[asterisk-users] Unstable phone connection

[Bryant Zimmerman]

D'Arcy J.M. Cain 
  
 If the device is registering and then dropping there are several usual 
items.  
 The router may be closing the ports on the device. 
 The router may have a AGL SIP helper that is causing issues. 
  
 Make sure that the device is sending out keep alive packets.
 Shut down any AGL helpers on the router.
 Make sure that the site is not double NATing
  
 Try using a stun server and see if that helps at all.
 Watch you console on your sip serer to see how long the device runs before 
losing connection.
  
 Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
  

----------------------------------------
 From: "D'Arcy J.M. Cain" <darcy at Vex.Net>
Sent: Thursday, March 12, 2015 2:40 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Unstable phone connection   
This is driving me to distraction. I have a switch with multiple
clients who are all working fine except for one and I can't figure out
what makes them different. I have tried every NAT setting in the ATA
(SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different
sip ports, different RTP ports and it still fails. I have left the
location with it working only to have it fail later. He always gets
registered but when a call is sent it doesn't respond so the caller
hears no ring and the phone does not ring.

Yesterday he mentioned that when the phone is working the WiFi slows
down significantly. No idea why or if it is related.

He has a radio station streaming music. I wondered if that might be
interfering. That's why I tried changing the SIP port and the RTP
ports but that didn't seem to help.

It smells like a network problem to me but I am running the same ADSL
device here and other clients are working behind a NAT gateway so I am
at a loss as to what might be wrong. Could it be the streaming?

Cheers.

--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy at Vex.Net
VoIP: sip:darcy at Vex.Net

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150312/7ba85ee2/attachment.html>