[Asterisk-Users] sip.conf not paying attention to allow/disallow

Race Vanderdecken asterisk at vanderdecken.com
Mon Nov 22 15:40:32 MST 2004


This is correct.

Out going calls default to the global capability.

Then some CODECS are added or subtracted.

But,

	The CODECS are not in a list the way you would think. They are
in a bit field.

	So the obvious list of 

		1. alaw
		2. ulaw	
		3. gsm

	where you would think that Asterisk would first try alaw, then
ulaw, then gsm. 
	This NOT what happens.

(Aside : The capabilities are blended into a bit field, "010110110". The
first bit that works is the winner.)

	But when you call out the default "globals" are dropped in to
the capablitiles bitfield with the "privates" only being used if the
jointCapabilites has not been previously set. Asterisk does not seem to
check the private CODECS going out the door. But that is only a short
quick look at the code.

The SIP channel is good, but not great. It is not a bug so much as an
unexpanded feature.

My advise, create a database.



Race "The Tyrant" Vanderdecken


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brian
Wilkins
Sent: 22 November 2004 12:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] sip.conf not paying attention to
allow/disallow

It's not a bug. You are setting global parameters. When you do that, it 
overrides the peer settings. Try setting for each individual peer. If
you 
have many peers (100 or more), use a database solution such as RealTime
or a 
script to build your sip.conf (one is included with the cvs).

On Monday 22 November 2004 09:55 pm, Brian C. Fertig wrote:
> I have found this same problem to be true.   I don't know what to do
to
> fix it.  I believe it's a bug but don't know for sure.  If you find a
> way drop me a line I would like to know.
>
> Brian
>
>
>
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew
> Boehm
> Sent: Monday, November 22, 2004 4:46 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] sip.conf not paying attention to
> allow/disallow
>
> In my sip.conf, under general I have:
>
> disallow=all
> allow=g729
> allow=alaw
> allow=ulaw
>
> Then I have a specific sip:
>
> [RNK]
> <clip>
> disallow=all
> allow=alaw
> allow=ulaw
> allow=gsm
>
> If I do this:
>
> exten => _9.,1,Dial(${EXTEN}@RNK,60)
>
> The call still goes out as G729 even though I've told the RNK to
> disallow
> g729. I need to be able to make other 729 calls but to this one
> paticular
> group, they need to be 711.
>
> Any ideas?
>
> Thanks,
> Matthew
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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 at 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

-- 
Brian Wilkins
Software Engineer
brian at hcc.net

Heritage Communications Corporation
  Melbourne, FL     USA     32935
321.308.4000 x33
http://www.hcc.net

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at 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





More information about the asterisk-users mailing list