yes, it does, but is used for the caller part. The caller is presented
in FROM hdr as SIP address and you can add a RPID hdr to add for the
caller a second PSTN like identity.
Regards,
Bogdan
Toyima Dias wrote:
Thanks Bogdan,
But what about the Remote-Party-ID? as this headers help to link SIP
identitys to PSTN numbers?
2011/3/1 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>
Hi Toyima,
you have 2 things here:
- domain routing - this is automatically done via DNS based
forwarding
- username routing - not sure why you want to change the dialed
username, but if there is not mapping logic for (B->12345 ,
C->47896, etc) you can use aliases or dialplan module for
translating these usernames.
Regards,
Bogdan
Toyima Dias wrote:
Hello,
I'm having doubts with the following architecture:
A---PROXY_A---PROXY_B---B
A calls B with the R-URI: B@PROXY_B <mailto:B@PROXY_B
<mailto:B@PROXY_B>>
This will reach at PROXY_A; but here i need to make some
dinamic changes...
if the destiny is B@PROXY_B <mailto:B@PROXY_B
<mailto:B@PROXY_B>> i must set the R-URI with a number:
12345@PROXY_B <mailto:12345@PROXY_B <mailto:12345@PROXY_B>>
If the destiny is C@PROXY_B <mailto:C@PROXY_B
<mailto:C@PROXY_B>> i must set the R-URI like this:
47896@PROXY_B <mailto:47896@PROXY_B <mailto:47896@PROXY_B>>
Setting different routes on the script is not the best way,
bacause i won't define routes for every destination at the
same domain. Dinamyc routing should help and also avps i
guess, am i right? what should i check to accomplish this?
Using Remote-Party-ID or/and P-Asserted-Identity will also help?
Any clarification will be much appreciated!
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Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and "know-how"
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Users mailing list
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--
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 28th February 2011
OpenSIPS solutions and "know-how"
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[email protected]
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