[Asterisk-Dev] NO DTMF in the Outgoing call

[Areski]

I m not trying to make asterisk sending DTMF.
It's the DTMF sent by the sip client that aren't recognized by Asterisk.
:/

Thanks anyway,
Areski


On Wed, 2004-02-11 at 17:16, Chris Heiser wrote:
> Areski,
> 
> I have had similar problems especially when attempting to send DTMF to a
> Cisco ATA186 attached to a PBX.  Attached is a patch for rtp.c to increase
> the duration of RFC2833 DTMF tones to a more reasonable value.  Try it out
> and see what happens!
> 
> --Chris
> 
> > -----Original Message-----
> > From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
> > admin at lists.digium.com] On Behalf Of Areski
> > Sent: Wednesday, February 11, 2004 10:50 AM
> > To: asterisk-dev
> > Subject: [Asterisk-Dev] NO DTMF in the Outgoing call
> > 
> > Hello all,
> > 
> > I cannot have DTMF working when I m making an outgoing call from Asterisk.
> > I tried on different servers with different  Asterisk versions.
> > I tested all configurations possible with X-lites and also with a
> > CiscoAS5300,
> > no way... always the same problem.
> > 
> > I was trying to do some modifications in chan_sip but I m pretty novice
> > with Asterisk sources and I didn't find out anything.
> > After scanning the sip log, I m thinking it's perhaps an RTP problem.
> > Below, I attached the sip log...
> > 
> > I will greatly appreciate if someone can give me some advices.
> > Kind regards,
> > Areski
> > 
> > 
> > 
> > ------------- SIP DEBUG OUTGOING-CALL : ASTERISK -> X-LITE ----------
> > 
> > 
> > We're at 192.168.1.252 port 16642
> > Answering with preferred capability 8
> > Answering with non-codec capability 1
> > 12 headers, 9 lines
> > Reliably Transmitting:
> > INVITE sip:phone1 at 192.168.1.29 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> > From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> > To: <sip:phone1 at 192.168.1.29>
> > Contact: <sip:asterisk at 192.168.1.252>
> > Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> > CSeq: 102 INVITE
> > User-Agent: Asterisk PBX
> > Date: Wed, 11 Feb 2004 15:42:36 GMT
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > Content-Type: application/sdp
> > Content-Length: 191
> > 
> > v=0
> > o=root 11215 11215 IN IP4 192.168.1.252
> > s=session
> > c=IN IP4 192.168.1.252
> > t=0 0
> > m=audio 16642 RTP/AVP 8 101
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-16
> >  (no NAT) to 192.168.1.29:5060
> > 
> > 
> > Sip read:
> > SIP/2.0 100 Trying
> > Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> > From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> > To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> > Contact: <sip:phone1 at 192.168.1.29:5060>
> > Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> > CSeq: 102 INVITE
> > User-Agent: X-Lite build 1079
> > Content-Length: 0
> > 
> > 
> > Sip read:
> > SIP/2.0 180 Ringing
> > Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> > From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> > To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> > Contact: <sip:phone1 at 192.168.1.29:5060>
> > Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> > CSeq: 102 INVITE
> > User-Agent: X-Lite build 1079
> > Content-Length: 0
> > 
> > 
> > Sip read:
> > SIP/2.0 200 Ok
> > Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> > From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> > To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> > Contact: <sip:phone1 at 192.168.1.29:5060>
> > Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> > CSeq: 102 INVITE
> > Content-Type: application/sdp
> > User-Agent: X-Lite build 1079
> > Content-Length: 197
> > 
> > v=0
> > o=phone1 161667652 161667652 IN IP4 192.168.1.29
> > s=X-Lite
> > c=IN IP4 192.168.1.29
> > t=0 0
> > m=audio 8000 RTP/AVP 8 101
> > a=rtpmap:8 pcma/8000
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-15
> > 
> > 10 headers, 9 lines
> > Found audio format ALAW
> > Found audio format UNKN
> > Found description format pcma
> > Found description format telephone-event
> > Capabilities: us - 8, them - 8/0, combined - 8
> > Non-codec capabilities: us - 1, them - 1, combined - 1
> > list_route: hop: <sip:phone1 at 192.168.1.29:5060>
> > set_destination: Parsing <sip:phone1 at 192.168.1.29:5060> for address/port
> > to send to
> > set_destination: set destination to 192.168.1.29, port 5060
> > Transmitting:
> > ACK sip:phone1 at 192.168.1.29:5060 SIP/2.0
> > Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> > From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> > To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> > Contact: <sip:asterisk at 192.168.1.252>
> > Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> > CSeq: 102 ACK
> > User-Agent: Asterisk PBX
> > Content-Length: 0
> > 
> >  (no NAT) to 192.168.1.29:5060
> > 
> > 
> > 
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