Re: [asterisk-users] Context for 302 Moved response
Hi Joshua, Thanks for the reply. In this case we get a special SIP header in the 302, but I guess we'll need to find another solution to use it. On Wed, 27 Apr 2022 at 21:27, Joshua C. Colp wrote: > On Wed, Apr 27, 2022 at 5:33 AM David Cunningham < > dcunning...@voisonics.com> wrote: > >> Hi Jon, >> >> Thank you for the reply. We wanted to read a particular SIP header in the >> 302 Moved response, but it seems that Asterisk creates a Local channel for >> the redirected call and the SIP_HEADER() function isn't available, so we >> can't really do what we wanted at all. >> > > Neither chan_sip or chan_pjsip provide such ability even if you had access > to the SIP or PJSIP channel. SIP_HEADER() gets headers from an incoming > INVITE, same for PJSIP_HEADER(). > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context for 302 Moved response
On Wed, Apr 27, 2022 at 5:33 AM David Cunningham wrote: > Hi Jon, > > Thank you for the reply. We wanted to read a particular SIP header in the > 302 Moved response, but it seems that Asterisk creates a Local channel for > the redirected call and the SIP_HEADER() function isn't available, so we > can't really do what we wanted at all. > Neither chan_sip or chan_pjsip provide such ability even if you had access to the SIP or PJSIP channel. SIP_HEADER() gets headers from an incoming INVITE, same for PJSIP_HEADER(). -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context for 302 Moved response
Hi Jon,
Thank you for the reply. We wanted to read a particular SIP header in the
302 Moved response, but it seems that Asterisk creates a Local channel for
the redirected call and the SIP_HEADER() function isn't available, so we
can't really do what we wanted at all.
Thanks anyway!
On Wed, 27 Apr 2022 at 18:57, Jon Bonilla (Manwe)
wrote:
> El Wed, 27 Apr 2022 12:27:03 +1200
> David Cunningham escribió:
>
> > Hello,
> >
> > Does anyone know of a way to have a call go to a particular context when
> a
> > 302 Moved is received in response to an invite? This is with chan_sip. We
> > tried setting __TRANSFER_CONTEXT but it didn't seem to have any effect.
> > Basically if a remote device returns a 302 Moved we want to send the call
> > somewhere different to all other calls.
> >
> > Thanks very much,
> >
>
>
> You can detect a 302 in the dialplan. Not perfect but does the job.
>
> same => n,GotoIf($[${EXISTS(${FORWARDERNAME})}]?sipcfu)
>
>
> --
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> https://pekepbx.com
>
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New Zealand: +64 (0)28 2558 3782
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Re: [asterisk-users] Context for 302 Moved response
El Wed, 27 Apr 2022 12:27:03 +1200
David Cunningham escribió:
> Hello,
>
> Does anyone know of a way to have a call go to a particular context when a
> 302 Moved is received in response to an invite? This is with chan_sip. We
> tried setting __TRANSFER_CONTEXT but it didn't seem to have any effect.
> Basically if a remote device returns a 302 Moved we want to send the call
> somewhere different to all other calls.
>
> Thanks very much,
>
You can detect a 302 in the dialplan. Not perfect but does the job.
same => n,GotoIf($[${EXISTS(${FORWARDERNAME})}]?sipcfu)
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[asterisk-users] Context for 302 Moved response
Hello, Does anyone know of a way to have a call go to a particular context when a 302 Moved is received in response to an invite? This is with chan_sip. We tried setting __TRANSFER_CONTEXT but it didn't seem to have any effect. Basically if a remote device returns a 302 Moved we want to send the call somewhere different to all other calls. Thanks very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] context local: unexpected KW _LOCAL
Hi,
Is the local context a reserved word in extensions.ael?
If so, what is it used for?
Can I define 'context local {};' somehow?
This is the error I'm getting:
ERROR[24659] ael.y: File: /etc/asterisk/extensions.ael, Line 67, Cols:
9-13: Error: syntax error, unexpected KW
_LOCAL, expecting 'default' or word
I don't mind using a different context name but would simply like to know why
this error shows up.
Thanks,
Vieri
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Re: [asterisk-users] context problem
Jonas Kellens wrote: [snip] register = 119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten = _52525252,1,NoOp(context TRUNKin - 52525252) exten = _52525252,n,GoTo(blabla,52525252,1) exten = _59595959,1,NoOp(context TRUNKin - 59595959) exten = _59595959,n,GoTo(blablabla,59595959,1) Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? Could you try removing the leading '_', as you seem to be expecting the exact number? Try that and let us know. Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context problem
On 01/20/2011 04:43 PM, Jose P. Espinal wrote: Jonas Kellens wrote: [snip] register = 119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 [TRUNKin] exten = _52525252,1,NoOp(context TRUNKin - 52525252) exten = _52525252,n,GoTo(blabla,52525252,1) exten = _59595959,1,NoOp(context TRUNKin - 59595959) exten = _59595959,n,GoTo(blablabla,59595959,1) Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? Could you try removing the leading '_', as you seem to be expecting the exact number? Try that and let us know. Regards, Hello, I have tried that yet. It did not make any difference... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context problem
On 01/20/2011 04:29 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, January 20, 2011 9:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] context problem
Hello list,
Asterisk 1.6.16.1
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
mailto:119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
mailto:119909:pas...@sip.prov.org/59595959
[119909]
type=friend
host=sip.prov.org
username=119909
defaultuser=119909
secret=passwd
context=TRUNKin
extensions.conf :
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten = _59595959,1,NoOp(context TRUNKin - 59595959)
exten = _59595959,n,GoTo(blablabla,59595959,1)
Problem :
the call always enters : exten = _52525252
and never : exten = _59595959
Why is that ??
Kind regards,
Jonas.
Because this an incoming call. What you are trying to accomplish
should be done via ex-girlfriend logic. The way your dialplan is
set up, it assumes you are dialing 525225252 or 59595959 instead of
receiving a call. Here is how the incoming should read
[TRUNKin]
- exten = s,1,answer
- exten = s/52525252,n,Goto(blabla,52525252,1)
- exten = s/59595959,n,Goto(blabla,59595959,1)
- exten = s,n,verbose(call is not from 5252 or 5959)
Hello,
the following is not working :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s/52525252,n,GoTo(blabla,52525252,1)
exten = s/59595959,n,GoTo(blablabla,59595959,1)
exten = s,n,NoOp(nothing)
CLI shows :
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] --
Executing [s@TRUNKin:1] NoOp(SIP/119909-0688, context TRUNKin -
s) in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] --
Executing [s@TRUNKin:2] NoOp(SIP/119909-0688, 775006 775006)
in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Auto
fallthrough, channel 'SIP/119909-0688' status is 'UNKNOWN'
What else can I try ?
Kind regards,
Jonas.
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Re: [asterisk-users] context problem
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} : .*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} : .*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
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Re: [asterisk-users] context problem
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, January 20, 2011 9:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011 04:29 PM, Danny Nicholas wrote:
_
size=2 width=100% align=center
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, January 20, 2011 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] context problem
Hello list,
Asterisk 1.6.16.1
I have the following registrations :
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[119909]
type=friend
host=sip.prov.org
username=119909
defaultuser=119909
secret=passwd
context=TRUNKin
extensions.conf :
[TRUNKin]
exten = _52525252,1,NoOp(context TRUNKin - 52525252)
exten = _52525252,n,GoTo(blabla,52525252,1)
exten = _59595959,1,NoOp(context TRUNKin - 59595959)
exten = _59595959,n,GoTo(blablabla,59595959,1)
Problem :
the call always enters : exten = _52525252
and never : exten = _59595959
Why is that ??
Kind regards,
Jonas.
Because this an incoming call. What you are trying to accomplish should be
done via ex-girlfriend logic. The way your dialplan is set up, it assumes
you are dialing 525225252 or 59595959 instead of receiving a call. Here
is how the incoming should read
[TRUNKin]
exten = s,1,answer
exten = s/52525252,n,Goto(blabla,52525252,1)
exten = s/59595959,n,Goto(blabla,59595959,1)
exten = s,n,verbose(call is not from 5252 or 5959)
Hello,
the following is not working :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s/52525252,n,GoTo(blabla,52525252,1)
exten = s/59595959,n,GoTo(blablabla,59595959,1)
exten = s,n,NoOp(nothing)
CLI shows :
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Executing
[s@TRUNKin:1] NoOp(SIP/119909-0688, context TRUNKin - s) in new
stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Executing
[s@TRUNKin:2] NoOp(SIP/119909-0688, 775006 775006) in new stack
[Jan 20 16:54:17] VERBOSE[26980] pbx.c: [Jan 20 16:54:17] -- Auto
fallthrough, channel 'SIP/119909-0688' status is 'UNKNOWN'
What else can I try ?
Kind regards,
Jonas.
The call is coming through with the ID 119909 from both trunks. You need to
be able to register the trunks as 119909 and some other number (119910?) or
otherwise you will have to query the SIP headers to get the actual
information from the duplicated trunks (maybe an AGI?)
