Hi, I?ve installed Asterisk for use as a SIP server. I can call people, but one strange thing happens: if I call someone with a SIP account outside my server (for example, sip:enum-echo-test at sip.nemox.net) everything is fine, if I call any Asterisk extension it also works, but the call gets disconnected in about 20 seconds. To be exact, audio is turned off but the SIP client still thinks it?s connected. Logs say ?no reply to our critical packet?. tcpdump shows that the packet does arrive at the destination. sip set debug shows this is what the packet contains: Retransmitting #6 (NAT) to 77.239.189.223:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=77.239.189.223 From: "Roma"<sip:roma at qwertty.com;transport=UDP>;tag=01785d5e To: <sip:echo at qwertty.com;transport=UDP>;tag=as068592d2 Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:echo at 78.46.49.80> Content-Type: application/sdp Content-Length: 285 v=0 o=root 25952 25952 IN IP4 78.46.49.80 s=session c=IN IP4 78.46.49.80 t=0 0 m=audio 30606 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv There?s NAT: computer (192.168.1.2) behind a router (77.239.189.223), the server (78.46.49.80) doesn?t have any NAT. I have even set DMZ host to 192.168.1.2, so I?m sure all packets reach it. As far as I understand, Asterisk expects the SIP client to reply to that packet with an ACK, the client receives the packet but does not reply. What have I configured incorrectly? In sip.conf I have nat=yes (otherwise I don?t hear anything), whatever I do with NAT settings of SIP clients does not help. Maybe there?s something wrong with the headers of the packet that makes the client think the packet is misaddressed? Twinkle says, ?you have the following registrations <sip:roma at 192.168.1.2>? while I?d expect <sip:roma at qwertty.com>. So how do I make sure the client sends its ACK? -- TIA Roman. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2272 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090313/fe85a9ab/attachment-0001.bin
-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Roman Odaisky Sent: Friday, March 13, 2009 9:57 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] No reply to our critical packet Hi, I've installed Asterisk for use as a SIP server. I can call people, but one strange thing happens: if I call someone with a SIP account outside my server (for example, sip:enum-echo-test at sip.nemox.net) everything is fine, if I call any Asterisk extension it also works, but the call gets disconnected in about 20 seconds. To be exact, audio is turned off but the SIP client still thinks it's connected. Logs say "no reply to our critical packet". tcpdump shows that the packet does arrive at the destination. sip set debug shows this is what the packet contains: Retransmitting #6 (NAT) to 77.239.189.223:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=7 7.239.189.223 From: "Roma"<sip:roma at qwertty.com;transport=UDP>;tag=01785d5e To: <sip:echo at qwertty.com;transport=UDP>;tag=as068592d2 Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:echo at 78.46.49.80> Content-Type: application/sdp Content-Length: 285 v=0 o=root 25952 25952 IN IP4 78.46.49.80 s=session c=IN IP4 78.46.49.80 t=0 0 m=audio 30606 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv There's NAT: computer (192.168.1.2) behind a router (77.239.189.223), the server (78.46.49.80) doesn't have any NAT. I have even set DMZ host to 192.168.1.2, so I'm sure all packets reach it. As far as I understand, Asterisk expects the SIP client to reply to that packet with an ACK, the client receives the packet but does not reply. What have I configured incorrectly? In sip.conf I have nat=yes (otherwise I don't hear anything), whatever I do with NAT settings of SIP clients does not help. Maybe there's something wrong with the headers of the packet that makes the client think the packet is misaddressed? Twinkle says, "you have the following registrations <sip:roma at 192.168.1.2>" while I'd expect <sip:roma at qwertty.com>. So how do I make sure the client sends its ACK? -- TIA Roman. --> Two thoughts (both could be wrong) 1. Do you have the incoming 10000-20000 holes in your firewall so the remote server can get it's reply back to *? 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Danny Nicholas
I have the same situation. My scenario is weird: I have a DID with IPkall that points to my asterisk server, and I have it play a message with Playback() after about 20 seconds call drops and give me the same message you get: "no reply to our critical packet" BUT I have a DID from Vitelity, and that one works fine no drops. I have no idea why. On Fri, Mar 13, 2009 at 12:37 PM, Roman Odaisky <roma at qwertty.com> wrote:> On Friday, 13.03.2009 17:50:57 Danny Nicholas wrote: > > > Next Step would be to check/update the firmware on your phones or router. > > I don?t think the router is to blame, it does deliver all the packets. And > there are no hardware phones, only numerous software SIP clients. > > Which (GNU/Linux) software clients are known to have maximum compatibility > with Asterisk? > > -- > TIA > Roman. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090313/23a57874/attachment.htm