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[asterisk-users] How to handle multiple IPs from one SIP carrier


 
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andersen at mwdental.com
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PostPosted: Wed May 07, 2008 2:02 pm    Post subject: [asterisk-users] How to handle multiple IPs from one SIP car Reply with quote

On my SIP carrier, I register to a proxy "sipconnect.dal0.cbeyond.net"
which ends up being 192.168.22.212 (They supply a T1 bundle)

#sip show peers
Name/username Host Dyn Nat ACL Port Status
<snip>
Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms)

Yesterday, they had a problem with their primary server and reverted
to a backup server for about 5 minutes. As chance would have it, I
received a call to one of my DIDs just before and just after the switch.
As you can see below, the first call was on their primary server and
the "Found peer" finds the Generic-8174691929 peer I have set up.

Using INVITE request as basis request -
BW124119297070508-1055880459 at bwas1-vir.atl0.cbeyond.net
Sending to 192.168.22.212 : 5060 (NAT)
Found peer 'Generic-8174691929' <<<<<<<<<<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

However, just after they changed to the backup service, I received the
call below.

Using INVITE request as basis request -
BW112003982070508-1664258428 at bwas2-vir.dal0.cbeyond.net
Sending to 192.168.25.212 : 5060 (NAT)
Found no matching peer or user for '192.168.25.212:5060' <<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

Since it was a different IP address, it found no matching peer
and failed to find a valid context to send the call to.

How should this be addressed in Asterisk to allow for such an incident?

Bill
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anthonyf at rockynet.com
Guest





PostPosted: Wed May 07, 2008 2:11 pm    Post subject: [asterisk-users] How to handle multiple IPs from one SIP car Reply with quote

andersen at mwdental.com wrote:
Quote:
On my SIP carrier, I register to a proxy "sipconnect.dal0.cbeyond.net"
which ends up being 192.168.22.212 (They supply a T1 bundle)

#sip show peers
Name/username Host Dyn Nat ACL Port Status
<snip>
Generic-8174691929/817469 192.168.22.212 N 5060 OK (41 ms)

Yesterday, they had a problem with their primary server and reverted
to a backup server for about 5 minutes. As chance would have it, I
received a call to one of my DIDs just before and just after the switch.
As you can see below, the first call was on their primary server and
the "Found peer" finds the Generic-8174691929 peer I have set up.

Using INVITE request as basis request -
BW124119297070508-1055880459 at bwas1-vir.atl0.cbeyond.net
Sending to 192.168.22.212 : 5060 (NAT)
Found peer 'Generic-8174691929' <<<<<<<<<<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

However, just after they changed to the backup service, I received the
call below.

Using INVITE request as basis request -
BW112003982070508-1664258428 at bwas2-vir.dal0.cbeyond.net
Sending to 192.168.25.212 : 5060 (NAT)
Found no matching peer or user for '192.168.25.212:5060' <<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

Since it was a different IP address, it found no matching peer
and failed to find a valid context to send the call to.

How should this be addressed in Asterisk to allow for such an incident?

Bill


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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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This is why Asterisk recommends dual registration. You reg with them for
out and the reg with you for in. Smile
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oej at edvina.net
Guest





PostPosted: Thu May 08, 2008 10:19 am    Post subject: [asterisk-users] How to handle multiple IPs from one SIP car Reply with quote

7 maj 2008 kl. 21.11 skrev Anthony Francis:

Quote:
andersen at mwdental.com wrote:
Quote:
On my SIP carrier, I register to a proxy
"sipconnect.dal0.cbeyond.net"
which ends up being 192.168.22.212 (They supply a T1 bundle)

#sip show peers
Name/username Host Dyn Nat ACL Port
Status
<snip>
Generic-8174691929/817469 192.168.22.212 N 5060 OK
(41 ms)

Yesterday, they had a problem with their primary server and reverted
to a backup server for about 5 minutes. As chance would have it, I
received a call to one of my DIDs just before and just after the
switch.
As you can see below, the first call was on their primary server and
the "Found peer" finds the Generic-8174691929 peer I have set up.

Using INVITE request as basis request -
BW124119297070508-1055880459 at bwas1-vir.atl0.cbeyond.net
Sending to 192.168.22.212 : 5060 (NAT)
Found peer 'Generic-8174691929' <<<<<<<<<<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

However, just after they changed to the backup service, I received
the
call below.

Using INVITE request as basis request -
BW112003982070508-1664258428 at bwas2-vir.dal0.cbeyond.net
Sending to 192.168.25.212 : 5060 (NAT)
Found no matching peer or user for '192.168.25.212:5060'
<<<<<<<<<<<<
Found RTP audio format 0
Found RTP audio format 100

Since it was a different IP address, it found no matching peer
and failed to find a valid context to send the call to.

How should this be addressed in Asterisk to allow for such an
incident?

Bill


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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


This is why Asterisk recommends dual registration. You reg with them
for
out and the reg with you for in. Smile


I would recommend that the service provider gives you all IP addresses
they might send calls from, not dual registration.
Sending from a different IP if you are a service provider is unusual
since so many customers have NATs that would stop all communication
anyway.

/O
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