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Re: [asterisk-users] context problem
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} : .*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} : .*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
Hello,
this is the result when using your config :
[Jan 20 17:33:50] -- Executing [s@TRUNKin:1]
NoOp(SIP/119909-06d7, context TRUNKin - s) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:2]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:3]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:4]
NoOp(SIP/119909-06d7, sip:s@11.11.12.112) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:5]
NoOp(SIP/119909-06d7, ) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:6]
NoOp(SIP/119909-06d7, 775006) in new stack
dialplan :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${SIP_HEADER(TO)})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID(num)})
Kind regards,
Jonas.
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Re: [asterisk-users] context problem
I always thought the last bit (after the /) is where the context in
sip.conf landed.
What about:
(sip.conf)
register = 119909:pas...@sip.prov.org/52525252
register = 119909:pas...@sip.prov.org/59595959
[52525252]
...
context = TRUNKin52
...
[59595959]
...
context = TRUNKin59
...
And split them out in extensions.conf?
I have a suspicion that you have 'context=TRUNKin' under the '[default]'
section of sip.conf - which is why they are hitting there in the first
place.
Then again, I have been known to be wrong ;)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 20 January 2011 16:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] context problem
On 01/20/2011 05:23 PM, Jeroen Eeuwes wrote:
Hi Jonas,
What else can I try ?
Yeah, Asterisk always assumes that from 1 ip address there can only be
inbound number. Not very user-friendly.
I think I've used something like this:
exten = s,1,Set(CALL-TO=${SIP_HEADER(TO)})
exten = s,n,Set(CALL-FROM=${CALLERIDNUM})
exten = s,n,GotoIf($[${CALL-TO} :
.*52525252.*]?TRUNKin,52525252,1)
exten = s,n,GotoIf($[${CALL-TO} :
.*59595959.*]?TRUNKin,59595959,1)
exten = s,n,etcetera
Best regards,
Jeroen Eeuwes
--
Hello,
this is the result when using your config :
[Jan 20 17:33:50] -- Executing [s@TRUNKin:1]
NoOp(SIP/119909-06d7, context TRUNKin - s) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:2]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:3]
NoOp(SIP/119909-06d7, 775006 775006) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:4]
NoOp(SIP/119909-06d7, sip:s@11.11.12.112) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:5]
NoOp(SIP/119909-06d7, ) in new stack
[Jan 20 17:33:50] -- Executing [s@TRUNKin:6]
NoOp(SIP/119909-06d7, 775006) in new stack
dialplan :
exten = s,1,NoOp(context TRUNKin - s)
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${CALLERID(all)})
exten = s,n,NoOp(${SIP_HEADER(TO)})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERID(num)})
Kind regards,
Jonas.
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Re: [asterisk-users] context problem
On 01/20/2011 10:58 AM, Jonas Kellens wrote: [snip] I have the following registrations : register = 119909:pas...@sip.prov.org/52525252 mailto:119909:pas...@sip.prov.org/52525252 register = 119909:pas...@sip.prov.org/59595959 mailto:119909:pas...@sip.prov.org/59595959 [snip] Problem : the call always enters : exten = _52525252 and never : exten = _59595959 Why is that ?? I may be wrong here, but I think you can only register once. The last registration received will overwrite the first one. You will need to specify a second entry and register that one separately. This is the same reason you cannot register two devices to the same extension. Have you checked the logs and verified that the SIP provider actually sends 59595959 when you dial that number? Or do you get sent 52525252 no matter what? Someone please correct me if I am wrong here. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context problem
I may be wrong here, but I think you can only register once. The last registration received will overwrite the first one. You will need to specify a second entry and register that one separately. This is the same reason you cannot register two devices to the same extension. Yes, that's very likely what is happening. The provider is seeing two SIP registrations arrive, for the same provider account, from the same peer at the same IP address. It is very likely that the second registration is (by design) replacing the first. Then, whenever someone dials a DID associated with this provider account, the provider is routing the call based on the information in the most current registration... it's either going to the context and extension specified in that registration (if their is one) or to the s extension for the relevant context. (Some providers do allow multiple registration for a given account, and will INVITE all of them when an incoming call arrives, but (if I recall correctly) the registrations have to come from different IP addresses (and perhaps different peers) in order to be recognized as being distinct.) There are probably several ways around this: (1) Use two different provider accounts, and associate each DID with a different account. Use two register statements, one per account, and specify different routing extensions on these. (2) Use a provider which will let you register once, and will pass through the DID number which was dialed as the target extension. (3) Use a provider which will let you set up your DIDs for hardwired-IP-address routing (i.e. no register being required) and who passes through the DID as the extension to be called. I recently set up an account with Vitelity, and they support option (3). I simply entered the public IP address of my SIP server for the routing, and everything works correctly... the incoming INVITE requests say sip:MYDID@MYIPADDRESS. Asterisk then uses MYDID as the desired extension in my dialplan, and routes the call appropriately. I'd suggest that the OP ask the current SIP provider whether they handle (2) i.e. whether it's possible for different DIDs associated with a single account to have different information in the INVITE requests sent to the registered client. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Context issue
Hi,
Running 1.4.15. I've a SIP user as below. My default context in
sip.conf is [incomming_pstn]
I'm having trouble with inbound calls going to the wrong context.
[test-ubi]
username=test-ubi
type=friend
secret=XXX
host=dynamic
canreinvite=no
context=testinbound
nat=yes
allow=ulaw
allow=gsm
allow=alaw
qualify=no
the testinbound context includes the code to prepend a 2 to the CLI
before passing it onto another context
[testinbound]
exten =
_,1,ExecIF($[${RECORDSIP}=TRUE],Monitor,wav|${TIMESTAMP}-${CALLE
RID(num)}-${EXTEN}-${UNIQUEID}.WAV)
exten = _,n,NoOp(REWRITE CALLERID)
exten = _,n,ExecIf($[ ${LEN(${CALLERID(num)})} = 4
]|Set|CALLERID(num)=2${CALLERID(num)})
exten = _,n,Goto(local,${EXTEN},1)
However, when a call comes in, its being passed to the
[incomming_pstn] context instead of [testinbound].
The Outbound server is dialling:
-- Executing [114...@from-sip-uk:2]
Dial(SIP/235012071833427-0a068a18, SIP/test-ubi/4201|40|r) in new
stack
-- Called test-ubi/4201
And that test-ubi account on there has the same SIP account setup.
The inbound server seems to skip the testinbound context completely
though, jumping straight to incomming_pstn, but I've no idea why.
I think it should be going to the context defined in test-ubi
ubiphone*CLI
-- Executing [4...@incomming_pstn:1]
Answer(SIP/192.168.50.132-b7d4f6b0, ) in new stack
-- Executing [4...@incomming_pstn:2]
SayDigits(SIP/192.168.50.132-b7d4f6b0, 2333) in new stack
-- SIP/192.168.50.132-b7d4f6b0 Playing 'digits/2' (language 'en')
-- SIP/192.168.50.132-b7d4f6b0 Playing 'digits/3' (language 'en')
.
But any idea why ???
