Douglas Garstang writes:
> Right minimising cost is priority #1... and in this particular
> case the cheapest route is NOT used.
lcr module was not designed to minimize monetary cost nor for sip
trunking between carriers. its design goal was much more modest: allow
an operator who offers v
Douglas Garstang writes:
> What do people do in this case?
lcr module has possibility to ignore prefix length and use regexp
instead. that is not without its own problems though.
-- juha
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Hi,
I have media proxy, openser, and monit installed in FreeBSD.
I have also had the x-lite windows client software working with the openser
server.
Now, if I want to make phone call from home to external phone number thru the
Openser server, what else I have to do to accomplish that?
Thanks
Sa
Hi Bogdan,
Thanks for explaining :)
Rgds,
Bai Shi
-Original Message-
From: Bogdan-Andrei Iancu [mailto:[EMAIL PROTECTED]
Sent: 2008年3月26日 21:54
To: Bai Shi
Cc: openser users; Andreas Granig
Subject: Re: UAC_REPLACE_FROM()
Hi Bai,
Bai Shi wrote:
> Hi all,
>
> I would like to share some e
Right minimising cost is priority #1... and in this particular case the
cheapest route is NOT used.
- Original Message
From: Andreas Sikkema <[EMAIL PROTECTED]>
To: users@lists.openser.org
Sent: Thursday, March 27, 2008 5:20:08 PM
Subject: Re: [OpenSER-Users] LCR Routing Choosing
On Wed, March 26, 2008 6:20 pm, Douglas Garstang wrote:
> Ok... I can do it if I use cr_user_rewrite_uri("sip:[EMAIL PROTECTED]",
> "1") in openser.cfg and then change the username from blank to '.*' (WHERE
> is that documented?? I only worked it out by tracing port 3306!!!).
>
> But... this isn't
On Mar 28, 2008, at 12:38 AM, Douglas Garstang wrote:
> However, when you get routes from multiple carriers, they don't
> always bill on the same prefix boundaries. In my example above, LCR/
> Carierroute would match against Carrier1 for anything starting with
> 1650. HOWEVER, since 165 is als
This isn't a question specific to LCR module, but it could be an LCR or
Carrierroute module issue.
Let's say you have to carriers...
Carrier PrefixCost
---
Carrier116500.3
Carrier21650.2
When LCR or Carrierroute try to match a dialed numbe
Hi Raul
Do you experience with iptables SIP module? Is it broken as well? I have been
curious but have not got around to testing it.
Thanks Rob
On Thursday 27 March 2008, Raúl Alexis Betancor Santana wrote:
> El Thursday 27 March 2008 17:13:38 Mike Fedyk escribió:
> > Hi,
> >
> > Some of our cus
El Thursday 27 March 2008 17:13:38 Mike Fedyk escribió:
> Hi,
>
> Some of our customers have been connecting behind NAT routers that mangle
> the sip headers but don't keep the ports open. I've added checks on the
> source port to catch this case. Has anyone done something similar or
> better? I
Hi,
I got the solution you recommended working, Daniel. I'll look into
cfgutils when I get some free time. Here is a snippet for doing range
checking in-script for anyone who is interested. This checks if the IP
lies in 172.20.62.0 / 24
---
# Calculate decimal representation of our test range
Hi,
Some of our customers have been connecting behind NAT routers that mangle
the sip headers but don't keep the ports open. I've added checks on the
source port to catch this case. Has anyone done something similar or
better? It'd be nice if this could be added as a flag to nat_uac_test() if
i
Parallel posting this to both CDRTool users and OpenSER users, so apologies
to members of both.
I have installed the latest version of CDRTool and running it with OpenSER
1.1.1 for the time being. When looking at the tcpdump of the records
leaving the OpenSER server via radiusclient-ng I am viewi
El Thursday 27 March 2008 16:24:45 [EMAIL PROTECTED] escribió:
> Hi Inaki,
>
> Thanks for your last help. I have found that I didn't add the rule for
> INVITE in my sip.xml. Beginners mistakes :-).
> Now when trying to make a call I get no route exception:
> StandardProxy[SipProcessor[4]] -
> j
Hi Inaki,
Thanks for your last help. I have found that I didn't add the rule for
INVITE in my sip.xml. Beginners mistakes :-).
Now when trying to make a call I get no route exception:
StandardProxy[SipProcessor[4]] -
javax.sip.TransactionUnavailableException: no route!
Can you please help
Hello,
On 03/27/08 14:25, VoIP Forums www.Go4Calls.com wrote:
> Hi All,
>
> How i can get the length of called number. i am trying to making
> routing logic base on length. Please help us how i can get the length
> i tried the fillowing but did not work.
>
> if (uri:len<=7) {
> r
Hi All,
How i can get the length of called number. i am trying to making routing logic
base on length. Please help us how i can get the length i tried the fillowing
but did not work.
if (uri:len<=7) {
rewritehostport("192.168.1.1:5060");
route(1);
Hi All,
I have noticed something strange in the version I am running, cannot find
the branch/transaction (tried them both) flags in the BYE messages (one app:
end_media_session if previously used). Is that a known issue? In the INVITE
and 200 OK the flags are there.
My version:
openser -V
version
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