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#470340 04/10/05 04:02 AM
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Right now it's only for their own phones. This will change in version 15 I am told because the UGW will be sip compliant and the sentinel will be built onto the UGW itself. Look for it later this summer or fall. Only works on Coral systems at the moment.


[This message has been edited by Coral Tech (edited April 10, 2005).]

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#470341 04/15/05 01:14 PM
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I think you guys might have missed the original point. The answer to the question is, there is no SIP central registry because SIP is not an address resolution protocol. SIP is only a gateway protocol that allows a client at any IP address to connect with a given server, which then can connect with the PSTN. All the protocol does is to inform the server that a client is authenticated as having a certain IP address, user name and set of capabilities. The server then acts as that client's proxy to originate and terminate calls on the PSTN.

Now what you might be referring to is the handoff, if one SIP phone calls another. According to its dialplan, the SIP server knows that a certain dialed number can be reached at a certain SIP address (which is a regular IPv4 internet address) and extension. Once the calling and called servers agree, the SIP phones are told to make their own native bridge and the server drops out of the voice path.

As other posters have said, this is in its infancy and the applications are not mature. SIP is definitely a standard, though, and can be read at https://www.ietf.org/rfc/rfc3261.txt?number=3261

#470342 04/15/05 02:44 PM
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Now we're getting somewhere. So I have a swith made by ABC and I want to call an end point in a switch made by DEF and I want it to be SIP to SIP. Can I do it now or in the future?

#470343 04/16/05 07:50 PM
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<font face="Verdana, Arial" size="2">Originally posted by MARK3906:
Now we're getting somewhere. So I have a swith made by ABC and I want to call an end point in a switch made by DEF and I want it to be SIP to SIP. Can I do it now or in the future?</font>

That's a number one priorty I was told on Thursday by a Comdial insider.

#470344 04/16/05 07:55 PM
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<font face="Verdana, Arial" size="2">Originally posted by altaphon:
I think you guys might have missed the original point. The answer to the question is, there is no SIP central registry because SIP is not an address resolution protocol. SIP is only a gateway protocol that allows a client at any IP address to connect with a given server, which then can connect with the PSTN. All the protocol does is to inform the server that a client is authenticated as having a certain IP address, user name and set of capabilities. The server then acts as that client's proxy to originate and terminate calls on the PSTN.

Now what you might be referring to is the handoff, if one SIP phone calls another. According to its dialplan, the SIP server knows that a certain dialed number can be reached at a certain SIP address (which is a regular IPv4 internet address) and extension. Once the calling and called servers agree, the SIP phones are told to make their own native bridge and the server drops out of the voice path.

As other posters have said, this is in its infancy and the applications are not mature. SIP is definitely a standard, though, and can be read at https://www.ietf.org/rfc/rfc3261.txt?number=3261
</font>

Altaphon that info is what has been needed - thanks for the excellent input!!

Harleyman

#470345 04/20/05 09:49 AM
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Spoke to Kentrox engineering today. The next release of the Q series router will allow for SIP NAT traversal without a VPN. This is what we're looking for!


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Allworx, Sangoma, FreePBX
Telephone & computer systems in the Jefferson City, Columbia MO area.
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