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[asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk


 
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john at millican.us
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PostPosted: Thu Jan 02, 2014 10:50 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


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nick at flhsi.com
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PostPosted: Thu Jan 02, 2014 10:52 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

Make sure you have nat=yes in your sip.conf either under globals or individual sip peer settings.

Nick Olsen
Network Operations (855) FLSPEED x106





From: "John Millican" <john@millican.us>
Sent: Thursday, January 02, 2014 10:50 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


--
_____________________________________________________________________
-- 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
john at millican.us
Guest





PostPosted: Thu Jan 02, 2014 11:07 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

top posting so as to not make thread even more confusing.

Nick,
I have nat=force_rport,comedia in sip.conf. It is my understanding that
nat=yes is deprecated?

Thanks,
JohnM


On 01/02/2014 10:51 AM, Nick Olsen wrote:
Quote:
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.

Nick Olsen
Network Operations
(855) FLSPEED x106



------------------------------------------------------------------------
*From*: "John Millican" <john@millican.us>
*Sent*: Thursday, January 02, 2014 10:50 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
*Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet ->
NAT/Firewall-> Asterisk

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


--
_____________________________________________________________________
-- 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





--
_____________________________________________________________________
-- 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

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nick at flhsi.com
Guest





PostPosted: Thu Jan 02, 2014 11:17 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

I believe you're correct. And that should be the correct setting.

However, You may want to do a packet sniff and confirm you're seeing the actual traffic as expected. Being that you see timeouts on the asterisk side. My bet is the rtp/sip traffic is going toward the device on a port it's not expecting. Or, The NAT device doesn't have a mapping for and being dropped at one of your routing devices.

Nick Olsen
Network Operations (855) FLSPEED x106





From: "John Millican" <john@millican.us>
Sent: Thursday, January 02, 2014 11:07 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -> NAT/Firewall-> Asterisk

top posting so as to not make thread even more confusing.

Nick,
I have nat=force_rport,comedia in sip.conf. It is my understanding that
nat=yes is deprecated?

Thanks,
JohnM


On 01/02/2014 10:51 AM, Nick Olsen wrote:
Quote:
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.

Nick Olsen
Network Operations
(855) FLSPEED x106



------------------------------------------------------------------------
*From*: "John Millican" <john@millican.us>
*Sent*: Thursday, January 02, 2014 10:50 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
*Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet ->
NAT/Firewall-> Asterisk

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


--
_____________________________________________________________________
-- 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





--
_____________________________________________________________________
-- 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
adamlists at plexicomm...
Guest





PostPosted: Thu Jan 02, 2014 11:32 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

top posting is superior anyway --- *ducking to avoid thrown objects*

If I recall correctly, when doing something like that with a polycom I
had to set the registration interval absurdly low, like 20 seconds or
something. I think the Polycom didn't send keepalives and that was the
workaround.


Quote:
top posting so as to not make thread even more confusing.

Nick,
I have nat=force_rport,comedia in sip.conf. It is my understanding that
nat=yes is deprecated?

Thanks,
JohnM


On 01/02/2014 10:51 AM, Nick Olsen wrote:
Quote:
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.

Nick Olsen
Network Operations
(855) FLSPEED x106



------------------------------------------------------------------------
*From*: "John Millican" <john@millican.us>
*Sent*: Thursday, January 02, 2014 10:50 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
*Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet ->
NAT/Firewall-> Asterisk

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


--
_____________________________________________________________________
-- 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






--
_____________________________________________________________________
-- 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
john at millican.us
Guest





PostPosted: Thu Jan 02, 2014 11:53 am    Post subject: [asterisk-users] Phone -> NAT/FIREWALL -> Internet -&g Reply with quote

Adam,
Thanks, I will try that this afternoon.
JohnM

On 01/02/2014 11:31 AM, Adam Moffett wrote:
Quote:
top posting is superior anyway --- *ducking to avoid thrown objects*

If I recall correctly, when doing something like that with a polycom I
had to set the registration interval absurdly low, like 20 seconds or
something. I think the Polycom didn't send keepalives and that was the
workaround.


Quote:
top posting so as to not make thread even more confusing.

Nick,
I have nat=force_rport,comedia in sip.conf. It is my understanding that
nat=yes is deprecated?

Thanks,
JohnM


On 01/02/2014 10:51 AM, Nick Olsen wrote:
Quote:
Make sure you have nat=yes in your sip.conf either under globals or
individual sip peer settings.

Nick Olsen
Network Operations
(855) FLSPEED x106



------------------------------------------------------------------------
*From*: "John Millican" <john@millican.us>
*Sent*: Thursday, January 02, 2014 10:50 AM
*To*: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
*Subject*: [asterisk-users] Phone -> NAT/FIREWALL -> Internet ->
NAT/Firewall-> Asterisk

Hello,
CentOS 6.x and Asterisk 11.x
I have an interesting, to me at least, situation. Using a Polycom
501(also tried with X-Lite). I have set up Asterisk to accept
registration from the Polycom and it registers successfully but then
withing 30 seconds on the CLI I get the message that the Polycom is
unreachable. The phone still shows that it is registered and if I try
to place a call from the phone to my Cell, my cell rings once and then
stops. I get a packet retransmission error:
WARNING[1303]: chan_sip.c:4174 retrans_pkt: Retransmission timeout
reached on transmission 689874757@192.168.0.100 for seqno 2 (Critical
Response)
Followed by:
n_sip.c:4203 retrans_pkt: Hanging up call xxxxxxxxxx@192.168.0.100 - no
reply to our critical packet
I am "assuming" that there is a problem with NAT. I have externip set
in sip.conf.
Any pointers to what I am missing?
Thanks,
JohnM


--
_____________________________________________________________________
-- 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








--
_____________________________________________________________________
-- 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
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