[Freeswitch-users] Dynamic SIP Gateways to register with

kokoska rokoska kokoska.rokoska at post.cz
Wed Apr 9 07:29:22 PDT 2008

Anthony Minessale napsal(a):
> Asterisk does let you reload, and it *tries* not to interrupt the calls 
> but it's like a game of Russian Roulette,
> because at some point for sure you *will* deadlock and/or seg fault when 
> you reload. 

Yes, Anthony, I heard/red about it a lot :-)
But form general user point of view it is better to have a chance it 
will work (calls stay alive) than have guarantee it will not work (calls 
go down). The worst can happen is to end up in same situation - all 
calls are dropped.

> There is an actual unavoidable race condition in many of 
> the things that they made reloadable  that  I have identified myself as 
> a long  time developer for the project and to this day still exist and 
> are a major bullet point to address on their roadmap for future 
> releases. 1.8 maybe?

Yes, I know you were/are Asterisk developer. And I realise that Asterisk 
without your work on ARA remain a toy.
And, of course, I'm very thankful to you!

> Our philosophy on reload is simple and strict.  Each module is 
> responsible independently for it's own ability to reload.
> In the case of sofia, it was a year after it was written until I 
> introduced the profile restart command because it's
> quite complicated and introduces a great challenge in stability.  Nearly 
> all of the elements in a sofia profile are things you must stop the 
> profile to change anyway. 

OK. Thank you very much, Anthony, for explanation. I didn't know that.

> There are a few innocuous params  that could be changed while it's 
> running  so  I will say there is a possibility to make a sofia reload 
> and a sofia profile reload that unlike profile restart looks for 
> profiles that do not yet exist and brings them online and when they do 
> exist only changes the params that do not require a full restart of the 
> sip engine (context to use, dialplan, moh prefs vs bind url, sip 
> specific options that cannot be changed).

Thank you for explanation again! But I'm interested only in "sip 
specific" parts of profile :-(

> As part of this process the profiles could be rescanned for new gateways 
> that do not already exist and bring them online if they do not already 
> exist.  Removing them would require a full restart of that profile.

OK. Thanks for the info!

> This idea I am willing to entertain but in the light of my horribly busy 
> schedule and the fact that the patch needs near surgical precision to 
> avoid tainting our release candidate state of stability. Not to mention 
> I have coded nearly 20,000 lines of mod_sofia pro bono  providing every 
> SIP feature anyone can ask for

Anthony, I realy respect all work you have done on Asterisk, FreeSWITCH, 
Sofia etc and I appreciate your help to me and all users too.
And, in particular, I just try to become FreeSWITCH user and thus 
looking for improvements helpful for everybody, I'm not your enemy.

> I would like to see the bounty for it 
> first. =D
OK. Give me a while :-)

Best regards,


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