call disconnect from provider because of a late re-register message

in addition to my initial question, how can I set a longer "register expire time" ?

I need urgent help about this.
Any comments would be appreciated.
Thanks.

I am very surprised that no one else has dealt with this problem before.
it is really disturbing. I've checked with some CPE vendors and learned that most of them use policies like:
- send the register at %80 of the expire period.
- send register 5 seconds before expire time.

I think we should be able to apply a similar solution on Asterisk. why not ?

still couldn't find out a workaround. any ideas would be helpful.
thanks.

guys,

I just need a comment on this issue. There might not be a supported solution but there should be a workaround.
Can anyone please advise ?

Thanks.

In your peer the regseconds variable combined with the expiry timer you already changed should get you where you need to be.

ignoreregexpire might also need to be adjusted, this is a global only variable and would need to be placed in sip_general_custom.conf.

I've just read on a forum that "regseconds" is a variable and not a setting.
Are you sure that it would help ? How ?

well, I know how the provider's sip proxy and how it behaves.
I thought I was clear enough when I tried to explain the problem.
in summary, my re-register packets are late sometimes and when that happens my active call is cut. I want to prevent it.
It seems very easy to achieve. If Trixbox would just send the register packet 10 seconds before the expire time, it would be done.
How can I configure it ?

In the given URL (http://www.voip-info.org/wiki-Asterisk+config+sip.conf) , I couldn't find a timer for sending the register message.
"defaultexpirey" is defining the value for this but it is also defining the expire time in the register message.
So, useless...

---

didn't understand that...

How do you know the REGISTER is related to the call disconnect? You need to run a trace and see what is actually disconnecting the call. Search for a thread called "Troubleshooting VoIP Call Quality Issues" in this forum.

What I found was that the messages were being dumped not because the asterisk was doing anything wrong but because the phones were trying to negotiate the RTP media path outofline from the trixbox; which I believe is default behavior and by design.

The problem is depending on how your nat is setup and the rest of the network parts of this communications will be dropped for being out of state.

I simply added 'canreinvite=no' to the trunk configuration and outbound calls were no longer dropping after 22 seconds.

Hope this helps.

--edit this seemed to work last evening but the problem returns this morning -- it is repeatable to a toll free number for a sametime unyte conferencing solution, however it seems that the toll number for the same service does not have problems (atleast today)

Hello

add these lines to your sip.conf in the trunk you want to force the registration to stay alive or in your sip_registrations.conf

registertimeout=60
maxexpiry=60

this will force trixbox to re-register the trunk every minute, after some testing we saw that this time is good to keep any registration alive

I hope this helps

These settings fixed my problem, added to the sip_general_custom.conf file as the other person suggested.
registertimeout=60
maxexpiry=60

My situation was, we have SIP trunks setup between an Avaya server and the Trixbox Asterisk server. The calls were consistently disconnecting after just over 1 minute. After adding these entries the next call I made was good for 5 minutes or more....I hung up myself after about 5 minutes so I think this is what solved the problem. Thanks