[Asterisk-Dev] NO DTMF in the Outgoing call
Areski
areski at e-group.org
Tue Feb 17 11:33:24 MST 2004
Hi folks,
I fix my problem using SIPDtmfMode() function.
I didn't know about this function... too stupid ;(
Regards,
Areski
On Mon, 2004-02-16 at 16:14, Areski wrote:
> 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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