Thanks,
Adrian
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Re: [asterisk-users] Context issue
How odd... If I specify the host=dynamic then it goes to the wrong context. If I specify the host=192.168.50.132, then it goes to the correct context. If I don't specify the host at all, then it also goes to the correct context... (but then of course I can't use that account for outbound calls..) Adrian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context issue
El 12/11/10 12:13, Adrian Marsh escribió: How odd... If I specify the host=dynamic then it goes to the wrong context. If I specify the host=192.168.50.132, then it goes to the correct context. If I don't specify the host at all, then it also goes to the correct context... (but then of course I can't use that account for outbound calls..) Adrian If you use host = dynamic, I think the device must register with Asterisk for incoming calls go to the right context. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Context vs. Custom Context
Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? Thanks a lot, Alejandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context vs. Custom Context
Alejandro Cabrera Obed wrote: Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? That sounds like a module for FreePBX or some other GUI. A context in Asterisk is just a context. There are no weights. If you define the same context twice you will likely get some sort of WARNING on the Asterisk console I think. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context vs. Custom Context
Yes, Custom Context is a module from FreePBX in order to define calling routes. Thanks. 2010/3/22 Leif Madsen leif.mad...@asteriskdocs.org Alejandro Cabrera Obed wrote: Dear all, if I use the CustomContext module in Asterisk in order to create new customized contexts for my extensions to managed outbound/inbound calls, do these custom contexts replace the original context defined in sip.conf, like context=from-internal ??? In other words, does a custom context have a bigger priority than context ??? That sounds like a module for FreePBX or some other GUI. A context in Asterisk is just a context. There are no weights. If you define the same context twice you will likely get some sort of WARNING on the Asterisk console I think. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context vs. Custom Context
Alejandro Cabrera Obed wrote: Yes, Custom Context is a module from FreePBX in order to define calling routes. I'd suggest using the FreePBX forums as I imagine the majority of people responding on this list are vanilla Asterisk users. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Context Switches and Load Average spike - Asterisk Version 1.4.22
I am running Asterisk V 1.4.22 Twice during the last two days the Context Switches on our box has gone from about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to 0.35 then spike again. When connecting to the console 'core show channels' will list the channels but not total calls. 'restart now' had no effect, the only way to stop Asterisk is to kill the process. Once Asterisk is killed, everything returned to normal, for about 20 hours, then it started again. The server is a dual - quad core machine. Linux has been up over 380 days. Has anyone experienced this before? -- -Thermal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Hello, I think there is a problem with chars in the extension name. I have a similar issue if i try to use my my que management macro with a extension with characters. On Mon, Aug 10, 2009 at 3:16 PM, Tarek Sawahtareksa...@hotmail.com wrote: i faced the same problem with callcentric.. when i register i had to add the extension .. like this egister = 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID which caused my context to go to the default context and never use the one i already setup.. so removing the extension in the registration string will solve the issue for me.. and i think it will do the same for you. regards -- AHD Tarek Sawah Date: Mon, 10 Aug 2009 12:55:41 +0200 From: patr...@erdbeere.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] context does not work
Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device On Aug 10, 2009, at 6:55 AM, Patrick Plattes patr...@erdbeere.net wrote: Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Thanks for the fast reply, but it does not help :-(. Bye, Patrick On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote: Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Patrick Plattes wrote: extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) What does dialplan show testing output? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
What does dialplan show testing output? [ Context 'testing' created by 'pbx_config' ] '261' = 1. Noop(261) [SIP] '262' = 1. Noop(262) [SIP] '263' = 1. Noop(263) [SIP] '264' = 1. Noop(264) [SIP] '_8001187e0' = 1. Dial(SIP/263) [pbx_config] -= 5 extensions (5 priorities) in 1 context. =- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Try putting exten = 8001187e0,1,Dial(SIP/263) in the [default]-context. I have the same issue. Apparently your SIP-provider send calls to your Asterisk-box from multiple IP's so that Asterisk cannot match the inbound call on source IP and therefore sends it to the default-context. Jonas. On Mon, 2009-08-10 at 13:26 +0200, Patrick Plattes wrote: Thanks for the fast reply, but it does not help :-(. Bye, Patrick On Mon, Aug 10, 2009 at 1:01 PM, Alex Balashovabalas...@evaristesys.com wrote: Try prefix your extension in extensions.conf with _, e.g. exten = _123,1,... -- Sent from mobile device ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
jonas kellens wrote: I have the same issue. Apparently your SIP-provider send calls to your Asterisk-box from multiple IP's so that Asterisk cannot match the inbound call on source IP and therefore sends it to the default-context. I'd second this suggestion. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
Hi Jonas, that works fine, but I think its just a work arround and not a real fix :-). For the moment it is okay and I'll try to fix the error next days. Thanks, Patrick Plattes ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
i faced the same problem with callcentric.. when i register i had to add the extension .. like this egister = 1777MYCCID:SUPERSECRET@callcentric.com/1777MYCCID which caused my context to go to the default context and never use the one i already setup.. so removing the extension in the registration string will solve the issue for me.. and i think it will do the same for you. regards -- AHD Tarek Sawah Date: Mon, 10 Aug 2009 12:55:41 +0200 From: patr...@erdbeere.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live™: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] context does not work
V1.6.1.0 [9290740] type = peer username = 9290740 fromuser = 9290740 secret = you-wish! host = sipgate.co.uk fromdomain = sipgate.co.uk insecure = port,invite context = inbound caninvite = no canreinvite = no nat = yes disallow = all allow = ulaw allow = alaw dtmfmode = info qualify = 5000 That works for me. Any inbound call to my 9290740 number goes to my inbound context and does what it should. PS - Don't forget to do a 'sip reload' when you change the sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 13:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] context does not work Hi Andrew, it didn't help. Which version of Asterisk do you use? Thanks On Mon, Aug 10, 2009 at 1:55 PM, Andrew Thomasa...@datavox.co.uk wrote: Underscore won't help as that's for pattern matching. Under the sip conf, have you tried adding 'fromuser=8001187e0' to the [8001187e0] bit? I have this in my Sipgate setup and it works. Worth a try. Cheers Andy -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Plattes Sent: 10 August 2009 11:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] context does not work Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register = 8001187e0:passw...@sipgate.de/8001187e0 [8001187e0] type=friend context=testing secret=password host=dynamic caninvite=no canreinvite=no qualify=yes extensons.conf: [testing] exten = 8001187e0,1,Dial(SIP/263) I don't know whats wrong here :-( Does anyone see my (usually) stupid error. Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Context documentation for the newbie!
Bsumrall, Take a look on this document, http://bef.eventphone.de/a/Ast.%20C.%20I._files/ast-ci-draft1.pdf /Mats On 6/1/07, C F [EMAIL PROTECTED] wrote: I can give the following example, let me know if it helps. Mr 1 has a child Mr 10 and another child Mr 11, now Mr 10 has Mr 100 and Mr 11 has Mr 111. Mr 10 adopts Mr 111. Also Mr 88 adopts Mr 10. Which brings us to the family tree, if you are a child of one, you are a grandchild of that ones parent, and as such included in that tree. Now one of the children could be adopted by some other parent as well, which makes that child a child of another parent hence a grandchild of that parents parent. Subistute child and adopt for include =, and Mr for context so you got: [1] include = 10 include = 11 [10] include = 100 include = 111 [11] include = 111 [88] include = 10 Within each context you got the instruction code, which is an extension (exten) prioritized with numbers (or n for next number). The instructions are executed one after the other, unless a jump is encountered. Each extension is a pointer within that context that starts the instruction set. In Asterisk one starts in a context, when an extension is called (by dialing, or s when the extension number wasn't given) Asterisk looks for that extension in that context, if it can't find it there it searches in that contexts family tree, if still no match it searches in default context, if still no match it searches for the i extension in the same order, if still no match then 404 is given. Hope this helps. On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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 ___ --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 ___ --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
[asterisk-users] Context documentation for the newbie!
Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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
Re: [asterisk-users] Context documentation for the newbie!
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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 ___ --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
RE: [asterisk-users] Context documentation for the newbie!
First off, I wanted to thank you for referring me to the O'Reily pdf. It has already helped a lot and now I know exactly where I am going wrong, but still do not have an answer! Almost every example on voip-info.org and O'Reily assume you are using an FXO or FXS card. I am 100% internet based. It hit me like a rock that I need to understand why this affects the channels differently. O'Reily states: [incoming] exten = s,1,Answer( ) exten = s,2,Playback(hello-world) exten = s,3,Hangup( ) If you have a channel or two configured, go ahead and try it out! Simply make a new extensions.conf file with this short dialplan. (If it doesn't work, check the Asterisk console for error messages, and make sure your channels are configured to send inbound calls to the [incoming] context.) Go figure! I am 100% SIP based and zero IAX and I am assuming this effects how asterisk looks a Zapata.conf? So, this would lead to the logical conclusion that if I do not configure incoming in Zapata, I configure it on the teliax authentication portion of sip.conf! It still didn't work. I can see my phone number coming in on the CLI, but zero transition into the basic commands of extentions.conf Teliax got the thing to work before. They simply stripped out everything and put in what appeared to be the exact example on their web site. I was playing around with extensions.conf only and now it doesn't work at all and the conceptual theory doesn't seem to apply when dealing with 100% only SIP vs. FXx So, can anyone point me into the right direction on documentation on understanding the differences of SIP vs. FXx? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mats Karlsson Sent: Thursday, May 31, 2007 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Context documentation for the newbie! http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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 ___ --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
RE: [asterisk-users] Context documentation for the newbie!
Got it basically working, but still need answers as to why SIP is so much different from FXx _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BSumrall Sent: Thursday, May 31, 2007 7:36 AM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Context documentation for the newbie! First off, I wanted to thank you for referring me to the O'Reily pdf. It has already helped a lot and now I know exactly where I am going wrong, but still do not have an answer! Almost every example on voip-info.org and O'Reily assume you are using an FXO or FXS card. I am 100% internet based. It hit me like a rock that I need to understand why this affects the channels differently. O'Reily states: [incoming] exten = s,1,Answer( ) exten = s,2,Playback(hello-world) exten = s,3,Hangup( ) If you have a channel or two configured, go ahead and try it out! Simply make a new extensions.conf file with this short dialplan. (If it doesn't work, check the Asterisk console for error messages, and make sure your channels are configured to send inbound calls to the [incoming] context.) Go figure! I am 100% SIP based and zero IAX and I am assuming this effects how asterisk looks a Zapata.conf? So, this would lead to the logical conclusion that if I do not configure incoming in Zapata, I configure it on the teliax authentication portion of sip.conf! It still didn't work. I can see my phone number coming in on the CLI, but zero transition into the basic commands of extentions.conf Teliax got the thing to work before. They simply stripped out everything and put in what appeared to be the exact example on their web site. I was playing around with extensions.conf only and now it doesn't work at all and the conceptual theory doesn't seem to apply when dealing with 100% only SIP vs. FXx So, can anyone point me into the right direction on documentation on understanding the differences of SIP vs. FXx? _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mats Karlsson Sent: Thursday, May 31, 2007 5:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Context documentation for the newbie! http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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 http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [asterisk-users] Context documentation for the newbie!
