[asterisk-users] Asterisk hangs up call after 20s

[Stefan Schmidt]

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.
> 
> 
> --
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-- 
Stefan Schmidt
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