Hi folks, Mehdi, what you asked is applicable in Asterisk but i think its not straght forward, as you are asking to work with signalling (No voice circuit), it need some code manipulation in SS7 and also in PRI. Im not sure if Mathew's libss7 supports signalling without voice but Im sure for libpri its something odd. So you need more understanding of Zaptel and libpri/libss7 to accomplish the above task. Mathew, don't you think so, am i right, i'd like you to correct me if im wrong. Regards. -- M. Shokuie Nia. SENA Co. On 13/12/2008, Matthew Fredrickson <creslin at digium.com> wrote: > > Mehdi Shirazi wrote: > > Hi > > I am working in a big telecom operator. > > we have huge amount of ISUP trunks but in some exchanges we don't have > > enough PRI Es . I have an ideas for solving this problem in lowest > > price as below : > > There should be a special signaling convertor like this: this converter > > receives 2 SS7 signaling channels from telecom switch and convert it to > > 120 D_channet(PRI) on same TE410 board then inside Telecom switch with > > configuring "semi permanent connection" > > every D_channel will map to channel 16 of E1s that supposed to work as > > PRIs so 120 D_channel will map to channel 16 of 120 E1s( from telecom > > switch point of view this E1s are SS7 trunks ). > > by this method with using just 4 E1 ports we can convert 120 ISUP E1 to > > PRI and sell them. > > > > signal convertor box <<--2 ss7 channel---Telecom switch -----120 > > PRI---->ISPs,PABXs > > signal convertor box --120 D_channel-->>Telecom switch > > ........''.........''............" > > > > I want to know is doing this method is possible with Asterisk+ > > libss7+libpri without modifying source cods ? > > If need modifying source code is it easy or need deep understanding of > > source codes ? > > what is your suggestion is it practical idea at all or not ? > > This is certainly doable and a not uncommon situation. I think a number > of people use libss7 for ISDN<->SS7 gatewaying. (In case you didn't get > my email earlier). > > Matthew Fredrickson > Digium, Inc. > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20081222/811f40ce/attachment.htm
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