[Asterisk-Dev] NO DTMF in the Outgoing call
Clif Jones
ctjones at earthlink.net
Fri Feb 13 10:25:11 MST 2004
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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