Re: [asterisk-users] enabling dialing by sip uri
On Thu, May 10, 2012 at 5:52 PM, Kevin P. Fleming wrote: > > You'll have to provide more details (primarily a CLI log) then in order for > anyone to be able to help you. You said that Asterisk "shows extension is > rejected", but extensions don't get rejected. Extensions can be 'not found', > but that's very different from rejected. > > Ok i will post more detailed log. -- -aft -- _ -- 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] enabling dialing by sip uri
On 05/10/2012 03:36 PM, Arif Hossain wrote: Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone, > and that URI does not resolve to the Asterisk server as its target, then the > INVITE request sent by the phone should not even be sent to Asterisk at all > (it should go to wherever the URI resolves to). > I'm using the asterisk's ip to form sip uri at the sip client. So it resolves to asterisk no doubt. You'll have to provide more details (primarily a CLI log) then in order for anyone to be able to help you. You said that Asterisk "shows extension is rejected", but extensions don't get rejected. Extensions can be 'not found', but that's very different from rejected. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- 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] enabling dialing by sip uri
On Thu, May 10, 2012 at 11:50 AM, Kevin P. Fleming wrote:
> On 05/10/2012 09:39 AM, Arif Hossain wrote:
>>
>> I have following sip account :
>>
>> Name/username Host Dyn
>> Forcerport ACL Port Status Description
>> demo-alice/demo-alice 192.168.7.47 D
>> N 1080 Unmonitored
>> demo-bob/demo-bob 192.168.7.47 D
>> N 5060 Unmonitored
>>
>> and i have set up the following extensions for them:
>>
>> ASTERISK_IP=192.168.7.39
>>
>> [users]
>> exten=>6001,1,Dial(SIP/demo-alice,20)
>> exten=>6002,1,Dial(SIP/demo-bob,20)
>>
>> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
>> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
>> exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
>> exten => _.,n,HangUp()u
>>
>> [macro-uri-dial]
>> exten=>s,n,NoOp(Calling as SIP address: ${ARG1})
>> exten=>s,n,Dial(SIP/${ARG1},60)
>>
>>
>> But if i dial sip uri the call does not happen. asterisk cli shows
>> extension is rejected.
>
>
> Asterisk is not a SIP proxy. If you are entering a SIP URI into your phone,
> and that URI does not resolve to the Asterisk server as its target, then the
> INVITE request sent by the phone should not even be sent to Asterisk at all
> (it should go to wherever the URI resolves to).
>
I'm using the asterisk's ip to form sip uri at the sip client. So it
resolves to asterisk no doubt.
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
> --
> _
> -- 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
--
-aft
--
_
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Re: [asterisk-users] enabling dialing by sip uri
On 05/10/2012 09:39 AM, Arif Hossain wrote:
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=>6001,1,Dial(SIP/demo-alice,20)
exten=>6002,1,Dial(SIP/demo-bob,20)
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten => _.,n,HangUp()u
[macro-uri-dial]
exten=>s,n,NoOp(Calling as SIP address: ${ARG1})
exten=>s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
Asterisk is not a SIP proxy. If you are entering a SIP URI into your
phone, and that URI does not resolve to the Asterisk server as its
target, then the INVITE request sent by the phone should not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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[asterisk-users] enabling dialing by sip uri
I have following sip account :
Name/username HostDyn
Forcerport ACL Port Status Description
demo-alice/demo-alice 192.168.7.47 D
N 1080 Unmonitored
demo-bob/demo-bob 192.168.7.47 D
N 5060 Unmonitored
and i have set up the following extensions for them:
ASTERISK_IP=192.168.7.39
[users]
exten=>6001,1,Dial(SIP/demo-alice,20)
exten=>6002,1,Dial(SIP/demo-bob,20)
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
exten => _.,n,HangUp()u
[macro-uri-dial]
exten=>s,n,NoOp(Calling as SIP address: ${ARG1})
exten=>s,n,Dial(SIP/${ARG1},60)
But if i dial sip uri the call does not happen. asterisk cli shows
extension is rejected.
--
-aft
--
_
-- 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
