Check the subscribers table - this is where authentication happens. The
domain table (as I understand it) is only used to check what is a local
domain and what isn't - but this isn't what makes a registration transaction
occur.
check_source_address() compares the IP against the address table -
Is this construct in the scripting variables page and I just missed it?
$(replyrs)
I had no idea you could throw in stuff like that to the variables.
On Thu, Feb 24, 2011 at 6:13 AM, Dave Singer dave.sin...@wideideas.comwrote:
Ronald,
The only time I've seen it be null in failure route is
I would do this by matching the method of the packet in the main routing
block, and then using textops to search for the 403 in a subsequent if () {}
block, and then just drop; or exit;
My methods are usually not the most elegant solution - I'll wait for Dave or
one of the Devs to chime in with a
Just two cents -
We make such a product already - and it's very tricky to get it working. We
have real-time routing decisions based on user-provided variables such as
location and presence. We can automatically direct calls based on
inactivity on the keyboard and mouse over a period of say 1
errors on the from header as I use uac_replace_from to
normalize traffic to determine jurisdiction, etc. I'd like to know how to
identify that traffic, so, would $fu be NULL in this scenario? Is there a
better method?
Thanks.
On Sun, Feb 13, 2011 at 7:13 AM, Tyler Merritt ty...@fonality.com wrote
I'm relatively new to OpenSIPs myself - but in my various experiments - the
idea that OS cannot parse a header is a bit foreign. As a SIP proxy - it's
job is more to use the routing logic in order to transmit packets from A
through to C via B (OS itself).
I have sent packets with bogus From and
Why not use an $avp and grab the Call ID header on the inbound packet and
then create some routing logic that checks the $avp against the return
packet Call ID header to validate it's the same thing? $avps can be made
available onreply with a modparam though forgive me if it's a bit late at
night
trouble with.
Dave
On Thu, Feb 3, 2011 at 4:01 PM, Tyler Merritt ty...@fonality.com wrote:
Dave,
The audio on some of the webinars that I have watched has been almost
unintelligible :( I like webinars - I present many of them in my work for
our customers, but I couldn't really hear
-INVITE fails because UAC/UAS expect RR
on first re-INVITE too
Just a supposition
Regards,
Bogdan
Tyler Merritt wrote:
Hi,
We've got three parties for this example: A, B, C
A - Asterisk end-point Polycom
B - Asterisk end-point Polycom
C - Outside end-point Uniden
Well now I've made further changes and it appears that the carrier must be
doing some late night maintenance - because I'm getting 0 replies to my 200
OKs now - so Asterisk just continues to re-generate them endlessly.
On Wed, Feb 9, 2011 at 7:20 PM, Tyler Merritt tmerr...@fonality.com wrote
All,
A very special issue I'm facing now. I'm using is_method from the reply
block (starting with a has_totag which matches and then goes inside that
block to check stuff out).
Here is the code:
xlog (L_INFO, Right before the reINVITE check!\n);
if ( is_method(INVITE) )
it to and see what happens. Though that
will affect any place it would use the Server header. Though I think
it is only a FYI header that is not required.
Seems like a very pick carrier. :-/ Oh the hoops they sometimes put
you through.
Dave
On Mon, Feb 7, 2011 at 8:14 PM, Tyler Merritt
I'm sorry in advance that I am not posting tcpdump packets. It's 11:20 and
I'm tired.
I have a strange issue.
Scenario:
A has DID A
B has DID B
Both A and B reside on the same UAS. HOWEVER, when A calls B using DID B -
the call should be routed to the carrier (and it comes right back in -
Nvm - for whatever reason - the value is now passing properly:
/usr/sbin/opensips[9213]: P-Called-Party-Id equals XXX
I swear I changed nothing... But I'm also quite sure the computer didn't
just decide to cooperate.
On Mon, Feb 7, 2011 at 4:55 PM, Tyler Merritt ty...@fonality.com
Using Opensips 1.6 -
Can anyone tell me what I'm doing wrong here?
if (is_present_hf(P-Called-Party-Id)) {
append_hf(GW: EPSILON\r\n);
$avp(oriUri) =
$(hdr(P-Called-Party-Id){s.substr,5,12});
xlog(L_INFO, Reply user now
. Check
that out. Depends on what you are trying to accomplish.
Dave
On Mon, Feb 7, 2011 at 3:48 AM, Tyler Merritt ty...@fonality.com wrote:
This started working - but the problem is that it's matching both sides
of
the conversation. I'm searching for a concrete method to say only
putting it in because it is
not a standard SIP header and it does not start with X-.
Turning up the debug would probably show why.
Dave
On Mon, Feb 7, 2011 at 6:02 AM, Tyler Merritt ty...@fonality.com wrote:
Using Opensips 1.6 -
Can anyone tell me what I'm doing wrong here
I have used if (status==100) and if (is_method(TRYING)) and placed
these in a variety of places, at the top of route[0], in route[1], in
onreply_route (where it actually works), but nowhere can I get the OUTGOING
100 TRYING packet to change...
I can't figure this out - there isn't very good
I tried this:
remove_hf(Record-Route, Server, Supported);
And it blows up with this:
Feb 4 22:11:42 [32421] DBG:core:find_cmd_export_t: remove_hf not found
Feb 4 22:11:42 [32421] CRITICAL:core:yyerror: parse error in config file,
line 612, column 23-24: unknown command remove_hf, missing
Hi,
We've got three parties for this example: A, B, C
A - Asterisk end-point Polycom
B - Asterisk end-point Polycom
C - Outside end-point Uniden
OpenSIPs sits in front of the Asterisk servers and communicates with a
carrier C5 switch directly (same local area network inside a lab facility)
headers. What would be the issue with
listing multiple remove_hf() for each header you need to take out?
Regards,
Bogdan
Tyler Merritt wrote:
I tried this:
remove_hf(Record-Route, Server, Supported);
And it blows up with this:
Feb 4 22:11:42 [32421] DBG:core:find_cmd_export_t
the audio? Bogdan - can I send you a mic better mic :)
http://pbxtra.fonality.com/products/hud/ *Tyler Merritt*. Sales Engineer
.
Contact: tmerr...@fonality.com ty...@fonality.com | 310.861.4300 x 8850
|
fonality.com http://www.fonality.com | SE
Bloghttp://fonalityse.wordpress.com/
http
/ *Tyler Merritt*. Sales Engineer
.
Contact: tmerr...@fonality.com ty...@fonality.com | 310.861.4300 x 8850
|
fonality.com http://www.fonality.com | SE
Bloghttp://fonalityse.wordpress.com/
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