Hi all, I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5. The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, everything is ok (negociation and phone call) but when we try to use the voicemail, Asterisk don't understand DTMF.
Here are some logs (SIP debug on) on a DTMF '2' receive : ************************************************************************* Sip read: INFO sip:[EMAIL PROTECTED]:5060 SIP/2.0 Call-ID: [EMAIL PROTECTED] Contact: <sip:0.0.0.0:5060> Content-Type: application/dtmf CSeq: 2047555 INFO From: "test003" <sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000007ad91f06 Max-Forwards: 31 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2283a438 Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-5764-36A3EE Content-Length: 3 2 10 headers, 1 lines Receiving DTMF! Feb 28 08:27:44 WARNING[9223]: chan_sip.c:6116 receive_info: Unable to parse INFO message from [EMAIL PROTECTED] Content Transmitting (NAT): SIP/2.0 415 Unsupported media type Via: SIP/2.0/UDP 0.0.0.0:5060;branch=z9hG4bK-5764-36A3EE;received=0.0.0.0.0;rport=5060 From: "test003" <sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000007ad91f06 To: <sip:[EMAIL PROTECTED];user=phone>;tag=as2283a438 Call-ID: [EMAIL PROTECTED] CSeq: 2047555 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 *************************************************************************** dtmfmode is RFC2833 in sip.conf Thx in advance. Regards. Guy Decarpentrie. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
