asterisk users - Dec 2005 - Why Won't Asterisk REINVITE?

We are trying to use Asterisk to set up a call between two SIP devices 
and then step out of the path.

- all systems have public IP addresses (no firewalls, no NAT).
- sip.conf has "canreinvite=yes" for both devices
- ulaw is the only permitted codec so we do not have transcoding issues 
(and a "sip show channels" confirms both legs at ulaw)

yet a SIP trace shows that Asterisk does even try to issue a reinvite.

What else should we look at to see where things are going wrong?

-- 
George Pajari, netVOICE communications    604 484 VOIP (484 8647 x102)
Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
                  www.netvoice.ca  www.ip-centrex.ca
      www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca

What does your dial command look like?
If you have Tt, wW, or hH, then asterisk will always stay in the path.

On 12/8/05, George Pajari <George.Pajari@netvoice.ca>
wrote:
> We are trying to use Asterisk to set up a call between two SIP devices
> and then step out of the path.
>
> - all systems have public IP addresses (no firewalls, no NAT).
> - sip.conf has "canreinvite=yes" for both devices
> - ulaw is the only permitted codec so we do not have transcoding issues
> (and a "sip show channels" confirms both legs at ulaw)
>
> yet a SIP trace shows that Asterisk does even try to issue a reinvite.
>
> What else should we look at to see where things are going wrong?
>
> --
> George Pajari, netVOICE communications    604 484 VOIP (484 8647 x102)
> Open Source VoIP/Telephony Specialists  1 877 NET VOIP (638 8647 x102)
>                  www.netvoice.ca  www.ip-centrex.ca
>      www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca
>
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T/t/H/h and other options to Dial require Asterisk to stay in the RTP 
stream.

George Pajari wrote:
> We are trying to use Asterisk to set up a call between two SIP devices 
> and then step out of the path.
> 
> - all systems have public IP addresses (no firewalls, no NAT).
> - sip.conf has "canreinvite=yes" for both devices
> - ulaw is the only permitted codec so we do not have transcoding issues 
> (and a "sip show channels" confirms both legs at ulaw)
> 
> yet a SIP trace shows that Asterisk does even try to issue a reinvite.
> 
> What else should we look at to see where things are going wrong?
>