Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Unstable phone connection


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
darcy at Vex.Net
Guest





PostPosted: Thu Mar 12, 2015 1:39 pm    Post subject: [asterisk-users] Unstable phone connection Reply with quote

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@Vex.Net
VoIP: sip:darcy@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
Back to top
BryantZ at zktech.com
Guest





PostPosted: Thu Mar 12, 2015 2:15 pm    Post subject: [asterisk-users] Unstable phone connection Reply with quote

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@Vex.Net> Sent: Thursday, March 12, 2015 2:40 PM To: asterisk-users@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@Vex.Net VoIP: sip:darcy@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
Back to top
darcy at Vex.Net
Guest





PostPosted: Thu Mar 12, 2015 2:58 pm    Post subject: [asterisk-users] Unstable phone connection Reply with quote

On Thu, 12 Mar 2015 15:14:24 -0400
"Bryant Zimmerman" <BryantZ@zktech.com> wrote:
Quote:
If the device is registering and then dropping there are several
usual items.
The router may be closing the ports on the device.

I don't see how. I am logged into the ATA through the router and I
don't lose the connection.

Quote:
The router may have a AGL SIP helper that is causing issues.

Can't find an AGL setting. There is a SIP checkbox. Pretty sure I
have that turned off but I can try to get someone to check.

Quote:
Make sure that the device is sending out keep alive packets.

I have that flag turned on.

Quote:
Shut down any AGL helpers on the router.

See above.

Quote:
Make sure that the site is not double NATing

There's only one router. It is the ADSL device as well.

Quote:
Try using a stun server and see if that helps at all.

I tried with and without. I am using stunserver.org.

Quote:
Watch you console on your sip serer to see how long the device runs
before losing connection.

I don't think it does. Both the server and the ATA think that they are
still registered but when a call comes in there is no ringing on the
line. If I split dial it rings the cell phone but I still hear no
ringing from the caller side unless registration is actually turned off
from the ATA and a "sip unregister" is issued.

--
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:darcy@Vex.Net
VoIP: sip:darcy@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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services