Almost every example on voip-info.org and O'Reily assume you are using an FXO or FXS card. I am 100% internet based. This 4 year old article will go a long way in explaing the basics, with examples. John Todd also has had his own heavily commented extensions and other config files online all that time. http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html ___ --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
Re: [asterisk-users] Context documentation for the newbie!
On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: First off, I wanted to thank you for referring me to the O'Reily pdf. I'm glad you found it useful. Almost every example on voip-info.org and O'Reily assume you are using an FXO or FXS card. Yes, we wrote the first edition of the O'Reilly book when the easiest way to get up and running with Asterisk was to use analog phones. It's amazing how quickly thta's changed. The second edition of the book will be out shortly, and has a lot more information on how to setup SIP and IAX devices to talk to Asterisk. -Jared ___ --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
Re: [asterisk-users] Context documentation for the newbie!
I can give the following example, let me know if it helps. Mr 1 has a child Mr 10 and another child Mr 11, now Mr 10 has Mr 100 and Mr 11 has Mr 111. Mr 10 adopts Mr 111. Also Mr 88 adopts Mr 10. Which brings us to the family tree, if you are a child of one, you are a grandchild of that ones parent, and as such included in that tree. Now one of the children could be adopted by some other parent as well, which makes that child a child of another parent hence a grandchild of that parents parent. Subistute child and adopt for include =, and Mr for context so you got: [1] include = 10 include = 11 [10] include = 100 include = 111 [11] include = 111 [88] include = 10 Within each context you got the instruction code, which is an extension (exten) prioritized with numbers (or n for next number). The instructions are executed one after the other, unless a jump is encountered. Each extension is a pointer within that context that starts the instruction set. In Asterisk one starts in a context, when an extension is called (by dialing, or s when the extension number wasn't given) Asterisk looks for that extension in that context, if it can't find it there it searches in that contexts family tree, if still no match it searches in default context, if still no match it searches for the i extension in the same order, if still no match then 404 is given. Hope this helps. On 5/31/07, BSumrall [EMAIL PROTECTED] wrote: Does anyone know where there is better documentation on understanding context relations and priorities with examples? http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Introduction Does tell me anything other than they point to each other. Not how or who comes first or even how to get them to work with each other! ___ --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 ___ --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
Re: [asterisk-users] Context default incoming ENUM
On 07:10, Wed 27 Sep 06, Ronald Wiplinger wrote: I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? If you find out let me know as well. I'm interested in this. I dont think it's possible though, because the call will come in just like any other unauthenticated call. It's not like ENUM is adding sip headers or something. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --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
[asterisk-users] Context default incoming ENUM
I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? bye Ronald ___ --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
[asterisk-users] Context
Dear I have two contexts how could I isolate context A from context B ,in other words I want to ban context A from calling context B Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
Re: [asterisk-users] Context
I have two contexts how could I isolate context A from context B ,in
other words I want to ban context A from calling context B
In sip.conf, define phones/extensions something like this:
[1000]
type=friend
other parameters as needed
context=cust-a
[1001]
type=friend
other parameters as needed
context=cust-a
[2000]
type=friend
other parameters as needed
context=cust-b
[2001]
type=friend
other parameters as needed
context=cust-b
In extensions.conf, define dialplans something like this:
[cust-a]
include=local-extn-cust-a
include=local-calls-a
include=misc-extns
include=no-match
[cust-b]
include=local-extn-cust-b
include=local-calls-b
include=misc-extns
include=no-match
[local-extn-cust-a]
exten = 1000,1,Dial(SIP/1000,15,r)
exten = 1000,2,Voicemail(1000|ug(6))
exten = 1000,102,Voicemail(1000|bg(6))
exten = 1000,103,Hangup
exten = 1001,1,Dial(SIP/1001,15,r)
exten = 1001,2,Voicemail(1001|ug(6))
exten = 1001,102,Voicemail(1001|bg(6))
exten = 1001,103,Hangup
[local-extn-cust-b]
exten = 2000,1,Dial(SIP/2000,15,r)
exten = 2000,2,Voicemail(2000|ug(6))
exten = 2000,102,Voicemail(2000|bg(6))
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,15,r)
exten = 2001,2,Voicemail(2001|ug(6))
exten = 2001,102,Voicemail(2001|bg(6))
exten = 2001,103,Hangup
[local-calls-a] ; outgoing pstn calls for cust-a
exten = _21X,1,Dial(Zap/g1/${EXTEN})
exten = _30X,1,Dial(Zap/g1/${EXTEN})
etc
[local-calls-b] ; outgoing pstn calls for cust-b
exten = _21X,1,Dial(Zap/g2/${EXTEN})
exten = _30X,1,Dial(Zap/g2/${EXTEN})
etc
[misc-extns]
exten = 3912,1,Wait(1)
exten = 3912,2,SayDigits(${CALLERID(num)})
exten = 3912,3,Hangup
[no-match]
exten = _X.,1,Answer
exten = _X.,2,GotoIF($[${EXTEN} != h]?10)
exten = _X.,10,Playback(invalid,skip)
exten = _X.,11,Hangup
In zapata.conf (assuming you have some zap pstn interfaces for each
customer), use something like this:
context=cust-a
other needed parameters
group=1
channel = 1,2
context=cust-b
other needed parameters
group=2
channel = 3,4
The above is a very simple example. Those extensions belonging to cust-a
cannot call those extension belonging to cust-b, and outgoing pstn calls
from each customer uses zap interfaces belonging to each customer.
If you're using [EMAIL PROTECTED], Trixbox, or some other pre-canned
implementation of asterisk, then pose your questions on their respective
support lists.
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Re: [asterisk-users] Context
That is the default behavior. If you don't include the contexts into each other, they can't call each other. On September 11, 2006 03:35, Khaled Chehab wrote: Dear I have two contexts how could I isolate context A from context B ,in other words I want to ban context A from calling context B Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] pgpIXF6QsrpyF.pgp Description: PGP signature ___ --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
[asterisk-users] context
Since I make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is located at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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
Re: [asterisk-users] context
That's the fourth time you've asked the same question in the space of a few hours - please have a little more patience and wait for someone to answer. On 12/07/06, Khaled Chehab [EMAIL PROTECTED] wrote: Since I make call forward to an extension l by default it will attach your DIAL(local/[EMAIL PROTECTED]) from the context from-internal which is linked to a trunk , The script is located at /var/lib/asterisk/agi-bin/dialparties.agi I $dialstring = 'Local/'.$extnum.'@from-internal'; How can I let it find the context ? automatically $context ? Instead of '@from-internal' Please help regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --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 -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --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
RE: [Asterisk-Users] context being ignored by inbound sip call
change context to context=remote in [general] in sip.conf you missing registration of peer :) turby -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of btb Sent: Thursday, February 23, 2006 4:10 AM To: Asterisk Non-Commercial Discussion Users Mailing List - Subject: [Asterisk-Users] context being ignored by inbound sip call hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no in extensions,conf, i have: [remote] exten = 7508,1,DISA(|internal) [internal] exten = 81,1,Dial(SIP/ion,20,tr) exten = 82,1,Dial(SCCP/82,20,tr) exten = 83,1,Dial(SIP/quark,20,tr) exten = 84,1,Dial(SIP/proton,20,tr) exten = 85,1,Dial(SIP/work1,20,tr) exten = 86,1,Dial(IAX2/work2,20,tr) yet when the call arrives, asterisk says: NOTICE[8100]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'default' what am i missing? thanks -ben ___ --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 ___ --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
Re: [Asterisk-Users] context being ignored by inbound sip call
Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. I would never recommend using a type=friend for a service provider connection. You need one peer for calling out and another for receiving calls, or at least add a host=hostname of provider's server to enable matching on IP on incoming calls. The problem here is, as you figured out Jonathan, that this peer section does not match the incoming call. Adding a host=hostname entry will help matching. /Olle ___ --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
Re: [Asterisk-Users] context being ignored by inbound sip call
Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. thanks jonathan- i originally had this entry as type=user, and switched to type=peer after finding the context was being ignored and reading that type=user may/is be(ing) phased out: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer i've tried type=user again (as well as type=peer), with some additional parameters (mostly guesses, because i don't yet fully understand registration): [7508] ;ipkall type = peer host = dynamic dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no insecure = very i gather the ideal method is to know the source ip and source port of the connection from my peer, and include that in the sip config? how can i make asterisk tell me where a connection is coming from? -ben ___ --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
Re: [Asterisk-Users] context being ignored by inbound sip call
On Feb 23, 2006, at 10.43, btb wrote: Johnathan Corgan wrote: btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/ from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. thanks jonathan- i originally had this entry as type=user, and switched to type=peer after finding the context was being ignored and reading that type=user may/is be(ing) phased out: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer i've tried type=user again (as well as type=peer), with some additional parameters (mostly guesses, because i don't yet fully understand registration): [7508] ;ipkall type = peer host = dynamic dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no insecure = very i gather the ideal method is to know the source ip and source port of the connection from my peer, and include that in the sip config? how can i make asterisk tell me where a connection is coming from? so, in answer to my own question, this ended up being what i needed in sip.conf: [ipkall] type = peer host = voiper.ipkall.com dtmfmode = rfc2833 context = remote callerid = ipkall incoming nat = no the key was the host parameter. as soon as i added that, matching occurred and the context was honored. thanks -ben ___ --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
[Asterisk-Users] context being ignored by inbound sip call
hello- i was messing around with a did from ipkall.com, and asterisk seems to be ignoring the context specified in the sip config. in sip.conf, i've added: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no in extensions,conf, i have: [remote] exten = 7508,1,DISA(|internal) [internal] exten = 81,1,Dial(SIP/ion,20,tr) exten = 82,1,Dial(SCCP/82,20,tr) exten = 83,1,Dial(SIP/quark,20,tr) exten = 84,1,Dial(SIP/proton,20,tr) exten = 85,1,Dial(SIP/work1,20,tr) exten = 86,1,Dial(IAX2/work2,20,tr) yet when the call arrives, asterisk says: NOTICE[8100]: pbx.c:1731 pbx_extension_helper: Cannot find extension context 'default' what am i missing? thanks -ben ___ --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
Re: [Asterisk-Users] context being ignored by inbound sip call
btb wrote: [7508] ;ipkall type = peer dtmfmode = rfc2833 context = remote callerid = ipkall incoming 7508 nat = no You've configured this entry as a peer, which is for dialing out, versus as a user, which is for incoming calls. Solution is to change to 'type=user'. If you really need a peer definition, you can use 'type=friend', which will cause * to create both a user and a peer entry for '7508' using the parameters listed. Some parameters are common to both peers and users so it saves space. Personally, I never use the 'type=friend' method, but rather maintain separate peer and user sections for outbound and inbound calls to/from other switches or endpoints. This helps _me_ keep things straight; others (probably most) prefer the combined 'type=friend' method, though. -Johnathan ___ --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
[Asterisk-Users] Context for SIP incoming (newbie question?)
