[asterisk-users] setting dtmf mode for a particular peer

[Brent Davidson]

Have you tried the using the "SIPDtmfMode" function in your dial plan?  
It can be used to change the DTMF mode between two points in a call.  
The problem, I would think, would be if your phones are set up to ONLY 
send inband audio then you have to find someway to get audio to 
transcode the DTMF from inband to info.  I'm not familiar enough with 
the specifics of Asterisk's behavior to know whether that "just works" 
or if it needs some special setup.  Try putting SipDtmfMode(info) just 
before the dial command and see what happens.

Good Luck,
Brent


Brian J. Murrell wrote:
> On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
>   
>> No, that's correct.  The problem is that you aren't using the peer definition
>> when you dial (as you said, you've never needed it before).
>>
>> Use
>> Dial(SIP/1234 at voipmich)
>> NOT
>> Dial(SIP/1234 at tf.voipmich.com)
>>     
>
> OK.  Trying exactly as you describe above, it does dial:
>
>     -- Executing [s at macro-ringingdial:2] Dial("SIP/1011002206-b631f650", "SIP/18668398145 at voipmich") in new stack
>
> With "sip set debug peer voipmich" I'd expect to see SIP packets for
> every digit I press on my phone, right?  I don't.  I don't see anything
> beyond the initial call establishment:
>
> Audio is at 67.193.45.68 port 11724
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x2 (gsm) to SDP
> Reliably Transmitting (NAT) to 69.41.0.50:5060:
> INVITE sip:18668398145 at tf.voipmich.com SIP/2.0
> Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
> From: "2003" <sip:nobody at nodomain>;tag=as5c70ce0e
> To: <sip:18668398145 at tf.voipmich.com>
> Contact: <sip:nobody at 67.193.45.68>
> Call-ID: 1090b17e076edb94740dfd9c4f436590 at nodomain
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX  
> Max-Forwards: 70
> Date: Wed, 09 Apr 2008 21:55:49 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Type: application/sdp
> Content-Length: 207
>
> v=0
> o=root 11375 11375 IN IP4 67.193.45.68
> s=session
> c=IN IP4 67.193.45.68
> t=0 0
> m=audio 11724 RTP/AVP 0 3 
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=silenceSupp:off - - - - 
> a=ptime:20
> a=sendrecv
>
> ---
>     -- Called 18668398145 at voipmich
>     -- SIP/voipmich-084a5500 is making progress passing it to SIP/1011002206-b631f650
>   == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 'SIP/1011002206-b631f650' in macro 'ringingdial'
>   == Spawn extension (macro-ringingdial, s, 2) exited non-zero on 'SIP/1011002206-b631f650'
>
> Of course, in there between the call being established and torn down, I
> did hit lots of digits on my phone.
>
> b.
>
>   
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