Re: More nf_conntrack_sip questions

[Patrick McHardy <kaber@trash.net>]

Philip Prindeville wrote:
> Patrick McHardy wrote:
>> Philip Prindeville wrote:
>>> I did a little investigation into my one-way voice issue, and noticed 
>>> that if I don't do voice-menus (i.e. where the Asterisk box itself 
>>> generates the first outbound INVITE, then passes-through the 2nd 
>>> INVITE once a handset picks up) then I get two-way voice (i.e. with 
>>> sending the call directly to the phone).  (In this topology, my 
>>> Asterisk box is also my firewall/NATting router...)
>>>
>>> If I enable the voice menus in the inbound dialplan, however, it can 
>>> hear the voice menus, but not the called-party when they pick up 
>>> their phone (extension).
>>>
>>> So someone (either the SIP conntrack module on the Asterisk border 
>>> firewall or else the SBC at the ILEC) is failing to look into the 2nd 
>>> INVITE (i.e. we're not rewriting it properly as it goes by, or the 
>>> SBC is failing to see it).
>>
>>
>> What module options are you using for the SIP helper and how is call
>> setup in asterisk configured (directrtpsetup, canreinvite, ...)?
>>
> 
> For the PSTN's switch:
> 
>  CanReinvite  : Yes

I vaguely recall some problem in the implementation of this feature,
something with missing bridging of the RTP streams that was still
necessary under some circumstances. Might be worth to try turning
it off.

What about the module options?