Hi All,
Is my setup possible? Or maybe the right question is "Is this correct?"
1 test server: installed ser and asterisk(didn't really understand much of it yet). 1 CISCO 1750: with 2 FXO. 2 UA's: X-Lite
Using SER only, I can make calls between extensions.
Using Asterisk alone, can I also call between extensions? And is an extension on asterisk different on the exetensions at ser? How can I make an extension in SER also an extension in asterisk?
Is it possible to route PSTN calls to the CISCO 1750 via asterisk? How about via ser? If both are possible, which should I use then?
I've read somewhere that combining SER with Asterisk, or vice versa, will be very good, how come?
Regards, Ron
You can do all of with with each one, and with both at the same time (.e., register your UAs through ser to in term rute to Asterisk.
Ser is, simply put. a (stateless) SIP Proxy. It's like a SIP layer-level router. Which is good if you need something that can scale a lot. Asterisk is a full-featured B2BUA, with a different logic and many applications that go beyond SIP routing. Asterisk can also interconnect dissimilar technologies like PSTN, SIP, H.323, MGCP, SCCP, IAX, but you won't get the came CPS (calls per second) as with Ser. If you have less than a few thousand phones, maybe Asterisk is all you need.
On Oct 16, 2004, at 4:47 AM, Ron Ramos wrote:
Hi All,
Is my setup possible? Or maybe the right question is "Is this correct?"
1 test server: installed ser and asterisk(didn't really understand much of it yet). 1 CISCO 1750: with 2 FXO. 2 UA's: X-Lite
Using SER only, I can make calls between extensions.
Using Asterisk alone, can I also call between extensions? And is an extension on asterisk different on the exetensions at ser? How can I make an extension in SER also an extension in asterisk?
Is it possible to route PSTN calls to the CISCO 1750 via asterisk? How about via ser? If both are possible, which should I use then?
I've read somewhere that combining SER with Asterisk, or vice versa, will be very good, how come?
Regards, Ron
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Hi All,
I would like to manipulate originating sip url when call destination is a pstn link (cisco voip gw eg.)
CLI is very important becasuse of accounting, so I can use valid CLI. That means eg. caller user like imedve@domain must have to change 9998877@domain before I send INVITE to a cisco gw.
I can find some rewrite functions which manipulate request URI.
Is there any way to rewrite From: HF based on an sql lookup ?
thanks, imedve
Hi,
--- Medve Istvan imedve@ew.hu wrote:
Is there any way to rewrite From: HF based on an sql lookup ?
Rewriting From header is a violation of RFC, it is not suggested. Try adding Remote-Party-ID header to the INVITE message, that should do it.
thanks, imedve
Regards,
===== Girish Gopinath gr_sh2003@yahoo.com
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