[Asterisk-Dev] NO DTMF in the Outgoing call
Areski
areski at e-group.org
Mon Feb 16 08:14:18 MST 2004
I found a way to make working the DTMF on the outgoing call with
cisco5300, I have to define dtmfmode=inband in the general context as
on the sample below:
[general]
port = 5060
context = default
disallow=all
allow=alaw
dtmfmode=inband
[usgateway]
type=friend
host=xxx.xxx.x.xx
;dtmfmode=inband
context=sip
The problem is that I cannot make working the DTMF for the incoming and
the outgoing call in the same time.
If I want the DTMF working for incoming call, I must remove from the
general header and setup it in the gateway configuration context as
below:
[general]
port = 5060
context = default
disallow=all
allow=alaw
;dtmfmode=inband
[usgateway]
type=friend
host=xxx.xxx.x.xx
dtmfmode=inband
context=sip
Really weird, NO ?!? Some ideas how to make it working in the both side?
Regards,
Areski
On Fri, 2004-02-13 at 18:25, Clif Jones wrote:
> I have a question about this. It appears that Asterisk will not
> negotiate DTMF mode with a SIP endpoint, so
> whatever this parameter is set to will be the only mode of operation.
> With SIP phones, it appears that you
> can override the global setting on a per-phone basis. SIP gateways do
> not have to have an entry here and
> in my experience do not register. How do I set the DTMF mode for these
> if I have more than one gateway
> and some work with dtmfmode=inband and some with dtmfmode=rfc2833?
>
> Jorge Merlino wrote:
>
> > In the sip.conf file there is a parameter called "dtmfmode" which can
> > take values of inband, rfc2833 or info. Try changing this and see if
> > it helps. X-lite also has an option to configure the DTMF mode, so
> > make it match with what you choose in the sip.conf file.
> >
> > Regards
> > Jorge
> >
> > Areski wrote:
> >
> >> 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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> >>
> >
> >
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