Zhang Shukun
2011-Jun-16 09:50 UTC
[asterisk-users] sipp application/dtmf-relay not work properly in Asterisk!
hi, everyone
i want to use sipp to auto answer the ivr? to simulate the keypad send
digital sequence? so i try to send DTMF by application/dtmf-relay, but i
have got this error message in the asterisk CLI, Could you help me? Thanks!
[Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't
know how to represent '
the whole CLI message as follows:
<--- SIP read from UDP:211.150.88.154:5067 --->
INFO sip:01025475845 at 211.150.88.155:5060 SIP/2.0
Via: SIP/2.0/UDP 211.150.88.154:5067;branch=z9hG4bK-32222-1-7;rport
From: 1000 <sip:1000 at 211.150.88.154:5067>;tag=1
To: 01025475845 <sip:01025475845 at 211.150.88.155:5060>
Call-Id: 1-32222 at 211.150.88.154
CSeq: 2 INFO
Contact: sip:1000 at 211.150.88.154:5067
Event: dtmf
Content-Type: application/dtmf-relay
Content-Length: 31
Signal= 11037845
Duration= 100
<------------->
--- (10 headers 2 lines) ---
Receiving INFO!
* DTMF-relay event received:
<--- Transmitting (NAT) to 211.150.88.154:5067 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 211.150.88.154:5067
;branch=z9hG4bK-32222-1-7;received=211.150.88.154;rport=5067
From: 1000 <sip:1000 at 211.150.88.154:5067>;tag=1
To: 01025475845 <sip:01025475845 at 211.150.88.155:5060>;tag=as7af6d579
Call-ID: 1-32222 at 211.150.88.154
CSeq: 2 INFO
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Jun 16 17:11:34] WARNING[26321]: rtp.c:3207 ast_rtp_senddigit_begin: Don't
know how to represent '
--
Appreciate your kindly advise and help.
Thanks & Regards
Sucan
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