[asterisk-users] Pickup re-invite

[Tim St. Pierre]

I have 800 kbps in both directions reliably at the endpoint location.  When I 
was testing, there weren't any computers in the office, or any other phones.

The server has a 10 Mb ethernet connection in a datacenter, and I usually 
don't see more than 8 channels at once, so I don't think it's bandwidth.

The endpoints I have been testing on have been rock solid in all other modes 
of operation, except Pickup.

Asterisk is trying to do an external RTP bridge, as evidenced below.

How do I make it not do that.  I have already specified canreinvite=no for all 
peers.

nat=yes for all the peers except the upstream carriers.  It's also set as the 
global default.


Retransmitting #6 (NAT) to (Phone Public IP):1126:
INVITE sip:5109 at 192.168.9.100;transport=udp SIP/2.0
Via: SIP/2.0/UDP (Server IP):5060;branch=z9hG4bK5e02a020;rport
From: *88 <sip:*88 at Server Domain>;tag=as64bce3f7
To: "T & S St. Pierre" <sip:5109 at Server Domain>;tag=d8b4a9e50086b57
Contact: <sip:*88 at Server IP>
Call-ID: b952c733074986a8b5c639a4c8a2fa95 at 192.168.9.100
CSeq: 102 INVITE
User-Agent: Communicate Freely 1.4
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)			<-	Why!!!
Content-Type: application/sdp
Content-Length: 266


On Tuesday 11 December 2007 00:58, dave cantera wrote:
>  tim,
>  sounds like a problem I had with bandwidth... too many devices
> communicating on the same network connection to the internet... have you
> tcpdump'd or used a bandwidth tool to see what the usage is? nat=yes or
> nat=no?  should be yes..
>  did you change the router between upgrades?
>  just some random thoughts..
>  daveC
>
>
>
>  Tim St. Pierre wrote:
> Hello Folks.
>
> I'm wondering if anyone has any helpful hints.
>
> I recently upgraded to 1.4.11, and I'm having problems with pickup, both
> directed, and the pickup feature.
>
> My server is on the public internet, and all phones are behind a NAT
> router, somewhere else on the public internet.
>
> When a ringing phone is picked up by another phone, you have audio for a
> few seconds, then the call is dropped.
>
> The console shows "No response to our critical packet"
>
> A SIP debug of the conversation between the phone and the server shows a
> re-invite request right when the call drops.  The phone is of course using
> the internal IP address as it's contact, and it looks to me like the server
> is trying to use it.
>
> I have canreinvite=no for both the general sip.conf, as well as per-peer.
>
> I am using the whole range of Aastra Enterprise IP phones.
>
> Interestingly enough, some phones show their true IP address and port in
> the Asterisk registration database.  I believe this is where the phones
> have successfully communicated with a uPNP router, and discovered their
> public address.  These phones can successfully pickup the call.
>
> If I pipe the pickup call through the Local channel, it works.
>
> Why is asterisk still trying to re-invite even though I have explicitly
> told it not to in the config?
>
> It worked fine in 1.2
>
> Any suggestions, or requests for more information?
>
> Thanks for any help.
>
> -Tim

-- 
Tim St. Pierre

IP telephony specialist
sip://5101@communicatefreely.net
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net