Please help me out with this To which context of the dial-plan does asterisk tries to match incoming calls when acting as a sip client? To be more specific: In extensions.conf Under which context should I place exten = 648064,1,Dial(TECH/peer) for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ? This is because I want to match one sip client to one context, and another sip client into another context. Is it possible? What is the correct way to do it?? Thanks, Alejandro ___ --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
RE: [Asterisk-Users] Context for SIP incoming (newbie question?)
If you have, in sip.conf, a register = blah:[EMAIL PROTECTED]/12345, you would also have: [blah] host=sip.blah.com context=from-blah Then, in extensions.conf, you would have: [from-blah] exten = 12345,1,Dial(whatever) ... Nabeel From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Mejía Evertsz Sent: January 27, 2006 5:42 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Context for SIP incoming (newbie question?) Please help me out with this To which context of the dial-plan does asterisk tries to match incoming calls when acting as a sip client? To be more specific: In extensions.conf Under which context should I place exten = 648064,1,Dial(TECH/peer) for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ? This is because I want to match one sip client to one context, and another sip client into another context. Is it possible? What is the correct way to do it?? Thanks, Alejandro ___ --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
[Asterisk-Users] Context Picker for interception and redirection
Going try my best to explain this and hopefully it will make sense:
We're trying to come up with something that we can only refer to as a
context picker. The idea is that if someone dials 98625551212, the
context picker will direct the call to the proper context based on
the dialing prefix, in this case 9. The context picker would then
re-write the extension and then Goto the proper context based on the
prefix. The context would need to miraculously read a variable set
by the context picker to match the dialed number pattern and execute
the proper Dial. The thing I can't seem to figure out is how to get
the context to read this variable set by the context picker as a
dialstring. For example (not syntactically correct, I know):
[contextpicker]
exten = _9NXXNXX,1,SetVar(L-EXT=${EXTEN:1})
exten = _9NXXNXX,2,GoTo(localoutbound,${L-EXT})
exten = _91NXXNXX,1,SetVar(LD-EXT=${EXTEN:1})
exten = _91NXXNXX,2,GoTo(ldoutbound,${LD-EXT})
exten = _8.,1,SetVar(INOC-EXT=${EXTEN:1})
exten = _8.,2,GoTo(inoc-dba,${INOC-EXT})
[localoutbound]
exten = ${L-EXT},1,Dial(SIP/localdump)
[ldoutbound]
exten = ${L-EXT},1,Dial(SIP/lddump)
[inoc-dba]
exten = ${INOC-EXT},1,Dial(SIP/inocdump)
Does this make sense? Is there a better way to achieve this?
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Re: [Asterisk-Users] Context Picker for interception and redirection
Got it working.. wow.. didn't think it would be this easy:
[test]
; Test SIP user's context
include = contextpicker
[contextpicker]
exten = _9NXXNXX,1,Set(LOCALEXT=${EXTEN:1})
exten = _9NXXNXX,2,GoTo(local-outbound-test,localout,1)
exten = _9NXXNXX,102,NoOp(seq 102 check)
exten = _91NXXNXX,1,Set(LDEXT=${EXTEN:1})
exten = _91NXXNXX,2,GoTo(cheapldprovider-outbound-test,ldout,1)
exten = _91NXXNXX,102,NoOp(seq 102 check)
exten = _8.,1,Set(INOCEXT={$EXTEN:1})
exten = _8.,2,GoTo(inoc-dba,s,1)
exten = _8.,102,NoOp(seq 102 check)
[local-outbound-test]
exten = localout,1,Dial(${LOCALIAXOUT}/${LOCALEXT},,r)
exten = localout,2,Playback(last-error-was)
exten = localout,3,SayDigits(${CAUSECODE})
exten = localout,4,Playback(tt-somethingwrong)
exten = localout,5,Hangup
exten = localout,102,NoOp(seq 102 check)
[cheapldprovider-outbound-test]
exten = ldout,1,Dial(${LDIAXOUT}/${LDEXT},,r)
exten = ldout,2,Playback(last-error-was)
exten = ldout,3,SayDigits(${CAUSECODE})
exten = ldout,4,Playback(tt-somethingwrong)
exten = ldout,5,Hangup
exten = ldout,102,NoOp(seq 102 check)
On 14-Dec-05, at 5:56 PM, Jason Lixfeld wrote:
Going try my best to explain this and hopefully it will make sense:
We're trying to come up with something that we can only refer to as
a context picker. The idea is that if someone dials 98625551212,
the context picker will direct the call to the proper context
based on the dialing prefix, in this case 9. The context picker
would then re-write the extension and then Goto the proper context
based on the prefix. The context would need to miraculously read a
variable set by the context picker to match the dialed number
pattern and execute the proper Dial. The thing I can't seem to
figure out is how to get the context to read this variable set by
the context picker as a dialstring. For example (not syntactically
correct, I know):
[contextpicker]
exten = _9NXXNXX,1,SetVar(L-EXT=${EXTEN:1})
exten = _9NXXNXX,2,GoTo(localoutbound,${L-EXT})
exten = _91NXXNXX,1,SetVar(LD-EXT=${EXTEN:1})
exten = _91NXXNXX,2,GoTo(ldoutbound,${LD-EXT})
exten = _8.,1,SetVar(INOC-EXT=${EXTEN:1})
exten = _8.,2,GoTo(inoc-dba,${INOC-EXT})
[localoutbound]
exten = ${L-EXT},1,Dial(SIP/localdump)
[ldoutbound]
exten = ${L-EXT},1,Dial(SIP/lddump)
[inoc-dba]
exten = ${INOC-EXT},1,Dial(SIP/inocdump)
Does this make sense? Is there a better way to achieve this?
