Hello, I have installed the latest version 12 that has been released (12.1.0.rc3).
I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request (1182 bytes) from UDP:10.25.153.150:5060 ---> INVITE sip:[email protected]:5060;user=phone SIP/2.0 Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2> Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Max-Forwards: 68 From: "39937841 39937841" <sip:39937841;[email protected]:5060 ;user=phone>;tag=02e3a8c0-33807b-t-2 To: <sip:[email protected]:5060;user=phone> Call-ID: [email protected] CSeq: 2 INVITE Contact: <sip:10.1.1.10;line=sr-N6IAzBMsz.MwzxPfPxFsMJZfWBc7MBVuOBV-W.y6MxV*> User-Agent: NetCentrex CCS Softswitch/7.16.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY Supported: 100rel P-Asserted-Identity: "39937841 39937841" <sip:39937841;[email protected]:5060;user=phone> Min-SE: 90 Privacy: none Content-Type: application/sdp Content-Length: 167 v=0 o=10.206.22.171 62708 2 IN IP4 10.206.22.171 s=SIP Call c=IN IP4 10.206.22.171 t=0 0 a=sendrecv m=audio 41040 RTP/AVP 8 a=rtpmap:8 PCMA/8000/1 a=ptime:20 <--- Transmitting SIP response (602 bytes) to UDP:10.25.153.150:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.25.153.150:5060 ;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2> Call-ID: [email protected] From: "39937841 39937841" <sip:39937841;[email protected] ;user=phone>;tag=02e3a8c0-33807b-t-2 To: <sip:[email protected];user=phone> CSeq: 2 INVITE Content-Length: 0 -- Executing [039988120@from-external:1] NoOp("PJSIP/sipp-00000000", " H E L L O ! ! !") in new stack -- Executing [039988120@from-external:2] DumpChan("PJSIP/sipp-00000000", "") in new stack Dumping Info For Channel: PJSIP/sipp-00000000: ================================================================================ Info: Name= PJSIP/sipp-00000000 Type= PJSIP UniqueID= 172.16.60.160-1394542052.0 LinkedID= 172.16.60.160-1394542052.0 CallerIDNum= 39937841;cpc=payphone CallerIDName= 39937841 39937841 ConnectedLineIDNum= (N/A) ConnectedLineIDName=(N/A) DNIDDigits= (N/A) RDNIS= (N/A) Parkinglot= Language= en State= Ring (4) Rings= 1 NativeFormat= (alaw) WriteFormat= alaw ReadFormat= alaw RawWriteFormat= alaw RawReadFormat= alaw WriteTranscode= No ReadTranscode= No 1stFileDescriptor= -1 Framesin= 0 Framesout= 0 TimetoHangup= 0 ElapsedTime= 0h0m0s BridgeID= (Not bridged) Context= from-external Extension= 039988120 Priority= 2 CallGroup= PickupGroup= Application= DumpChan Data= (Empty) Blocking_in= (Not Blocking) Variables: ================================================================================ -- Executing [039988120@from-external:3] Answer("PJSIP/sipp-00000000", "") in new stack <--- Transmitting SIP response (1060 bytes) to UDP:10.25.153.150:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.25.153.150:5060 ;rport;received=10.25.153.150;branch=z9hG4bK587.67258295.0 Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuqC93X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJ3lqwWxarW.gqWReJMEPA36juW6WBzR363RVA3Ejugx3* Record-Route: <sip:10.25.153.150;lr;ftag=02e3a8c0-33807b-t-2> Call-ID: [email protected] From: "39937841 39937841" <sip:39937841;[email protected] ;user=phone>;tag=02e3a8c0-33807b-t-2 To: <sip:[email protected] ;user=phone>;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c CSeq: 2 INVITE Contact: <sip:172.16.60.160:5060> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER, REGISTER Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 193 v=0 o=- 62708 4 IN IP4 172.16.60.160 s=Asterisk c=IN IP4 172.16.60.160 t=0 0 m=audio 13644 RTP/AVP 8 c=IN IP4 172.16.60.160 a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP request (703 bytes) from UDP:10.25.153.150:5060 ---> ACK sip:172.16.60.160:5060 SIP/2.0 Via: SIP/2.0/UDP 10.25.153.150:5060;branch=z9hG4bKcydzigwkX Via: SIP/2.0/UDP 10.1.1.10;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.qLOBF6zGZLMBj7MBvuMx3AMB1jmxuq3w3X3heroEWvH9vsCFN43qdAMxyAMxyAMxyAMlMZMxpJWBeIME3ugSVwWx3A3BPAMxIqg.jZWxqL3BqwMRjsW.j* Max-Forwards: 67 From: "39937841 39937841" <sip:39937841;[email protected]:5060 ;user=phone>;tag=02e3a8c0-33807b-t-2 To: <sip:[email protected]:5060 ;user=phone>;tag=b23cda89-931c-4a95-85c5-0ec8b03f895c Call-ID: [email protected] CSeq: 2 ACK User-Agent: NetCentrex CCS Softswitch/7.16.0 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, INFO, PRACK, UPDATE, NOTIFY Content-Length: 0 > 0x99694c0 -- Probation passed - setting RTP source address to 10.206.22.171:41040 -- Executing [039988120@from-external:4] Read("PJSIP/sipp-00000000", "dataEntry,"why-no-answer-mystery",10,,1,4") in new stack -- Accepting a maximum of 10 digits. -- <PJSIP/sipp-00000000> Playing 'why-no-answer-mystery.alaw' (language 'en')
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
