[Serusers] ser on linksys router

[Alexander Hoffmann]

Hello Klaus,

On Friday 14 January 2005 12:53, Klaus Darilion wrote:
> Alexander Hoffmann wrote:
> > Hi Klaus,
> >
> > On Thursday 13 January 2005 22:38, Klaus Darilion wrote:
> >>Richard wrote:
> >>>I can think of two application which might be appealing.
> >>>
> >>>The first one is a pbx which can be deployed in a company. All internal
> >>>calls are routed through it. One can distribute the central ser server
> >>>functions into multiple smaller ser servers.
> >>
> >>I guess for PBX applications asterisk is better.
> >
> > That's interesting. I tried to build up my private home pbx with a
> > combination of SER and ASTERISK. The result: The missing call routing
> > capabilities which * describes with the nice word "hairpin" makes it
> > useless for PBX architectures. ASTERISK might be useful for something
> > especially because it supports NT mode for ISDN, but I cannot see for
> > what. Maybe you can help me a little and tell me how to overcome this big
> > "hairpin" issue. I would be very thankful because that would allow me to
> > set up a very nice PBX replacement which was almost completed before I
> > was stopped by the "hairpin".
>
> What do you mean with hairpin? AFAIK Cisco uses this term and means
> "VoIP<->VoIP" calls in their gateways. Hairpin is also used in STUN
> terminology and means that the NAT router forwards packets from inside1
> to inside2 although the packets are addressed to the external socket of
> inside2.
>
> What does hairpin in asterisk terminology mean?
>
This is my situation:
I use one ISDN card in NT mode and connect an ISDN-analog converter to it, in 
order to use analog phones. The card is controlled by Asterisk and it is 
configured to route any call established by an analog phone to the SER on the 
same machine.
There is another ISDN card in this computer used to receive incoming PSTN 
calls. This card is also controlled by asterisk and it is also configured to 
route all incoming calls to SER.
This setup works very well but there is one big issue: If you pick up the 
analog phone, the call goes through * to SER. Now if you want to place a PSTN 
call, SER forwards to * in order to reach PSTN. Asterisk then reports a loop 
which is not correct ! The reason why this does not have to be a loop is that 
the fact that an INV for the same call hits * twice does not necessarily mean 
to have a loop. In our case also the direction is important: * is configured 
to route all INV from the ISDN cards to SER but INVs coming from SER are 
terminated at the ISDN cards. Thus it is absolutely ok if there is an INV 
from ISDN card1 -> going to SER -> returning to * -> terminated at ISDN 
card2.
You can easily configure * to route calls like this, but it will complain 
about loops. If you comment out the loop detection in the source code 
(because this is not a loop here) then Asterisk will run into a deadlock.
I googled for a solution and saw people discussing similar issues and talking 
about hairpins. IMHO: What ever you call this, it stays a bug in ASTERISK. 
If you have any suggestions what to do, please let me know !

Thanks,
Alex