asterisk users - Dec 2004 - One SIP peer use 2 diff codecs?

I asked this question once before with no answer. Hopefully someone can
help me as I cannot see a way to do this. I am wanting to differentiate
inbound calls voice from FAX. The purpose of course voice gets g729 and
FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
SIP peer (like it should) but it's always goes to the prefered codec.
Anyone have suggestions on how to force 711 for fax detection?


	Thanks.

	Matt

Matt Schulte wrote:
>I asked this question once before with no answer. Hopefully someone can
>help me as I cannot see a way to do this. I am wanting to differentiate
>inbound calls voice from FAX. The purpose of course voice gets g729 and
>FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
>SIP peer (like it should) but it's always goes to the prefered codec.
>Anyone have suggestions on how to force 711 for fax detection?
>  
>
You can't do this.

--Eric

So what's the work around? Have faxes come from a diff IP?

-----Original Message-----
From: Eric Wieling aka ManxPower [mailto:eric@fnords.org] 
Sent: Monday, December 20, 2004 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] One SIP peer use 2 diff codecs?


Matt Schulte wrote:
>I asked this question once before with no answer. Hopefully someone can
>help me as I cannot see a way to do this. I am wanting to differentiate
>inbound calls voice from FAX. The purpose of course voice gets g729 and
>FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
>SIP peer (like it should) but it's always goes to the prefered codec. 
>Anyone have suggestions on how to force 711 for fax detection?
>  
>
You can't do this.

--Eric
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Now that I finally have someones attention :) I can explain the rest.
The problem is I'm sending calls out through SER (with digest of
course). So in my sip.conf I have:

[sipfarm]
insecure=very ; Because SER inbound doesn't know how to auth
host=blah.blah.net
type=peer
context=incoming
username=+1314XXXXXXX
secret=XXXXXXXXXXX
canreinvite=no
disallow=all
allow=g729
allow=ulaw


The problem is I have to Dial(SIP) *and* pass digest auth to SER, which
means I have to specify host=blah.blah.net in sip.conf for sipfarm.
Unless there's a way to pass Digest via Dial(SIP) cmd... two peers
inbound won't work because host= will always match.. 

Ideas? Thanks for your input :-)

	Matt

-----Original Message-----
From: Eric Wieling aka ManxPower [mailto:eric@fnords.org] 
Sent: Monday, December 20, 2004 11:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] One SIP peer use 2 diff codecs?


Matt Schulte wrote:
>So what's the work around? Have faxes come from a diff IP?
>  
>
Well have them come into a different user/friend at least.  The IP can 
be the same if you are authenticating on username/secret rather than IP.
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Give the FAX SIP device a different account and force it to Ulaw. For
example if the user was account 1111 you could create 1111F for fax
and 1111V for voice and have sperate allow/deny codecs

> I asked this question once before with no answer. Hopefully someone can
> help me as I cannot see a way to do this. I am wanting to differentiate
> inbound calls voice from FAX. The purpose of course voice gets g729 and
> FAX gets 711 (ulaw). The problem I'm having is everytime it matches the
> SIP peer (like it should) but it's always goes to the prefered codec.
> Anyone have suggestions on how to force 711 for fax detection?
exten fax,1,SetVar(SIP_CODEC=alaw)
exten fax,2, ...

AFAIK, this is implemented for outgoing calls only, although I don't  
think it'll be too hard to do it on incoming also.

roy

>> How about this variable? :-)
>> ${SIP_CODEC}: Used to set the SIP codec for a call
>
> That only works for calls going OUT from Asterisk.  It does nothing 
> for incoming calls.  By the time the dialplan is called the codec is 
> already set.
perhaps tha should be changed, then, to allow more control over codec 
handling?

roy

Here, here,

I agree. Each channel should be able to decide which codec to try first.
This will cause more call setup messages.

1. Hi, do you have GSM?
	No

2. Hi again, do you have G.723
	No, go Fish.

3. Hmmm, do you have (cringe) G.729?
	Yes, do you have License for your Minkey?


My understanding of SIP is that the caller suggests, the callee replies
with their preference(s).

So the call setup is a little more complicated but not that hard to do,
just need some routing logic on the caller side. Sort of the way a
gatekeeper decides where to route a call.

Race	


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Roy Sigurd
Karlsbakk
Sent: 22 December 2004 06:25
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] One SIP peer use 2 diff codecs?
>> How about this variable? :-)
>> ${SIP_CODEC}: Used to set the SIP codec for a call
>
> That only works for calls going OUT from Asterisk.  It does nothing 
> for incoming calls.  By the time the dialplan is called the codec is 
> already set.
perhaps tha should be changed, then, to allow more control over codec 
handling?

roy

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