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Leonmeijer

Registers but times out always

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Hello I'm running PBXnSIP 4.5.1.1107 Zeta Perseids (Win64)

 

And extensions are just unable to connect;

Not from my home network.

Not from any smartphone.

Not from the Win 2008 server where the PBX runs on.

 

I'm getting quite mad about it because I've been trying a few years now and just registered the phones to the sip registars directly instaed of using the product.

I'm not going to upgrade to v5 because the license costs and it's a pitty there is no newer v4.5 version around.

 

Registrationg goes OK, but after 15 secs times out (there is no 15 second timeout set anywhere on the system).

 

Directly when I try to call after registration to 8100 (voicemail) it goes:

[6] 2013/05/31 14:06:18: SIP TCP/TLS timeout on my.external.ip.address:49984, closing connection

 

and this one:

 

[9] 2013/05/31 14:06:12: Registration for account "Test Phone" <sip:100@localhost> expired, removing contact <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>"

 

TLS is disabled, tried both UDP and TCP.

Firewall settings are OK even without firewall I have no luck.

 

Can someone PLEASE get this fixed?

 

 

 

[5] 2013/05/31 14:06:07: SIP Tx tcp:my.external.ip.address:49984:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-4jmrch;rport=49984;received=my.external.ip.address

From: "100" <sip:100@localhost>;tag=na1r0k

To: "100" <sip:100@localhost>;tag=f0163d3e69

Call-ID: gkjhts0d@snom

CSeq: 1474 SUBSCRIBE

User-Agent: snomONE/4.5.1.1107 Zeta Perseids

WWW-Authenticate: Digest realm="localhost",nonce="52c1ffa043d16f26daaf05da27c81027",domain="sip:100@localhost",algorithm=MD5

Content-Length: 0

 

[5] 2013/05/31 14:06:07: SIP Rx tcp:my.external.ip.address:49984:

SUBSCRIBE sip:100@localhost SIP/2.0

Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-qlbtdt;rport

From: "100" <sip:100@localhost>;tag=na1r0k

To: "100" <sip:100@localhost>

Call-ID: gkjhts0d@snom

CSeq: 1475 SUBSCRIBE

Max-Forwards: 70

Contact: <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>"

Supported: outbound, gruu

Event: message-summary

Accept: application/simple-message-summary

User-Agent: snom-YYY/9.3.7XXX

Authorization: Digest realm="localhost",nonce="52c1ffa043d16f26daaf05da27c81027",response="d98297037105ea646ae9719dee7d37ce",username="100",uri="sip:100@localhost",algorithm=MD5

Expires: 3600

Content-Length: 0

 

[5] 2013/05/31 14:06:07: SIP Tx tcp:my.external.ip.address:49984:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP my.internal.ip.address:49984;branch=z9hG4bK-qlbtdt;rport=49984;received=my.external.ip.address

From: "100" <sip:100@localhost>;tag=na1r0k

To: "100" <sip:100@localhost>;tag=f0163d3e69

Call-ID: gkjhts0d@snom

CSeq: 1475 SUBSCRIBE

Contact: <sip:the.ip.of.the.pbx:5060;transport=tcp>

Require: outbound

Date: Fri, 31 May 2013 12:06:07 GMT

Server: snomONE/4.5.1.1107 Zeta Perseids

Expires: 0

Content-Length: 0

 

[9] 2013/05/31 14:06:12: Registration for account "Test Phone" <sip:100@localhost> expired, removing contact <sip:100@my.internal.ip.address:49984;transport=tcp;line=j8766z>;reg-id=1;+sip.instance="<urn:uuid:396b5946-418e-4f9c-b571-45e238ed2424>"

[6] 2013/05/31 14:06:18: SIP TCP/TLS timeout on my.external.ip.address:49984, closing connection

[5] 2013/05/31 14:06:18: Registration 8vjv97sl@snom closed connection, removing it

[5] 2013/05/31 14:06:18: Registration gkjhts0d@snom closed connection, removing it

[8] 2013/05/31 14:06:18: Release SIP thread 4

[8] 2013/05/31 14:06:56: Received SIP connection 5 from my.external.ip.address:49991

[5] 2013/05/31 14:06:56: SIP Rx tcp:my.external.ip.address:49991:

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I would first narrow down the problem by having a phone in the LAN, where there is no firewall or router in the middle. That helps to find out where exactly the problem is.

 

The PBX (version 4 as well as version 5) closes the connection after some time if there was no traffic on the socket (usually after just a few seconds). Usually the device registers, and then the PBX changes the timeout to the registration interval. You should be able to see in the REGISTER response what the value is.

 

In the SUBSCRIBE response, you see a value of 0. That is raising questions. You can check the pnp.xml file for the parameters "nat_tcp" and "max_expires" if there is anything unusual.

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I would first narrow down the problem by having a phone in the LAN, where there is no firewall or router in the middle. That helps to find out where exactly the problem is.

 

The PBX (version 4 as well as version 5) closes the connection after some time if there was no traffic on the socket (usually after just a few seconds). Usually the device registers, and then the PBX changes the timeout to the registration interval. You should be able to see in the REGISTER response what the value is.

 

In the SUBSCRIBE response, you see a value of 0. That is raising questions. You can check the pnp.xml file for the parameters "nat_tcp" and "max_expires" if there is anything unusual.

 

Thanks for the info, there's no such file pnp.xml but in pbx.xml there's:

 

"<min_expires/><max_expires/>"

 

(no value, so both are probably set to 0), what are correct values for these properties?

 

nat_tcp is set to 180 there.

 

The same LAN gives the same result even using the same machine through 127.0.0.1 / localhost doen't do anything which I guess should 'always' work...

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Set min_expires to 30 and max_expires to 300...

 

Set these values by editing the pbx.xml file, restarted it and it seems to work again which is great, thanks, but now the audio sounds crackling instead of hearing a voice when e.g. calling the mailbox. Guess this might be related to codec preference?

 

What is the default order for codecs and which ones can be better removed from the list?

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Two-way crackling kind of good news. That sounds like problems with the packet delivery. How much bandwidth is available between the PBX and the server? Any chance that packets can get lost? The codecs have a different sensibility for packet loss; but if there is no packet loss every codec should sounds smooth.

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Two-way crackling kind of good news. That sounds like problems with the packet delivery. How much bandwidth is available between the PBX and the server? Any chance that packets can get lost? The codecs have a different sensibility for packet loss; but if there is no packet loss every codec should sounds smooth.

 

 

The available upstream is about 10 Mbit and the downstream is 120 Mbit (the server itself has a +/- 400Mbit upstream).

 

I checked my snom 360 it actually has "better" audio then the softphone but the softphone (the snom one) didn't want to install because the visual C++ was "missing" and the VC++ setup told me that a newer version is allready installed so I just renamed all the files and ran it...

 

Connections from a mobile phone to the pbx (from mobile to my voip provider to the pbx) are fine so the problem should be in my LAN then. I will check the router and probably I'll get it fixed.

 

Thanks for the help and I'll be back here if i'm having issues fixing it.

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