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Posts: 37 | Thanked: 9 times | Joined on Jan 2011
#1
So SIP works great. Send and receive calls. Fine. I go on about my day. Call comes in. I notice that it doesn't come into the n900. I take the call.

Shortly thereafter, I check asterisk ("sip show peers") ... hrm. Not registered.

So... I've been iterating. If I "edit" the sip account on the phone and hit "save" ... it immediately re-registers and everything works. I set the keep-alive interval down to 2 minutes (although my FreeBSD firewall is quite good --- whatever it takes).

But still. After some amount of time that I haven't precisely measured, SIP stops registering.

Has anyone taken a crack at this problem? My search-foo isn't good enough to tease this out of the board, here.
 
Posts: 37 | Thanked: 9 times | Joined on Jan 2011
#2
Both to give this a bump and to give more information, I ran a TCP dump overnight and the phone just stopped registering for no reason I can divine.

In particular, while it's working you get pairs of 3 packets Register->Trying->Unauthorized followed by Register->Trying->OK. This is normal --- the first attempt is without password and the Unauthorized returns a hash. The second uses the hash to login. Good.

But at the end of the dump, the last new SIP attempts only do the first part --- they don't retry with the password. Then after 2 or 3 of those attempts, attempts stop alltogether.
 
Posts: 37 | Thanked: 9 times | Joined on Jan 2011
#3
OK... I think I may see the problem here. Each REGISTER adds one line:

Contact: <sipaveN900@xx.yy.zz.aa:57599;transport=udp>;exp ires=0

... and the packet gets longer and longer until it's chopped off and invalid. At a 2 minute refresh, it shouldn't be rolling around different port mappings. Even if it was, it should only keep a limited number of the Contact: lines is subsequent packets.
 
Posts: 306 | Thanked: 106 times | Joined on Feb 2010
#4
There was a bug report in Freeswitch jira about this. I have a keep alive of 1 minute and registration works fine with asterisk.
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Voice choppy on sip calls
Please vote for bug number 10388
 

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