Re: [asterisk-users] SOLVED: No reply to our critical packet
Hi, Next Step would be to check/update the firmware on your phones or router. I dismissed this advice at first, but it was the one that worked in the end. The D-Link DSL-2500U ADSL router was to blame, it must have been interfering with SIP packets (maybe an outdated version of the SIP conntrack module or something like that). The 1.50 firmware version solved the problem and also gave the impression of working faster overall than 1.20. Thanks a lot. Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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
[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-t...@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: Romasip:r...@qwertty.com;transport=UDP;tag=01785d5e To: sip:e...@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:e...@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:r...@192.168.1.2” while I’d expect sip:r...@qwertty.com. So how do I make sure the client sends its ACK? -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] No reply to our critical packet
Hi, thanks for the quick reply. 1. Do you have the incoming 1-2 holes in your firewall so the remote server can get it's reply back to *? This was what I checked first. Both firewalls let everything through. 2. If #1 is ok, try putting an Answer command in front of your Dial Command. Doesn’t help, alas. Also, it works the same (disconnect after 20 seconds) both for Dial and Echo, regardless of presence of Answer. -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] No reply to our critical packet
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. smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] No reply to our critical packet
Ringing() followed by Wait(1) I made it exten = echo,1,Ringing() exten = echo,2,Wait(1) exten = echo,3,Playback(abandon-all-hope) exten = echo,4,Echo() to no avail. This looks like a client issue, though all of my clients fail. Which clients are the most standards conforming? Also, maybe Asterisk isn’t what I need? I need a server with which several people would register accounts, they’ll be able to place calls among themselves and also call other SIP accounts, as well as calling PSTN numbers using predefined accounts (so that it would be possible to share one paid account between several users). So a SIP server + possibility for dialplans through 3rd party SIP servers. Maybe something like SER would suffice? Or SER as a proxy in front of Asterisk is the way to go? -- TIA Roman. smime.p7s Description: S/MIME cryptographic signature ___ -- 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
