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.

The second one is for nat. For internal calls, there is no nat issue. For external calls, the router can be sit on both inside and outside on the network. You can avoid the nat issue on SIP messages.

Richard _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

This sip ALG has some problems. It is work ok with one phone behind NAT, but not multiple phones. SIP is very complicated; probably part of reason why there is no SIP conntrack module available in POM yet.

In my experience, * is not something I'd recommend to just anyone. Unless one is willing to spend significant amount of time to enhance the code... Richard

Hi Klaus, On Thursday 13 January 2005 22:38, Klaus Darilion wrote: 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".

Hello Klaus, On Friday 14 January 2005 12:53, Klaus Darilion wrote: 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 !