Hello, you have a typicall nat issue. Asterisk receives messages from the phone but cannot send any messages back (thats why it tries to resend the 200 ok message 6 times).
try setting qualify=yes to your sip peers config to keep the nat port open. best regards stefan Am 22.12.10 13:44, schrieb Gilles: > Hello > > I have an Asterisk 1.4 server and two XLite softphones, where > Asterisk and the local XLite phone are located in a LAN behind a NAT > router, and the remote XLite phone is located elsewhere on the Net > behind its own NAT router: > > http://img252.imageshack.us/img252/3749/asterisknat.png > > I'm having the following issue: When the _local_ XLite calls out the > remote XLite, everything works fine; However, when the _remote_ XLite > calls the local XLite, things work OK until precisely 20s, where > Asterisk decides to hang up, and displays the following error message > in the console: > > ================== > WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on > transmission > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno > 2 (Critical Response) > > WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call > e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no > reply to our critical packet. > == Spawn extension (my-phones, local-xlite-extension, 1) exited > non-zero on 'SIP/unused-008008e4' > ================== > > I'm no SIP expert, but based on the debug session, before deciding to > hang up, Asterisk tries 6 times to send an OK message to the remote > XLite, and doesn't seem to get an answer. FWIW, after Asterisk has > hung up, the remote XLite remains off-hook, oblivious to this error > and keeps displaying "Call established": > > www.pastebin.com/x6MgnrpG > > There's also this oddity on line 50: "Scheduling destruction of SIP > dialog". > > FWIW, in sip.conf, for the remote XLite user, I tried "nat=no" and > "nat=yes", with no difference. I'm actually not sure how to configure > a remote user which happens to be listed in sip.conf (it's behind a > NAT router but it registers with Asterisk, so... is it NATed or not?), > and am surprised it actually rings and sends/receives voice with no > problem, regardless of this parameter. > > I found discussions about using "t1min=500" in sip.conf, but it made > no difference either. > > Has someone already experienced this and knows what can be done? > > Any hint much appreciated. > > > -- > _____________________________________________________________________ > -- 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 -- Für weitere Fragen stehen wir gerne unter [email protected] oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // [email protected] // Tel 059944-2440// ------------------------------------------------- SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // ------------------------------------------------- -- _____________________________________________________________________ -- 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
