[Asterisk-Users] Maximum retries exceeded w/SIP

[Stephen Varga]

On Sat, 2003-09-20 at 23:00, Brad Waite wrote:
> Steve,
> 
> If that's the case, why is it that I could get the first 6 seconds of the 
> demo-abouttotry message?

RTP requires two one way UDP streams.

	phone ---------> asterisk
	phone <--------- asterisk

The RTP stream can be routed from the * box to the phone, but not the
other way (unless you did what you stated below). So essentially you
have a one-way conversation.

> As it turns out, if I set up a static route for my inside network on Laptop with 
> the external interface of the firewall as the gateway, everything works fine. 
> Of course, I had to turn off my anti-spoofing rules.

I am guessing you want to have a phone somewhere else on the Internet so
this solution does not meet your requirements.

> And what's the nat=yes option supposed to do in sip.conf?

I don't know the answer to that one. I am new the *, and have already
started down the path that you are going and wanted to help so you don't
have to repeat all troubles I had.

It sounds like you more than one real IP address to work with, if that
is the case there may be a way to make it work in your setup. Let me
know.

Steve

> Brad
> 
> 
> Stephen Varga wrote:
> 
> > Unfortunetly this setup does not work, when * sends SDP info in the
> > INVITE process on how to establish the audio session *'s real IP address
> > is in the packet and the outside phone tries to connect to this IP
> > address, which of course is unreachable because of the firewall. For
> > this to work you need to move * to the firewall and the firewall's ip
> > address in the SIP.CONF file.
> > 
> > HTH,
> > Steve
> > 
> > On Sat, 2003-09-20 at 12:07, Brad Waite wrote:
> > 
> >>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
> >>
> > 
> > 
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> 
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