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Re: [Asterisk-Users] Context Picker for interception and redirection
You could also use macros, looks a bit cleaner perhaps.exten = _9NXXNX,1,Macro(local-outbound-test,${EXTEN:1})exten = _91NXXNX,1,Macro(cheapprovider-outbound,${EXTEN:1})exten = _8., 1, Macro( ioc, ${EXTEN:1})
[macro-local-outbound-test]exten = s,1,Dial(${LOCALIAXOUT}/${ARG1},,r)exten = s,2,Playback(last-error-was)exten = s,3,SayDigits(${CAUSECODE})exten = s,4,Playback(tt-somethingwrong)
exten = s,5,Hangupexten = s,102,NoOp(seq 102 check)[macro-cheapprovider-outbound]...[macro-ioc]...On 12/14/05, Jason Lixfeld
[EMAIL PROTECTED] wrote:
Got it working.. wow..didn't think it would be this easy:[test]; Test SIP user's contextinclude = contextpicker[contextpicker]exten = _9NXXNXX,1,Set(LOCALEXT=${EXTEN:1})exten = _9NXXNXX,2,GoTo(local-outbound-test,localout,1)
exten = _9NXXNXX,102,NoOp(seq 102 check)exten = _91NXXNXX,1,Set(LDEXT=${EXTEN:1})exten = _91NXXNXX,2,GoTo(cheapldprovider-outbound-test,ldout,1)exten = _91NXXNXX,102,NoOp(seq 102 check)
exten = _8.,1,Set(INOCEXT={$EXTEN:1})exten = _8.,2,GoTo(inoc-dba,s,1)exten = _8.,102,NoOp(seq 102 check)[local-outbound-test]exten = localout,1,Dial(${LOCALIAXOUT}/${LOCALEXT},,r)
exten = localout,2,Playback(last-error-was)exten = localout,3,SayDigits(${CAUSECODE})exten = localout,4,Playback(tt-somethingwrong)exten = localout,5,Hangupexten = localout,102,NoOp(seq 102 check)
[cheapldprovider-outbound-test]exten = ldout,1,Dial(${LDIAXOUT}/${LDEXT},,r)exten = ldout,2,Playback(last-error-was)exten = ldout,3,SayDigits(${CAUSECODE})exten = ldout,4,Playback(tt-somethingwrong)
exten = ldout,5,Hangupexten = ldout,102,NoOp(seq 102 check)On 14-Dec-05, at 5:56 PM, Jason Lixfeld wrote: Going try my best to explain this and hopefully it will make sense:
We're trying to come up with something that we can only refer to as a context picker.The idea is that if someone dials 98625551212, the context picker will direct the call to the proper context
based on the dialing prefix, in this case 9.The context picker would then re-write the extension and then Goto the proper context based on the prefix.The context would need to miraculously read a
variable set by the context picker to match the dialed number pattern and execute the proper Dial.The thing I can't seem to figure out is how to get the context to read this variable set by the context picker as a dialstring.For example (not syntactically
correct, I know): [contextpicker] exten = _9NXXNXX,1,SetVar(L-EXT=${EXTEN:1}) exten = _9NXXNXX,2,GoTo(localoutbound,${L-EXT}) exten = _91NXXNXX,1,SetVar(LD-EXT=${EXTEN:1})
exten = _91NXXNXX,2,GoTo(ldoutbound,${LD-EXT}) exten = _8.,1,SetVar(INOC-EXT=${EXTEN:1}) exten = _8.,2,GoTo(inoc-dba,${INOC-EXT}) [localoutbound] exten = ${L-EXT},1,Dial(SIP/localdump)
[ldoutbound] exten = ${L-EXT},1,Dial(SIP/lddump) [inoc-dba] exten = ${INOC-EXT},1,Dial(SIP/inocdump) Does this make sense?Is there a better way to achieve this?
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[Asterisk-Users] Context confict question??
Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten = 600,1,Dial(ZAP/1,20) include = small-business [small-business] exten = 600,1,Dial(ZAP/2,15) Thanks ___ --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
Re: [Asterisk-Users] Context confict question??
Hi, The one in [big-business] has higher priority than the one in [small-business]Included context has lower priority. Hope this helps. Andy On 12/2/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi,If I have an extension in a context and I have another context with thesame extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension ineach context[big-business]exten = 600,1,Dial(ZAP/1,20)include = small-business[small-business]exten = 600,1,Dial(ZAP/2,15) Thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [Asterisk-Users] Context confict question??
On Fri, 2005-12-02 at 16:07, Chuck Bunn wrote: Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten = 600,1,Dial(ZAP/1,20) include = small-business [small-business] exten = 600,1,Dial(ZAP/2,15) Thanks It's never caused any problems for me. Enjoy yourself. Jon Carnes ___ --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
Re: [Asterisk-Users] Context confict question??
Hi, When you say it has a higher priority what does that mean?? Does that mean that a call to extension 600 always goes to the higher priority unless it is busy? Thanks Andy Kuo wrote: Hi, The one in [big-business] has higher priority than the one in [small-business] Included context has lower priority. Hope this helps. Andy On 12/2/05, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten = 600,1,Dial(ZAP/1,20) include = small-business [small-business] exten = 600,1,Dial(ZAP/2,15) Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.11/191 - Release Date: 12/2/2005 ___ --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
[Asterisk-Users] Context mix-up
I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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
[Asterisk-Users] Context mix-up
Help, my messages to the list disappear. I will post a follow-up to this message in just a sec. ___ --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
[Asterisk-Users] Context mix-up
I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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
Re: [Asterisk-Users] Context mix-up
Thor: All your messages seem to be making it to the list ok - I've seen this email at least 3 times. Are you perhaps blocking the list somewhere in your anti-spam setup? Roger Thor Atle Rustad wrote: I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --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
Re: [Asterisk-Users] Context mix-up
I have now received the messages I sent today, this seems to have happened after I updated some settings at digium.com's list server. Why that would matter, I don't know. According to the list server, I had a bounce score of 1 (of 5). Therefore I changed a setting or two just let the server I still exist. Maybe the fault lies within gmail.com? Still, the two I sent yesterday remain in cyberspace. I have been able to post follow-ups all along, but yesterday, when creating a new thread, I didn't see it, nor any replies. Thor ___ --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
Re: [Asterisk-Users] Context mix-up
I have two fwd accounts, and I want them to behave differently. It
took me a while to figure out why it wouldn't work, but finally I
realized that the last definition in sip.conf is the one that steals
the show.
Its a common issue with sip since it matches on ip address, etc. Check
the archives on 'how' sip finds a matching sip.conf entry. Change your
fwd accounts to iax and you will have more control.
In my case with fwd #61890, incoming calls include the fwd number, so
extensions.conf entries like this:
exten = 61890,1,NoOp,${CALLERID}
exten = 61890,2,Goto(bus-ivr-main|s|1)
exten = 61890,3,Hangup
work just fine. Your second number would simply have a different
exten = statement.
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Re: [Asterisk-Users] Context mix-up
I have now received the messages I sent today, this seems to have happened after I updated some settings at digium.com's list server. Why that would matter, I don't know. According to the list server, I had a bounce score of 1 (of 5). Therefore I changed a setting or two just let the server I still exist. Maybe the fault lies within gmail.com? Still, the two I sent yesterday remain in cyberspace. I have been able to post follow-ups all along, but yesterday, when creating a new thread, I didn't see it, nor any replies. I had the same problem which resulted from our broadband connection being down for a couple of days over Thanksgiving. Apparently an undeliverable email from the list server triggers a 'stop' function, and revisiting the list server page returns the sending of email again. Not a problem for me as long as one is aware of the functionality. (Kind of hard to miss it though with 200+ emails per day.) ___ --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
[Asterisk-Users] Context mix-up
I have two fwd accounts, and I want them to behave differently. It took me a while to figure out why it wouldn't work, but finally I realized that the last definition in sip.conf is the one that steals the show. Simplified, I have this: register = account1:[EMAIL PROTECTED]/88 register = account2:[EMAIL PROTECTED]/87 [fwdaccount1] context = context1 host=fwd.pulver.com . [fwdaccount2] context = context2 host=fwd.pulver.com . In extensions.conf: [context1] exten = 88,1,NoOp(Testing context1) [context2] exten = 87,1,NoOp(Testing context2) What happens in my case, is that every call goes into the context defined _last_ in sip.conf. So any call to account1 will be branded context2 and fail, because extension 88 is not defined in context2. Calls to account2 will work ok. If the two definitions in sip.conf trade places, the whole thing will work the other way around. [fwdaccount2] context = context2 host=fwd.pulver.com . [fwdaccount1] context = context1 host=fwd.pulver.com . Calls to either account will be branded context1 and fail if account 2 was called. If this is how it is supposed to work, the workaround must be to let both accounts enter the same context and differentiate their behavior based on the extension dialed. Not difficult, but I thought it would be possible to let them have different contexts from the start. Thor ___ --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
[Asterisk-Users] Context restrictions for long distance access, examples not clear?
Hi, I am trying to limit access to long distance in my dial plan but I am really confused by the examples I am seeing (perhaps I am misunderstanding how context work). The following example was given in a previous posting. [extensions] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup [local] exten = _XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _XX,2,Congestion [long-distance] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1NXXNXX,2,Congestion [local-users] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup include = local include = extensions [long-users] exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup include = local include = long-distance include = extensions What I do not understand is how this restricts access. Since the context 'extensions' is included in both would that not give all users access to local and long distance??? Or is there some sort of order of entry thing with context??? I supposed that zapata.conf would include a reference to extensions - that would be the only reason for having the extension context... Also since the extensions appear under local-users and long-users followed by the include 'extensions' wouldn't this generate an error since the extension already exist (ie in local users has the extension 8478414198 with a priority of 1 and the include statement means that another extension 8478414198 with a priority of 1 in the same context 'local-users') Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Context restrictions for long distance access, examples not clear?
