Hi list,we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 version and face the following problem: one of our peer (voicetrading.com) doesn't accept our calls anymore, we receive a timeout error "Packet timed out after 32000ms with no response".
Switching back to 1.6 make things working again! In sip.conf we have nat=no, peer conf is: [myPeerDef] type=peer host=111.111.1.111context=honeypot
insecure=invite
directmedia=no
disallow=all
allow=ulaw,alaw
dtmfmode=inband We aren't registered, our IP is authorized by their system. Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their) Working one with 1.6: Audio is at 222.222.22.22 port 26002 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as52190c5c To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22> Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:10:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 284043376 284043376 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26002 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- -- Called myPeerDef/0000033666666666 <--- SIP read from UDP:111.111.1.111:5060 ---> SIP/2.0 183 Session progress Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as52190c5c To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554 Contact: sip:0000033666666666@111.111.1.111:5060 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 159 v=0 o=CARRIER 1338135040 1338135040 IN IP4 77.72.168.74 s=SIP Call c=IN IP4 77.72.168.74 t=0 0 m=audio 18456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 77.72.168.74:18456 Peer doesn't provide video-- SIP/myPeerDef-00000007 is making progress passing it to SIP/104-00000006
<--- SIP read from UDP:111.111.1.111:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as52190c5c To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554 Contact: sip:0000033666666666@111.111.1.111:5060 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 INVITE Server: (Very nice Sip Registrar/Proxy Server) Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE Content-Type: application/sdp Content-Length: 159 v=0 o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74 s=SIP Call c=IN IP4 77.72.168.74 t=0 0 m=audio 18456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (11 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 77.72.168.74:18456 Peer doesn't provide video list_route: hop: <sip:0000033666666666@111.111.1.111:5060>set_destination: Parsing <sip:0000033666666666@111.111.1.111:5060> for address/port to send to
set_destination: set destination to 111.111.1.111, port 5060 Transmitting (no NAT) to 111.111.1.111:5060: ACK sip:0000033666666666@111.111.1.111:5060 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK5afa4cc0;rport Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as52190c5c To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554 Contact: <sip:0033333333333@222.222.22.22> Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 102 ACK User-Agent: TOOTAiAudio Content-Length: 0 --- -- SIP/myPeerDef-00000007 answered SIP/104-00000006Scheduling destruction of SIP dialog '2c974a0a2b08abe320ed388433e47d7e@222.222.22.22' in 32000 ms (Method: INVITE) set_destination: Parsing <sip:0000033666666666@111.111.1.111:5060> for address/port to send to
set_destination: set destination to 111.111.1.111, port 5060 Reliably Transmitting (no NAT) to 111.111.1.111:5060: BYE sip:0000033666666666@111.111.1.111:5060 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK2e708816;rport Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as52190c5c To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554 Call-ID: 2c974a0a2b08abe320ed388433e47d7e@222.222.22.22 CSeq: 103 BYE User-Agent: TOOTAiAudio X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 From 1.8 not working: Audio is at 26704 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- -- Called SIP/myPeerDef/0000033666666666 Retransmitting #1 (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 111.111.1.111:5060: INVITE sip:0000033666666666@111.111.1.111 SIP/2.0 Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Max-Forwards: 70 From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as61c0d9a7 To: <sip:0000033666666666@111.111.1.111> Contact: <sip:0033333333333@222.222.22.22:5060> Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b@222.222.22.22:5060 CSeq: 102 INVITE User-Agent: TOOTAiAudio Date: Sun, 27 May 2012 16:14:39 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer Content-Type: application/sdp Content-Length: 199 v=0 o=root 298186421 298186421 IN IP4 222.222.22.22 s=TOOTAiAudio PBX c=IN IP4 222.222.22.22 t=0 0 m=audio 26704 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=ptime:20 a=sendrecv Thanks for any hint. -- Daniel -- _____________________________________________________________________ -- 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