[Asterisk-Dev] NO DTMF in the Outgoing call

Wed Feb 11 08:50:05 MST 2004

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