SS7 is ?simply? another data channel. As a 56k or 64K channel, it is not a great deal different, as far as stress goes, than a D channel on a PRI. (Only discussing stress on the system.) You do have to have two SS7 channels per the SS7 standards, for reliability. From there on, there should be little to no difference how many 64K channels of audio/modem-tones you can handle. IMO. Cary Fitch _____ From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Marcelo Pacheco Sent: Friday, November 26, 2010 12:24 PM To: asterisk-ss7 at lists.digium.com Subject: Re: Incoming calls through SS7 for data modemtransmissions - possible?? I had a quad E1 PCI setup with a PRI to the PSTN and a PRI to an AS5300, the AS5300 was exactly handling analog modem calls. Works ok, as long as the CPU is always lightly loaded. Changing one E1 from PRI to SS7 should make no diference whatsoever if you're using libss7, since libpri uses DAHDI and chan_dahdi for bridging the call. It was like 6 years ago, so I don't even have the scripts. Just saying it should work, specially with faster (newer) CPUs. Jos? Pablo M?ndez Soto wrote: Thank you Horacio and Cary. We will try receiving SS7, routing via SIP, answering on the AS5300, then looping back to itself (out PRI, in PRI ports) in order to invoke the modem termination. This way we may be able to spare the TDM cards in Asterisk and reuse the E1 ports installed in the gateway. Best regards, Jos? Pablo M?ndez <http://lh6.ggpht.com/_i2ybhYj6aPg/TOiAVvoDepI/AAAAAAAAAEs/DdN0iDAVpx0/s512/ ccvp_voice_sm.jpg> 2010/11/24 Horacio J. Pe?a <horape at compendium.com.ar> Hola! ZapRAS seems to work only with ISDN calls. "This command is not for use with analog lines; it does not provide a modem emulator." (http://www.voip-info.org/wiki/view/Asterisk+cmd+ZapRAS) You need something doing the modulation. It seems that iaxmodem is your best bet, and you'll have to make a good bunch of work on it to be able to use as you want to. If your client has the cisco gateways, I'd suggest you to keep them. They are very reliable and tested, and with MICA cards they have not a high resale value, so you'll probably end with them as paperweights unless you happen to have some stack of C549 cards to repurpose them. Saludos, H On Wed, Nov 24, 2010 at 07:58:37PM -0600, Jos? Pablo M?ndez Soto wrote: > Hello, > We are working on implementing a solution for a medium service > provider. They were previously using a Cisco AS5300 gateway with some > PRI trunks to receive modem calls, then route them out the Internet. > The Telco they were buying the trunks from, discovered this > configuration and restricted them due to legal conventions, and stated > that in order to continue doing this, they would have to talk SS7 > directly. > We are planning on solving this by placing an Asterisk server with some > TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to > the AS5300 for the dial-up to complete after authenticating against a > RADIUS server. > My questions is: can we use only Asterisk to complete/terminate the > dial-up connection, removing the AS5300 out of the picture? We would > probably need a PPP channel configuration to link the modem connection > with the Internet. > Current topology to be set-up: > Telco --> SS7 --> TE410P-AsteriskServer --> ISDN --> AS5300 --> > Internet > Ideal topology: > Telco --> SS7 --> TE410P-AsteriskServer --> Internet > Some posts talk about zapRAS being able to accomplish this, not quite > sure though > Sounds like possible: > [1]http://lists.digium.com/pipermail/asterisk-users/2004-January/026956 > .html > [2]http://lists.digium.com/pipermail/asterisk-users/2009-November/24021 > 8.html > Sounds like not possible: > [3]http://lists.digium.com/pipermail/asterisk-users/2009-November/24020 > 2.html > Thanks in advance, > Jos? Pablo M?ndez > > References > > 1. mailto:asterisk-users at lists.digium.com > 2. http://lists.digium.com/pipermail/asterisk-users/2009-November/240218.html > 3. http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html > -- > _____________________________________________________________________ > -- 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 -- Horacio J. Pe?a horape at compendium.com.ar horape at uninet.edu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20101126/7f47c852/attachment-0001.htm
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