thx for the suggestion, but anything after the fact (what I mean is once
gets into asterisk script or context), is no good.
What I am trying to accomplish is to have asterisk pick the correct context
base on the sippeers table. The only way asterisk will identify by is the
user name in the URL
thx for the help, if you like you can read my reply to Mike.
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thx for the suggestion, I don't think asterisk reads the IP from any of the
header or in any part of SIP message. I think asterisk read the IP from the
IP at the network layer. anyhow, if you like you can read my reply to Mike.
Thx again. Have you ever encounter the usr_loc module that stop
I believe the suggestion is to:
1. Allow all calls from OpenSIPs to hit your dial plan (insecure=invite)
2. In the dial plan, check for the existence of a custom header, this customer
header should be inserted by opensips to indicate the original IP
(append_hf(X-Original-IP: $si);)
3. do
check this resource
http://www.opensips.org/Documentation/Script-CoreFunctions#toc43
2013/10/14 Brett Nemeroff br...@nemeroff.com
I believe the suggestion is to:
1. Allow all calls from OpenSIPs to hit your dial plan (insecure=invite)
2. In the dial plan, check for the existence of a
Hello,
I'm use this procedure to know original IP (long way):
first define a context on opensips trunk configurated on Asterisk., then
I use this dialplan block:
[opensip]
exten = _X.,1,Set(uri=${CHANNEL(uri)})
same = n,Set(uri2=${CHANNEL(from)})
same = n,Set(uri=${CUT(uri,@,2)})
same =
sorry, again, I am not looking for to know the source ip IN asterisk
script. I need to direct the incoming call to different context within
asterisk by the source IP. I DO NOT want to have 1 big context script in
the extensions.conf. I want to be able to able to direct inbound call to
different
Yes, I know. But with that, I will end up with 1 HUGE context (script) in
the extensions.conf. So if I have 10 different customer, I would have to
have 1 HUGE context script to maintain. Vs. if I can direct differnet
Inbound calls to different context from beginning then the script is much
You do not need to manipulate core variables. You have to add a header
to pass the source ip to asterisk.
esample append_hf(X-src-ip: $si\r\n)
Il 10/10/2013 02.05, bluerain ha scritto:
Are you sure? Can you tell my which function call in opensips? I know how
to manipulate the core
cool, thx for that, I will try it! Thank you very much for your help!
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Just FYI, I tried, I insert your line in the method invite and right before
the routing, Asterisk didn't seem to care. It still care about the prior
Hop IP.
So what I mean is that
from 199.33.33.33 -- opensip 22.55.33.33 (and then I put your line) --
Asterisk server.
Asterisk server identified
the sugestion from Stefano is to you transport the ip information from
opensips to asterisk, when you are in asterisk you get that variable and
validate the customer, if just opensips will talk with asterisk so you dont
need the ip address on asterisk, just in opensips, you made all validation
on
Hi bluerain,
Our production setup is using OpenSIPS as proxy to many Asterisk instances
via load_balancer module.
I can see from the SIP logs on our Asterisk servers the SIP headers sent
from OpenSIPS to Asterisk still contains the original IP (From, Contact,
and the bottom one Via)
Have you
Just FYI
you need to write down some code in asterisk to manage the new header
obviusly
Il 11/10/2013 02.30, bluerain ha scritto:
Just FYI, I tried, I insert your line in the method invite and right before
the routing, Asterisk didn't seem to care. It still care about the prior
Hop IP.
So
I've try to search on internet but not much info. I currently have Asterisk
server setup to have sip trunk with customers on a peer type. This way,
no registration need and that asterisk server will identify the inbound call
base on IP address matching. But now I would like to put OPENSIPS in
opensips can add an header with the real IP
and asterisk can use that header to know the real IP
Il 09/10/2013 17.02, bluerain ha scritto:
I've try to search on internet but not much info. I currently have Asterisk
server setup to have sip trunk with customers on a peer type. This way,
no
Are you sure? Can you tell my which function call in opensips? I know how
to manipulate the core variable, but $si is read only. And I think if you
define a peering resource in asterisk, it will try to match it by the
source IP at the network layer and not within the INVITE. Please tell me
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