[asterisk-users] SIP Problems continue...

[Ken Williams]

My configs that I've reworked in the process of trying to fix this SIP
problem actually started from Freepbx.
 
I removed and reinstalled Asterisk last night, things seem to be working
smoother, I'll no by noon if the problem is fixed or not.
 
Thanks for the help from everyone,
Ken

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Deepak
Naidu
Sent: Wednesday, May 09, 2007 11:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SIP Problems continue...


A small way to make little easy, I dont know it people are ok to that,
try integrating freepbx & asterisk so you know what the sip configs
should look like when things are all well.
 
Things might stop working if there is a bug or change in configs.
 
--
Deepak

Ken Williams <ken at intermountainelectronics.com> wrote:

	I mean that SIP phones cannot answer incoming calls or make
outgoing
	calls. When a call comes in on ZAP, it actually rings all the
phones
	like normal, but when you try to answer no one is there. In
addition,
	when you try to dial out you eventually get a message on the
phones
	saying unable to communicate with the server. So there is some
traffic
	still traveling on the SIP channel (the server's dialing
extensions from
	an incoming ZAP call) but no further communication...almost as
if it's a
	one way street of communication. The server can send data out on
SIP
	but isn't receiving any.
	
	As for your issue, we haven't really had that (thankfully), so I
don't
	think you're heading down the horrible spot we're in right now.
	
	Tonight I'm going to remove all aspects of Asterisk and
reinstall fresh,
	if that fails I'll format & reinstall the entire box. 
	
	-----Original Message-----
	From: asterisk-users-bounces at lists.digium.com
	[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Adam
	Moffett
	Sent: Wednesday, May 09, 2007 12:08 PM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: Re: [asterisk-users] SIP Problems continue...
	
	I also get the mysterious SIP INVITE channels.
	10.101.2.204 xxx 748e8b0a625 00102/00000 unkn No Init:
	INVITE
	
	And I also am running 1.4.4 on CentOS4. Is that a pattern or
just
	coincidence?
	
	
	
	The other symptom you mention is this
	"...the SIP phones couldn't communicate with the server, though
there 
	was no error message on the server and everything appeared fine
on the 
	server."
	
	Do you mean no calls in or out until you reboot? I don't have
that 
	thankfully, but I do have a guy telling me that incoming audio
just goes
	
	away for a few seconds at a time. He says also that it sometimes
goes 
	away for long enough time that he was mistaking it for a dropped
call. 
	But if he waits long enough it pretty generally always comes
back. I 
	have consistent solid network performance from the asterisk
server to 
	the ATA (and believe me, I've looked very hard for a network
problem), 
	and I don't know what to look at next.
	
	Incidentally, the guy hasn't called me since I rebooted last
week. Is 
	this similar to how your situation started?
	
	
	
	*********************************
	Adam Moffett
	Plexicomm, LLC
	adam at plexicomm.net
	ph: 866-759-4678x104
	*********************************
	
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