[Serusers] ser on linksys router

[Klaus Darilion]

Hi Alexander!

Alexander Hoffmann wrote:

> 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 !

Yes, I'm aware of this problem - this is IMO a bug in asterisk. This 
scenarios is not a "loop", but a "spiral" and spirals should be allowed. 
Its probably because af the bad dialog matching code in asterisk. I 
would suggest to use an ATA (SIPURA, Grandstream) instead of asterisk to 
connect an anlog phone.

regards,
klaus