten => 2093,n,Background(MainMenu)
>exten => 1,1,Goto(InternalExtension,2093,1)
>
>Then the file would play. First of all I don't get why this is?It
>doesn't even seem to refer to the code in my sip.conf?I don't get
it.
>Secondly whilst moving this code to the default context means I can
>hear my initial welcome menu, when I press '1' to interrupt the menu
>and move to menu option 1 (another sound file) it won't let me
>interrupt and I eventually get the error "Timeout but no rule
't' in
>context 'default".
>
>Does anyone have any ides where the problem might be?
>
>Many thanks,
>Aisling.
>
>
>
>-------------------Legal Disclaimer---------------------------------------
>
>The above electronic mail transmission is confidential and intended only for
the person to whom it is addressed. Its contents may be protected by legal
and/or professional privilege. Should it be received by you in error please
contact the sender at the above quoted email address. Any unauthorised form of
reproduction of this message is strictly prohibited. The Institute does not
guarantee the security of any information electronically transmitted and is not
liable if the information contained in this communication is not a proper and
complete record of the message as transmitted by the sender nor for any delay in
its receipt.
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>