[asterisk-users] reinvite

[Mark Michelson]

Jeff LaCoursiere wrote:
> On Mon, 9 Feb 2009, Jeff LaCoursiere wrote:
> 
>> I've never used "reinvite" in systems I have installed to date, and I have 
>> finally run across a situation where it would be preferred.
>>
>> A remote office has a flaky Internet connection.  With G729 encoding the 
>> calls to the central office over the 'net are tolerable.  One Linksys 2102 
>> drives two phones at this location, and when the first one calls the second 
>> one it travels to the central office and back, which is no longer tolerable.
>>
>> For each sip peer I have "canreinvite=yes", but I am a bit confused as to the 
>> correct options on the 2102 to use this feature.  Is anyone doing this with 
>> 2102s that can give me some pointers?
>>
> 
> I have been playing around with this in my "lab" and cannot seem to make 
> it work as expected.
> 
> I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5.
> 
> I have two Polycom IP501s on a local LAN behind a NAT gateway.
> 
> Both Polycom's register with the remote server and can call each other 
> without issues.
> 
> Both SIP contexts have nat=yes, canreinvite=yes.  The caller is 223, the 
> callee is 222.
> 
> eth0 is the outside (public) interface, XXX is my dynamic IP.
> 
> I trapped a conversation on the asterisk server with:
> 
> tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22
> 
> While this was running I made a call between the two extensions for a few 
> seconds then hungup.
> 
> I opened this capture in etherreal and can see the following:
> 
> 223->AST  INVITE 222 at AST
> AST->223  407 Proxy auth required
> 223->AST  ACK
> 223->AST  INVITE 222 at AST, with proxy-auth info
> AST->223  100 Trying
> AST->223  200 OK
> 223->AST  ACK
> 
> Then I see the RTP traffic begin back and forth.  I am confused on two 
> fronts - first where is the INVITE from AST to 222?  Not sure how I missed 
> capturing that side of the conversation.  And of course where is the AST 
> "reinvite"?  It isn't occurring since I can clearly see the RTP traffic 
> flowing via the asterisk server.
> 
> Any ideas?
> 
> Cheers,
> 
> j
> 

Asterisk may not be sending reinvites to the phones due to options you have 
passed to the Dial application. If Asterisk needs to intercept DTMF for a 
feature, then Asterisk will not send reinvites to the endpoints to redirect the 
media. For instance, if you have the 't' or 'T' options enabled in your Dial 
application, then Asterisk will not send reinvites to the endpoints even if you 
have configured chan_sip to allow reinvites to be sent. Other factors which can 
contribute are use of applications like Monitor and MixMonitor which require the 
media to go through Asterisk.

Mark Michelson