Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Michael Graves



Sounds like the registration interval in the phones is less than the required registration interval of the server. I had this occur when using a SIP phone with an ITSP.



Michael



On Wed, 28 Jun 2006 12:04:40 -0400, Von L. wrote:



>Hello,

>

>Here is a breakdown of the issue I am experiencing. I have three remote

>employees, in various states, who have Polycom 501 phones. They are

>unable to receive incoming calls after a few minutes of the phones being

>plugged in. They work immediately after being plugged in, but they lose

>the ability shortly thereafter. They can always make outbound calls, but

>only to real phone numbers, not extensions.

>

>They each have NAT routers, and I have triple checked that they have

>opened/forwarded the correct ports, basically 5060-3 UDP. Once they

>plug the phone it (power and ethernet) I see on the CLI console of the

>asterisk server that the phones register:

>

>Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.

>Written by Mark Spencer <[EMAIL PROTECTED]>

>=

>Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on

>bell (pid = 3652)

>nell*CLI>

>Verbosity is at least 10

>-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600

>

>Here is the top part of my sip.conf

>

>;_

>;sip.conf

>;_

>

>[general]

>port=5060

>bindaddr=0.0.0.0

>externip=XXX.XXX.XXX.XXX

>localnet=XXX.XXX.XXX.XXX/255.255.255.248

>canreinvite=no

>tos=reliability

>srvlookup=yes

>disallow=all

>allow=ulaw

>dtmfmode=rfc2833

>nat=yes

>ignoreregexpire=yes

>

>I know it has something to do with the NAT because if I plug my Polycom

>directly into my cable modem, thus making it sit on the Internet and

>have a real IP, everything works just fine.

>

>I am curious what I am missing.

>

>Thanks.

>

>Von L.

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>

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>To UNSUBSCRIBE or update options visit:

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>






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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Dr. Michael J. Chudobiak

Von L. wrote:

plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.

They each have NAT routers, and I have triple checked that they have
opened/forwarded the correct ports, basically 5060-3 UDP. Once they



See the "NAT Issues" section at http://www.voip-info.org/wiki/view/IAX.
(The page is for IAX2, but the NAT issues are relevant for UDP SIP ports
too).

Basically, some NAT routers "forget" UDP mappings after a VERY short
time (like 30 seconds). Took me a while to figure that out.


- Mike

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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Dr. Michael J. Chudobiak

Von L. wrote:

plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.

They each have NAT routers, and I have triple checked that they have
opened/forwarded the correct ports, basically 5060-3 UDP. Once they



See the "NAT Issues" section at http://www.voip-info.org/wiki/view/IAX. 
(The page is for IAX2, but the NAT issues are relevant for UDP ISP ports 
too).


Basically, some NAT routers "forget" UDP mappings after a VERY short 
time (like 30 seconds). Took me a while to figure that out.



- Mike
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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Tom Vile

FYI, when we had NAT routers at both locations setting qualify=yes
did not work.

On 6/28/06, Michiel van Baak <[EMAIL PROTECTED]> wrote:

On 12:04, Wed 28 Jun 06, Von L. wrote:
> Hello,
> ;_
> ;sip.conf
> ;_
>
> [general]
> port=5060
> bindaddr=0.0.0.0
> externip=XXX.XXX.XXX.XXX
> localnet=XXX.XXX.XXX.XXX/255.255.255.248
> canreinvite=no
> tos=reliability
> srvlookup=yes
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> nat=yes
> ignoreregexpire=yes

Show us one of the phone entries.
Basically check if the following is set there:
nat=yes
qualify=yes

The qualify=yes will send packets so the nat states stay open.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Michiel van Baak
On 12:04, Wed 28 Jun 06, Von L. wrote:
> Hello,
> ;_
> ;sip.conf
> ;_
> 
> [general]
> port=5060
> bindaddr=0.0.0.0
> externip=XXX.XXX.XXX.XXX
> localnet=XXX.XXX.XXX.XXX/255.255.255.248
> canreinvite=no
> tos=reliability
> srvlookup=yes
> disallow=all
> allow=ulaw
> dtmfmode=rfc2833
> nat=yes
> ignoreregexpire=yes

Show us one of the phone entries.
Basically check if the following is set there:
nat=yes
qualify=yes

The qualify=yes will send packets so the nat states stay open.
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Tom Vile

You have to lower the registration interval in the phones to under a
minute otherwise the NAT hole closes and no calls come in.

Polycom has said that they are going to be putting in a keep alive in
the firmware at some point.

On 6/28/06, Von L. <[EMAIL PROTECTED]> wrote:

Hello,

Here is a breakdown of the issue I am experiencing. I have three remote
employees, in various states, who have Polycom 501 phones. They are
unable to receive incoming calls after a few minutes of the phones being
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.

They each have NAT routers, and I have triple checked that they have
opened/forwarded the correct ports, basically 5060-3 UDP. Once they
plug the phone it (power and ethernet) I see on the CLI console of the
asterisk server that the phones register:

Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on
bell (pid = 3652)
nell*CLI>
Verbosity is at least 10
-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600

Here is the top part of my sip.conf

;_
;sip.conf
;_

[general]
port=5060
bindaddr=0.0.0.0
externip=XXX.XXX.XXX.XXX
localnet=XXX.XXX.XXX.XXX/255.255.255.248
canreinvite=no
tos=reliability
srvlookup=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
ignoreregexpire=yes

I know it has something to do with the NAT because if I plug my Polycom
directly into my cable modem, thus making it sit on the Internet and
have a real IP, everything works just fine.

I am curious what I am missing.

Thanks.

Von L.
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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[Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes.

2006-06-28 Thread Von L.
Hello,

Here is a breakdown of the issue I am experiencing. I have three remote
employees, in various states, who have Polycom 501 phones. They are
unable to receive incoming calls after a few minutes of the phones being
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.

They each have NAT routers, and I have triple checked that they have
opened/forwarded the correct ports, basically 5060-3 UDP. Once they
plug the phone it (power and ethernet) I see on the CLI console of the
asterisk server that the phones register:

Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=
Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running on
bell (pid = 3652)
nell*CLI>
Verbosity is at least 10
-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600

Here is the top part of my sip.conf

;_
;sip.conf
;_

[general]
port=5060
bindaddr=0.0.0.0
externip=XXX.XXX.XXX.XXX
localnet=XXX.XXX.XXX.XXX/255.255.255.248
canreinvite=no
tos=reliability
srvlookup=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
nat=yes
ignoreregexpire=yes

I know it has something to do with the NAT because if I plug my Polycom
directly into my cable modem, thus making it sit on the Internet and
have a real IP, everything works just fine.

I am curious what I am missing.

Thanks.

Von L.
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