What context are your phones in? (context= in sip or iax config) If your phones are in the local-users context, they will be able to dial numbers found in local-users, extensions and local. If your phones are in the long-users context, they will be able to dial numbers in long-users, local, long-distance and extensions. Extensions in a context are handled in the order they are listed. In this case, I would remove the entries which are also in extensions from the local-users and long-users extensions. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Friday, November 18, 2005 12:00 PM To: Asterisk - Users Subject: [Asterisk-Users] Context restrictions for long distance access,examples not clear? Hi, I am trying to limit access to long distance in my dial plan but I am really confused by the examples I am seeing (perhaps I am misunderstanding how context work). The following example was given in a previous posting. [extensions] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup [local] exten = _XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _XX,2,Congestion [long-distance] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1NXXNXX,2,Congestion [local-users] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup include = local include = extensions [long-users] exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup include = local include = long-distance include = extensions What I do not understand is how this restricts access. Since the context 'extensions' is included in both would that not give all users access to local and long distance??? Or is there some sort of order of entry thing with context??? I supposed that zapata.conf would include a reference to extensions - that would be the only reason for having the extension context... Also since the extensions appear under local-users and long-users followed by the include 'extensions' wouldn't this generate an error since the extension already exist (ie in local users has the extension 8478414198 with a priority of 1 and the include statement means that another extension 8478414198 with a priority of 1 in the same context 'local-users') Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Context restrictions for long distance access, examples not clear?
Jonathan k. Creasy wrote: What context are your phones in? (context= in sip or iax config) If your phones are in the local-users context, they will be able to dial numbers found in local-users, extensions and local. If your phones are in the long-users context, they will be able to dial numbers in long-users, local, long-distance and extensions. Extensions in a context are handled in the order they are listed. In this case, I would remove the entries which are also in extensions from the local-users and long-users extensions. This last statement isn't quite true. Included contexts are included in the order they are listed, but extensions in a context are *not* handled in the order they are listed. Asterisk sorts them and that sorted order is how extensions are handled, *followed* by included contexts in the order they are included (which have their extensions sorted by Asterisk first). Check out http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting at the bottom of the page. Cheers, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context configuration with AstTapi
Hi
I am using Asterisk
TAPI driver with Outlook and have many contacts with numbers listed as +44 1XXX
XX which is international dialling for UK. My Asterisk context is as
follows:
[outlook]
exten =
_0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
exten =
_00.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
How can I set up the
context to dial a number starting with +44 from Outlook. I have
tried:
exten =
_+.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
and
exten =
_+44.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:2})
but both do not dial
number. Can Asterisk be set to recognise "+" and change it to
"00"?
Thanks for your
help
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Re: [Asterisk-Users] context question
this can help u: SIP.CONF [1] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 1 pickupgroup = 1 callerid = "Bruno De Luca 1" 1 secret = 1234 mailbox = 1 context=1 [2] host = dynamic type = friend language = it qualify = no dtmfmode = rfc2833 callgroup = 2 pickupgroup = 2 callerid = "Bruno De Luca 2" 2 secret = 1234 mailbox = 2 context=2 [3] ... context=1 [4] ... context=2 EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
That doesnt really help. As stated in the email you replied to what is to prevent someone doing say [1] exten = 1,1,goto(2,1,1) or customer A *and* customer B trying to define the same context name, to use your example lets say they both want to create context '1'. I want to be able to create 1 system that has multiple users who are able to create their own dialplans without naming collisions with other customers or gotos going to other customers, etc. This is more for a virtual hosting type setup so I can have one large machine instead of many smaller ones, thus allowing for better ROI. While many have suggested that I learn the basics of contexts (as you did) no one has been able to ansewr the actual question asked making me think there is no current answer, and an AGI is the way to go. That way I can have more control over what data is observed and all that. I just didnt want to write an AGI if there was an existing solution, especially if it was part of asterisk itself and not an external program. On Mon, 2005-09-26 at 09:31 +0200, Bruno De Luca wrote: this can help u: EXTENSIONS.CONF [1] exten = 1,1,Dial(SIP/1) exten = 3,1,Dial(SIP/3) [2] exten = 2,1,Dial(SIP/2) exten = 4,1,Dial(SIP/4) trixter http://www.0xdecafbad.com wrote: They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Friday, September 23, 2005 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] context question They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context question
On Sat, 2005-09-24 at 09:10 -0400, Alex Vishnev wrote: I briefly looked thru the code and I don't believe there is a way to separate the context or really make them independent. I know exactly what you want to accomplish. I think it could be done with a little trick. For example, every customer on hosted pbx would be given some kind of unique identifier. The back-end would silently place the identifier at the beginning or the end of the context making the new name totally unique. The front-end would hide identifier from users view and just present the name of the context. That way, customers can name their context anything they like and there would be no collision. In that case, Goto would also be local to the context as the real context name will contain customer id. Does that work for you? no, because as I stated I didnt like that for personal reasons. That sounds exactly what I was thyinking too, prepending some customer specific identifier. If that is the only way to do this, then I think I will just have to run everything through an AGI, which can differentiate between customers since none of the 'dialplan' is in extensions.conf :) Thanks though, at least its confirmed that this doesnt exist (yet anyway). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context question
Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context question
They are aware of each other in 2 senses. First you can goto() them. I wanted to stop the ability of someone to put in a goto() in their dialplan to a context that is someone elses (think asterisk hosting). Second naming collissions. I wanted to stop two people from having the same name and causing grief that way. That is why I made the references about prepending some customer id or something, but I dont think that is the best way to accomplish this (personal preference), so it will either be an AGI to accomplish this or it will be something else that already exists that I havent been able to locate as yet. On Fri, 2005-09-23 at 21:50 -0500, [EMAIL PROTECTED] wrote: I may be missing something, but aren't all contexts unaware of each other be default? If I do the following [contexta] exten = 3200,1,Dial(SIP/3200,5) [contextb] exten = 3300,1,Dial(SIP/3300,5) Each context has a phone and they can't call each other. The are completely isolated. Unless I'm missing what you are trying to do trixter http://www.0xdecafbad.com wrote: Is there any way within asterisk to limit the scope of contexts, basically to make one context totally unaware of another. The application I had in mind involved allowing users to create their own dial plans. To that end I wanted to make it so that a given user could not call a different users dialplan. I could filter everything and prepend a customer id to every context they specify, but that can get ugly fast, especially when the parser misses something. If this doesnt exist I can surely do it with an agi, and that is the road I am headed down right now, but why duplicate an effect that may already exist? Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context overlap?
I have an auto-attendant for day and night. When the [businesshours] AA runs, it executes exten = s1 through exten = s9, then continues with exten = s10 in [nightmode], even though they are in different contexts. This was working fine until I added more 's' extensions in night mode. When I comment out all 's' extensions 10 and above in [nightmode], it works fine again. Is this a bug or am I missing something? Dylan. Asterisk CVS-v1-0-02/24/05-13:18:42 [auto-attendant] ; Business Hours include = businesshours|08:00-16:59|mon-fri|*|* ; After Hours (anything that doesn't match holiday or businesshours) include = nightmode [businesshours] exten = s,1,Answer exten = s,2,DigitTimeout,15 exten = s,3,ResponseTimeout,20 exten = s,4,Background(thank-you-for-calling) exten = s,5,Background(if-u-know-ext-dial) ... exten = s,9,Background(to-hear-menu-again) [nightmode] exten = s,1,Answer exten = s,2,DigitTimeout,15 exten = s,3,ResponseTimeout,20 exten = s,4,Background(thank-you-for-calling) exten = s,5,Background(if-u-know-ext-dial) ... exten = s,10,Background() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Context overlap?
, if hopefully im understanding your question. Even tough they are in different context.. i think thats wrong, since you are including both contexts in auto-attendant, they both are in the same context. The bussinesshours run first because its included first (in the hours set in the configuration), and if you dont have higher S extensions in the night context all works fine, but when you have more that 10 (more than bussinesshours) in night, then it continues with the next S extension. Hope i have been clear, and hope it helps you. Best Regards -Moisés Silva On Apr 6, 2005 10:36 PM, Dylan VanHerpen [EMAIL PROTECTED] wrote: I have an auto-attendant for day and night. When the [businesshours] AA runs, it executes exten = s1 through exten = s9, then continues with exten = s10 in [nightmode], even though they are in different contexts. This was working fine until I added more 's' extensions in night mode. When I comment out all 's' extensions 10 and above in [nightmode], it works fine again. Is this a bug or am I missing something? Dylan. Asterisk CVS-v1-0-02/24/05-13:18:42 [auto-attendant] ; Business Hours include = businesshours|08:00-16:59|mon-fri|*|* ; After Hours (anything that doesn't match holiday or businesshours) include = nightmode [businesshours] exten = s,1,Answer exten = s,2,DigitTimeout,15 exten = s,3,ResponseTimeout,20 exten = s,4,Background(thank-you-for-calling) exten = s,5,Background(if-u-know-ext-dial) ... exten = s,9,Background(to-hear-menu-again) [nightmode] exten = s,1,Answer exten = s,2,DigitTimeout,15 exten = s,3,ResponseTimeout,20 exten = s,4,Background(thank-you-for-calling) exten = s,5,Background(if-u-know-ext-dial) ... exten = s,10,Background() ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context
How do I how to send a call as [EMAIL PROTECTED] ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context
On Mar 26, 2005, at 9:38 PM, AS wrote: How do I how to send a call as [EMAIL PROTECTED] ? Not exactly sure what you are asking. If you are trying to dial a specific extension within a specific context then I use a GoTo. [currentcontext] exten = 8885551212,1,GoTo(anothercontext,100,1) [anothercontext] exten = 100,1,Dial(whatever) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context
Hi, We are trying to redirect some DDIs from one machine to another machine over IAX2. When we redirect the number as: exten = 0730184220,1,Dial(IAX2/username) it sends the call fine to the other end. However, at the other end the calls comes in as s@''. In the past when we have bought DDIs off other providers, we have received the call as [EMAIL PROTECTED] Any suggestions would be appreciated? Cheers, Sahil - -Original Message- - From: [EMAIL PROTECTED] - [mailto:[EMAIL PROTECTED] On Behalf Of Jerry - Sent: Sunday, 27 March 2005 1:55 PM - To: Asterisk Users Mailing List - Non-Commercial Discussion - Subject: Re: [Asterisk-Users] context - - - On Mar 26, 2005, at 9:38 PM, AS wrote: - - How do I how to send a call as [EMAIL PROTECTED] ? - - Not exactly sure what you are asking. If you are trying to - dial a specific extension within a specific context then I - use a GoTo. - - [currentcontext] - exten = 8885551212,1,GoTo(anothercontext,100,1) - - [anothercontext] - exten = 100,1,Dial(whatever) - - ___ - Asterisk-Users mailing list - Asterisk-Users@lists.digium.com - http://lists.digium.com/mailman/listinfo/asterisk-users - To UNSUBSCRIBE or update options visit: - http://lists.digium.com/mailman/listinfo/asterisk-users - - ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] context
You need to specify what number on the remote server you are trying to
reach like this:
Exten = 0730184220,1,Dial(IAX2/username:passwd/${EXTEN})
Which will dial the same number (0730184220) on the remote Asterisk
server. Or, you could even add the context to the dial statement like
this:
Exten = 0730184220,1,Dial(IAX2/username:passwd/[EMAIL PROTECTED])
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AS
Sent: Saturday, March 26, 2005 10:13 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] context
Hi,
We are trying to redirect some DDIs from one machine to another machine
over IAX2.
