[Asterisk-Users] Maximum retries exceeded w/SIP
Andy Sackheim
andys at tds-inc.com
Sun Jun 20 05:09:35 MST 2004
Brian:
Thanks!
I looked through the list and didn't see a correlation between what I was seeing and those parameters. Must have missed it.
Thanks for your help.
Andy
----- Original Message -----
From: Brian K. West
To: asterisk-users at lists.digium.com
Sent: Saturday, June 19, 2004 11:49 PM
Subject: Re: [Asterisk-Users] Maximum retries exceeded w/SIP
Usage of externip= and localnet= are what you are looking for.
These all have been covered more than once in the mailing list...
Remember GOOGLE IS YOUR FRIEND!! :P
bkw
----- Original Message -----
From: Andrew Sackheim
To: asterisk-users at lists.digium.com
Sent: Saturday, June 19, 2004 9:29 PM
Subject: [Asterisk-Users] Maximum retries exceeded w/SIP
I struggled with this for several hours tonight.Turns out that if you have an * machine behind NAT, you must put the PUBLIC address in the bindaddr in sip.confIf you don't put it in, the Contact: header contains the NATted address and the sip phone can't get back to *.I don't know what happens if you mix and match sip phones on the local network -- it might not work unless the sipphone uses the public address as well. Hope this helps as I see this thread come up again and again... Andy -------------------Steve,
Sure, I could put all my machines on the public Internet, but that defeats the
purpose of having a firewall in the first place.
As an alternative, I could only place the * server on the outside, but I'd
rather not give the script-kiddies another box to pound.
Steve Totaro wrote:
> Can you disable your firewall? i am about to start this phase of asterisk
> an would like help from one newbie to another. otherwise this newbie will
> let you know how i did it.
>
>
> ----- Original Message -----
> From: "Brad Waite" <brad at wcubed.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Saturday, September 20, 2003 9:07 AM
> Subject: [Asterisk-Users] Maximum retries exceeded w/SIP
>
>
>
>>First of all, I'd like to send a big "thank you" to all the folks who have
>>helped me get this far.
>>
>>Now on to the next problem. Here's my current network setup:
>>
>>
>>The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1)
>> |
>> +--- Laptop (public IP)
>>
>>natd is set up with the following rules:
>>
>>redirect_port udp 10.0.0.253:10000-20000 10000-20000
>>redirect_port udp 10.0.0.253:5060 5060
>>
>>* is set up with the demo/sandbox config.
>>
>>I'm using XLite as my SIP client and have configured it on PC to work with
>
> *.
>
>>I'm able to do everything I've tried so far. I should, though - I'm on
>
> the inside.
>
>>However, when trying to make a call from the outside (via Laptop),
>
> something's
>
>>breaking. I've set up the SIP proxy in XLite to be the external interface
>
> on
>
>>the firewall, and am able to log into the proxy without difficulty. And
>
> while I
>
>>can begin conversations, I can't keep them going for long.
>>
>>For instance, when trying to call 500 at 10.0.0.253 (or 500 at FWpublicIP), I
>
> get most
>
>>of the "demo-abouttotry" message - "I am about to attempt an IAX
>
> connection to a
>
>>demonstration server located at Di" - at which point it gets cut off. The
>>console spits out the following error:
>>
>>File chan_sip.c, Line 443 (retrans_pkt): Maximum retries exceeded on call
>>FB9CEC48-7CE1-4171-895B-2DF048ED5D1F at 12.252.156.250 for seqno 12384
>
> (Response)
>
>>
>>Any ideas what could be going on? My first guess is the firewall, but I
>
> can't
>
>>figure out why some of the packets would get through while others
>
> apparently are
>
>>not. I'm at a loss.
>>
>>Brad Waite
>>aka HankPoacher
>>
>>_______________________________________________
>>Asterisk-Users mailing list
>>Asterisk-Users at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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