Hi All,
pl help me with this basic question.
I have a users (soft clients) with usernames having Alphabetics.
I want to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this done
using extensions.conf)
my expected result is :
alice at pbx.com should be able to call bob at pbx.com
where pbx.com is astersik.
Can you pl let me know how I can achieve this?
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100427/1b17f3a1/attachment.htm
I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten => 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten => alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote:> Hi All, > > pl help me with this basic question. > > I have a users (soft clients) with usernames having Alphabetics. > I want to use Asterisk as my server. > > How should I have the dial plans as there are no numbers involved . > so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) > > my expected result is : > alice at pbx.com should be able to call bob at pbx.com > where pbx.com is astersik. > > Can you pl let me know how I can achieve this? > > > -- > _____________________________________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100427/c59e65d9/attachment.htm
Thanks a lot jim for the reply. My issue is : there is no numbers involved. I have soft clients. when a user (bob) calls Alex, he just opens his sip client and types in alex at pbx.com so how can I write the translations for a case like that? the examples you gave are when there are numbers..can u pl give me complete numbering plam ________________________________ From: Jim Dickenson <dickenson at cfmc.com> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Sent: Tue, April 27, 2010 7:09:45 PM Subject: Re: [asterisk-users] Dial plan question. I am not sure what your problem is. You can have a numeric extension dial an alphabetic sip user. exten => 123,1,Dial(SIP/somename) The soft phone registers to your box with whatever username you set up. If your phone can dial alpha then you can have exten => alpha,1,Dial(SIP/$(EXTEN}) -- Jim Dickenson mailto:dickenson at cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote: Hi All,> > >pl help me with this basic question. > > >I have a users (soft clients) with usernames having Alphabetics. >I want to use Asterisk as my server. > > >How should I have the dial plans as there are no numbers involved . >so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) > > >my expected result is : >alice at pbx.com should be able to call bob at pbx.com >where pbx.com is astersik. > > >Can you pl let me know how I can achieve this? > > >-- >_____________________________________________________________________ >-- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100427/3938a168/attachment.htm
here is the dail plan I am using:
my extensions file:
[globals]
[ext-sip]
host=provider.sip.com
[default]
exten => bob,1,Dial(SIP/${EXTERN}@ext-sip,20)
----
expected dialing plan:
when some one calls bob,
Asterisk should add bob at provider.sip.com and sent to the external world.
But that is not working,.
can you pl let me know what I am missing?
Also,
is there a way that Asterisk will read completely bob at provider.sip.com from
the received sip message and forwards directly to that domain.
That means,
When we receive a Request to bob at provider.sip.com, Asterisk should send that
to the outgoing interface to bob at provider.sip.com\.
some plan like..
extern=>bob at x.com,1,Dial(SIP/{EXTERN},20)...????
________________________________
From: Aditya Kumar <adityakumar345 at yahoo.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Sent: Tue, April 27, 2010 10:11:16 PM
Subject: Re: [asterisk-users] Dial plan question.
Thanks a lot jim for the reply.
My issue is :
there is no numbers involved. I have soft clients.
when a user (bob) calls Alex,
he just opens his sip client and types in alex at pbx.com
so how can I write the translations for a case like that?
the examples you gave are when there are numbers..can u pl give me complete
numbering plam
________________________________
From: Jim Dickenson <dickenson at cfmc.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users
at lists.digium.com>
Sent: Tue, April 27, 2010 7:09:45 PM
Subject: Re: [asterisk-users] Dial plan question.
I am not sure what your problem is. You can have a numeric extension dial an
alphabetic sip user.
exten => 123,1,Dial(SIP/somename)
The soft phone registers to your box with whatever username you set up.
If your phone can dial alpha then you can have
exten => alpha,1,Dial(SIP/$(EXTEN})
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
http://www.cfmc.com/
On Apr 27, 2010, at 6:48 PM, Aditya Kumar wrote:
Hi All,>
>
>pl help me with this basic question.
>
>
>I have a users (soft clients) with usernames having Alphabetics.
>I want to use Asterisk as my server.
>
>
>How should I have the dial plans as there are no numbers involved .
>so How can I make the configuration to work ( with numbers I can get this
done using extensions.conf)
>
>
>my expected result is :
>alice at pbx.com should be able to call bob at pbx.com
>where pbx.com is astersik.
>
>
>Can you pl let me know how I can achieve this?
>
>
>--
>_____________________________________________________________________
>-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100427/9804fcd5/attachment-0001.htm
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:> exten => bob,1,Dial(SIP/${EXTERN}@ext-sip,20)Where did you define EXTERN? S
Thanks a lot Jim and Ryan.
It worked with changing the order as you suggested.
--
Few more questions on Dial plan:
use case:
when some one in my pbx calls 100.200, I have translations well defined, Media
also (media via asterisk) --Works.
when some one calls bob, or for any names I am adding Domain and call is been
sent to the other party -- Works, no media...
For the cases when it is talking to the external work,
I want Astersik not to do anything with the SDP/payload.
I want it to send as it is to the external proxy.
How can I achieve this? so that the SDP/payload will not be modified for users
talking to the external world.
I want media for those external devices to come Directly to the users in my
pbx.
----
Do
you mean you want
exten => bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user
ext-sip to that system's extension bob?
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
http://www.cfmc.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100428/e7b49fe8/attachment.htm
On Tue, Apr 27, 2010 at 8:48 PM, Aditya Kumar <adityakumar345 at yahoo.com>wrote:> Hi All, > > pl help me with this basic question. > > I have a users (soft clients) with usernames having Alphabetics. > I want to use Asterisk as my server. > > How should I have the dial plans as there are no numbers involved . > so How can I make the configuration to work ( with numbers I can get this > done using extensions.conf) > > my expected result is : > alice at pbx.com should be able to call bob at pbx.com > where pbx.com is astersik. > > Can you pl let me know how I can achieve this? > >You would need to setup each user in sip.conf like so: [alice] type=friend context=alpha-names fromuser=alice secret=password domain=pbx.com [bob] type=friend context=alpha-names fromuser=bob secret=password domain=pbx.com etc etc.. Then in your extensions.conf, you would setup: [alpha-names] ; Dial by name exten => alice,1,Verbose(Calling alice) exten => alice,n,Dial(SIP/alice,20) exten => alice,n,Hangup() exten => bob,1,Verbose(Calling bob) exten => bob,n,Dial(SIP/bob,20) exten => bob,n,Hangup() etc etc. You could also use pattern matching in your extensions.conf like this: [alpha-names] ;Dial by name, pattern matching exten => _.,1,Verbose(Calling ${EXTEN}) exten => _.,n,Dial(SIP/${EXTEN},20) exten => _.,n,Hangup() except that's going to catch everything, including the built-in 'h', 'i', and 't' extensions (you can look these up on voip-info.org for more info on those). Configure each of your softphone clients with the usernames you defined in your sip.conf (i.e the softphone on Alice's computer would have a username of alice, password of password, and domain of pbx.com, using the asterisk server as your registrar / proxy server address, same with Bob's softphone). Your softphone has to allow alpha dialing from contacts though. You haven't mentioned which softphone you're using, if you do that we may be able to give you specifics for that softphone as well. -- Thanks, --Warren Selby http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100428/cb21d742/attachment.htm