[Asterisk-Dev] NO DTMF in the Outgoing call

[Jorge Merlino]

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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