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tilghman at mail.jeffa...
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 Posted: Wed Apr 09, 2008 4:18 pm Post subject: [asterisk-users] setting dtmf mode for a particular peer |
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On Wednesday 09 April 2008 14:12:08 Brian J. Murrell wrote:
| Quote: | When I make a toll-free call using tf.voipmich.com DTMF doesn't work.
According to this post:
http://www.trixbox.org/forums/trixbox-forums/help/enum-strangeness it's
because voipmich needs dtmfmode set to "info".
How do I specify this for a single SIP peer (tf.voipmich.com) given that
I normally don't register to them.
I have tried creating a sip.conf entry:
[voipmich]
type=peer
fromuser=nobody
fromdomain=nodomain
host=tf.voipmich.com
dtmfmode=info
But that does not appear to be working. Maybe my approach is all wrong.
Any ideas?
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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)
--
Tilghman |
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brent at texascountryt...
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| Posted: Wed Apr 09, 2008 5:21 pm Post subject: [asterisk-users] setting dtmf mode for a particular peer |
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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:
| Quote: | On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
| Quote: | 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)
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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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brent at texascountryt...
Guest
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| Posted: Thu Apr 10, 2008 11:13 am Post subject: [asterisk-users] setting dtmf mode for a particular peer |
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Brian J. Murrell wrote:
| Quote: | On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Does anyone know if Asterisk will convert an inband DTMF from one sip
channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
channel?
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You might also try "canreinvite=no" for both your phone and the sip
peer. I think it's normal procedure for Asterisk to drop out of the
call path once the call is established between two peers. The
canreinvite directive forces asterisk to remain as an intermediary, and
it will probably do the transcoding that way. If I'm not mistaken this
is also useful for making calls between two system that have no common
codecs.
-Brent |
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