Trying internal call 100 -> 101 Asterix+FreePBX (192.168.1.2) 100: Cisco 7941 (192.168.1.158) 101: Grandstream HT486 (192.168.1.3) Call is dropped with busy signal due to "603 Declined", see debug below. ---8<--- <--- SIP read from 192.168.1.158:49182 ---> INVITE sip:101@miraculix.int.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 Max-Forwards: 70 Date: Wed, 30 Jan 2008 GMT CSeq: 101 INVITE User-Agent: Cisco-CP7941G/8.0 Contact: Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 21243 0 IN IP4 192.168.1.158 s=SIP Call t=0 0 m=audio 30194 RTP/AVP 0 8 18 101 c=IN IP4 192.168.1.158 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (18 headers 13 lines) --- Sending to 192.168.1.158 : 5060 (no NAT) Using INVITE request as basis request - 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 <--- Reliably Transmitting (no NAT) to 192.168.1.158:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded;received=192.168.1.158 From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: ;tag=as6fc5b9bc Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7736ab5a" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158' in 32000 ms (Method: INVITE) Found user '100' miraculix*CLI> <--- SIP read from 192.168.1.158:50468 ---> ACK sip:101@miraculix.int.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbbb04ded From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: ;tag=as6fc5b9bc Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 Date: Wed, 30 Jan 2008 GMT CSeq: 101 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- miraculix*CLI> <--- SIP read from 192.168.1.158:49182 ---> INVITE sip:101@miraculix.int.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 Max-Forwards: 70 Date: Wed, 30 Jan 2008 GMT CSeq: 102 INVITE User-Agent: Cisco-CP7941G/8.0 Contact: Proxy-Authorization: Digest username="100",realm="asterisk",uri="sip:101@miraculix.int.mydomain.com",response="9049ba2387fb625bb0d087c32128d18a",nonce="7736ab5a",algorithm=MD5 Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Supported: replaces,join,norefersub Allow-Events: kpml,dialog Content-Length: 275 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 21243 0 IN IP4 192.168.1.158 s=SIP Call t=0 0 m=audio 30194 RTP/AVP 0 8 18 101 c=IN IP4 192.168.1.158 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (19 headers 13 lines) --- Sending to 192.168.1.158 : 5060 (no NAT) Using INVITE request as basis request - 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 Found user '100' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.158:30194 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.158:30194 Looking for 101 in from-internal (domain miraculix.int.mydomain.com) list_route: hop: miraculix*CLI> <--- Transmitting (no NAT) to 192.168.1.158:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a;received=192.168.1.158 From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Scheduling destruction of SIP dialog '001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158' in 32000 ms (Method: INVITE) <--- Reliably Transmitting (no NAT) to 192.168.1.158:5060 ---> SIP/2.0 603 Declined Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a;received=192.168.1.158 From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: ;tag=as1f7d6359 Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> miraculix*CLI> <--- SIP read from 192.168.1.158:50469 ---> ACK sip:101@miraculix.int.mydomain.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.158:5060;branch=z9hG4bKbfaa102a From: "100" ;tag=001aa17bdcb002914ca32cfc-074f6117 To: ;tag=as1f7d6359 Call-ID: 001aa17b-dcb00064-82f8c96e-1475c011@192.168.1.158 Date: Wed, 30 Jan 2008 GMT CSeq: 102 ACK Content-Length: 0 ---8<--- Relevant snippet from extensions_additional.conf: ---8<--- [ext-local] include => ext-local-custom exten => 100,1,Macro(exten-vm,100,100) exten => 100,n,Hangup exten => 100,hint,SIP/100 exten => ${VM_PREFIX}100,1,Macro(vm,100,DIRECTDIAL) exten => ${VM_PREFIX}100,n,Hangup exten => vmb100,1,Macro(vm,100,BUSY) exten => vmb100,n,Hangup exten => vmu100,1,Macro(vm,100,NOANSWER) exten => vmu100,n,Hangup exten => vms100,1,Macro(vm,100,NOMESSAGE) exten => vms100,n,Hangup exten => 101,1,Macro(exten-vm,101,101) exten => 101,n,Hangup exten => 101,hint,SIP/101 exten => ${VM_PREFIX}101,1,Macro(vm,101,DIRECTDIAL) exten => ${VM_PREFIX}101,n,Hangup exten => vmb101,1,Macro(vm,101,BUSY) exten => vmb101,n,Hangup exten => vmu101,1,Macro(vm,101,NOANSWER) exten => vmu101,n,Hangup exten => vms101,1,Macro(vm,101,NOMESSAGE) exten => vms101,n,Hangup ---8<---