When we redirect the number as:
exten = 0730184220,1,Dial(IAX2/username)
it sends the call fine to the other end. However, at the other end the
calls comes in as s@''. In the past when we have bought DDIs off other
providers, we have received the call as [EMAIL PROTECTED]
Any suggestions would be appreciated?
Cheers,
Sahil
- -Original Message-
- From: [EMAIL PROTECTED]
- [mailto:[EMAIL PROTECTED] On Behalf Of Jerry
- Sent: Sunday, 27 March 2005 1:55 PM
- To: Asterisk Users Mailing List - Non-Commercial Discussion
- Subject: Re: [Asterisk-Users] context
-
-
- On Mar 26, 2005, at 9:38 PM, AS wrote:
-
- How do I how to send a call as [EMAIL PROTECTED] ?
-
- Not exactly sure what you are asking. If you are trying to
- dial a specific extension within a specific context then I
- use a GoTo.
-
- [currentcontext]
- exten = 8885551212,1,GoTo(anothercontext,100,1)
-
- [anothercontext]
- exten = 100,1,Dial(whatever)
-
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[Asterisk-Users] context of transfer
How set the context of Transfer function? There are 2 context in extensions.conf. [con1] exten = _0.,1,Dial(SIP/[EMAIL PROTECTED]) [con2] exten = 812,1,Transfer(001345566);How can use the dialplan of context con1? Thanks! Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context of transfer
How set the context of Transfer function? There are 2 context in extensions.conf. [con1] exten = _0.,1,Dial(SIP/[EMAIL PROTECTED]) [con2] exten = 812,1,Transfer(001345566);How can use the dialplan of context con1? Thanks! Bill Chen ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Context fails so falling back to extension s ?
I realize it's bad form, but I'd really appreciate some hand holding here. AMP is making me pull my hair out and the mountain of configuration data in extensions.conf is starting to get to me... I have the first channel configured in zapata.conf to take incoming contexts to from-pstn. I'm using AMP neat Incoming Calls configuration page and extensions-additional.conf's include statement is uncommented. The damn thing still doesn't work... I get this error when calling in: Feb 10 15:55:17 VERBOSE[3477]: -- Starting simple switch on 'Zap/2-1' Feb 10 15:55:22 ERROR[3477]: fsk_serie made mylen 0 (-46) Feb 10 15:55:22 WARNING[3477]: CallerID feed failed: Success Feb 10 15:55:22 WARNING[3477]: CallerID returned with error on channel 'Zap/2-1' Feb 10 15:55:22 VERBOSE[3477]: == Starting Zap/2-1 at from-pstn,s,1 failed so falling back to exten 's' Feb 10 15:55:22 VERBOSE[3477]: == Starting Zap/2-1 at from-pstn,s,1 still failed so falling back to context 'default' Feb 10 15:55:22 WARNING[3477]: Channel 'Zap/2-1' sent into invalid extension 's' in context 'default', but no invalid handler Extension 's'? I thought 's' meant Start, not an actual extension. If there's something I'm not reading or need to read again, don't hesitate to hit me with a clue stick. Regards, aaron.glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Context fails so falling back to extension s ?
Extension 's'? I thought 's' meant Start, not an actual extension. If there's something I'm not reading or need to read again, don't hesitate to hit me with a clue stick. Sort of. 's' is used when there is no matching extension in the context. It's the fallback extension if there's no match. http://www.voip-info.org/wiki-Asterisk+s+extension You don't list your extensions.conf, but taking a stab at it, you would put in something like: [from-pstn] exten = s,1,Dial(YourInternalExtension,15) 'Dial whatever your internal extension is for 15 seconds exten = s,2,Hangup() 'Hang up the line if nobody answers. You could put in a goto to fire the call to the [from-internal] context in extensions_additional.conf so it can have voicemail logic. I found that the best part of AMP is they have a really really good extensions.conf you can use as a template to make a customized dialplan. Starting from the base AMP extensions.conf and extensions_additional.conf, I have modified my dialplan *way* beyond what AMP can do, but it's AMP's template that got me started. I shudder to think of the hours I would have wasted creating all of the dialplan logic over again from scratch without AMP giving me a leg-up. Now, I don't even use AMP anymore except for FOP and the call detail logs. YMMV. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] context wide variable scope
Maybe i missed this somewhere, but is it possible to define a variable with a scope of the current context? I know i can define a system wide variable, and i can define one that is valid for the duration of the channel, but is it possible to define a variable that comes into scope for every channel that comes into a context? I don't think so, but i wanted to make sure. - jeremy -- Jeremy Hinton A little nonsense Senior Network Manager now and then Continental VisiNet Broadband is relished by [EMAIL PROTECTED]the wisest men 757 873 4500 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] context wide variable scope
On Fri, 2005-01-14 at 14:13 -0500, Jeremy Hinton wrote: Maybe i missed this somewhere, but is it possible to define a variable with a scope of the current context? I know i can define a system wide variable, and i can define one that is valid for the duration of the channel, but is it possible to define a variable that comes into scope for every channel that comes into a context? I don't think so, but i wanted to make sure. You can if you use the DB functions, kind of. If you could better describe an example of your problem, there may be better/other solutions for you to use. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ${CONTEXT} variable
Hi all,
Is there an equivalent of the ${CONTEXT} variable that represents the
*original* context of the call? i.e. If a call originates in the
'internal' context, no matter where it goes, this alternate version of
${CONTEXT} would never change from saying 'internal'?
I realize I could set this using the dialplan but I just wonder if there
this already exists, and if not, would there be any objection to adding
it? It could be ${CALL_CONTEXT} or ${ORIGINAL_CONTEXT}, or similar.
Thanks,
Chris
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Re: [Asterisk-Users] ${CONTEXT} variable
Christopher L. Wade wrote:
Hi all,
Is there an equivalent of the ${CONTEXT} variable that represents the
*original* context of the call? i.e. If a call originates in the
'internal' context, no matter where it goes, this alternate version of
${CONTEXT} would never change from saying 'internal'?
I realize I could set this using the dialplan but I just wonder if there
this already exists, and if not, would there be any objection to adding
it? It could be ${CALL_CONTEXT} or ${ORIGINAL_CONTEXT}, or similar.
Thinking about this, the name of the variable might be ${DEVICE_CONTEXT}
instead. This seems more in keeping with what I was intending the
variable to represent, which is the 'context=' line from the appropriate
config file.
Thanks,
Chris
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Re: [Asterisk-Users] ${CONTEXT}
On Sun, 2004-08-29 at 21:44, Steve Maroney wrote:
I have some problems with my extensions.conf. When a call from pstn comes
in, the call gets put into the [from-fxo] context. From there the caller
is able to dial sip extensions that are included from the [sip-extenions]
context.
When a sip extension is dialed and connected, and then at some point
transfered, the ${CONTEXT} variable is changed from [from-fxo] to
[from-sip]. This leaves the caller from the pstn open to all extenions
that normally only my sip (trusted) clients would be able to dial, such as
outgoing calls on my other FXO ports.
Is the changing on the ${CONTEXT} variable by design (and needs to
secrured in my dialplan) or a bug ?
Post a snippit of your dialplan. Without this, you leave us guessing as
to whether you did the right thing or not.
--
Steven Critchfield [EMAIL PROTECTED]
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