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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110616/5073f824/attachment.htm>