By "sending out the call in E.164 format" do you mean instructing the
gateway (or something other than CDRTool) to use that format? And are you
saying to set the destination id to '1305 USA-MIAMI' or to just '1305' and
the description to USA-MIAMI? I don't suppose by any chance you have access
to
ha do wrote:
> but when i make call Openser doesnt talk to FreeRadius
Define "doesn't talk."
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
___
I'm sorry, I don't understand. Do you mean within CDRTool or like on our
actual gateway which feeds info to the freeradius db?
On Thu, Jul 31, 2008 at 4:41 PM, David Villasmil <
[EMAIL PROTECTED]> wrote:
> You MUST set the dialed numbers as E.164 ( I think that's the number ),
> like +1305 for M
You MUST set the dialed numbers as E.164 ( I think that's the number ), like
+1305 for Miami, US. or else you'll never get it working...
d
On Thu, Jul 31, 2008 at 11:47 AM, Brian Del Shasta <[EMAIL PROTECTED]
> wrote:
> Ok, I changed the time zone to America/Chicago. If anyone else has any
> lu
Hello everybody.
A patch has been published on tracker, which fixes a problem related to not
define a macro which makes fail the module compilation in Solaris.
It has been registered with Request ID: 2034302*
*
Hope it be useful.
Best regards.
Sergio Gutiérrez.*
*
__
Ok, I changed the time zone to America/Chicago. If anyone else has any luck
setting up ratings within CDRTool, I would love to speak with you, thanks...
On Thu, Jul 31, 2008 at 10:18 AM, Adrian Georgescu <[EMAIL PROTECTED]>wrote:
> Central Time is not s valid timezone
> Further than this your ra
On 07/31/08 18:25, Darren Sessions wrote:
> Off technical topic, I know, but has the Kamailio logo already been
> designed?
>
no -- I hope community will have an important role in proposing some.
Most of the people involved in administration focus to migrate the
existing stuff (mainly web, lis
Off technical topic, I know, but has the Kamailio logo already been
designed?
- Darren
_
[EMAIL PROTECTED]
http://www.darrensessions.com
http://www.linkedin.com/in/dsessions
_
___
Users mai
Central Time is not s valid timezone
Further than this your rates look ok at first glance, if it does not
work maybe other can help with their experiences.
Adrian
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Sorry, I forgot to show my profile in the previous post, here it is:
Ops,Trusted peer,Domain,Subscriber,Profile Id,Rate Id1,00-H1,Rate
Id2,H1-H2,Rate Id3,H2-H3,Rate Id4,H3-24
21,1,24,1,0,1,0,1,0
On Thu, Jul 31, 2008 at 9:52 AM, Adrian Georgescu <[EMAIL PROTECTED]>wrote:
> You need to add a
You need to add at list one customer entry, associated profiles <
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Okay, here's what I currently have:
Destinations
Ops,Trusted peer,Domain,Subscriber,Destination Id,Description
21888,Unbillable
21866,Unbillable
21877,Unbillable
21800,Unbillable
21,Billable
Customers
Ops,Trusted
peer,Domain,Subscriber,WeekDay,Fallback,WeekEnd,Fallback,Timezo
Yes
You need to add at list one customer entry, associated profiles and
finally the rates linked with the profile which contain the actual
prices for each destination.
You have example rating files that you can modify.
Adrian
On Jul 31, 2008, at 3:28 PM, Brian Del Shasta wrote:
That's n
I am afraid that rating all calls the same regardless of destination
will not work unless you define some destination to match all your
traffic.
The rating engine must find a destination prefix from which it derives
the price. Read the RATING.txt to understand the logic behind it.
Adrian
>>
Hi Neill,
The outgoing audio works fine. The audio that is not passed is from PSTN ->
Phone.
The Snom does have symmetrical RTP set. I did have this working about a week
ago then it stopped working!
How does asterisk deal with NAT, as if I connect the phone directly to Asterisk
everything
Which direction is the one way audio in - from the Phone to the PSTN or
Vice-Versa - I assume audio outbound is OK and it's the audio returning to
the phone from the PSTN you're missing?
Do you have symmetrical RTP set in the Line settings (under NAT) on the
SNOM?
Is Kamailio inside or out
I am having problems with a NAT device. I have an asterisk server that is
hosted on a public IP and when I connect my Snom phones to it directly audio
passes correctly both ways. When I introduce Kamailio 1.3 into the mix I get
one way audio.
I have fixed the SDP and contacts etc and everythin
Jason,
by bad over here. I was with my brains still in vacation :(.
You need to use nathelper module before saving into the database:
modparam("nathelper", "received_avp", "$(avp(i:31))")
modparam("registrar", "received_avp", "$(avp(i:31))").
The rport one is for the case when you proxy the req
Thanks DanB
ill give that a try. I got it all to work by forcing the Snoms to register
with 5060, but was more concerned about the possibility of this being a bug
(hence the post to devel list above). If I do an openserctl ul show. I can
see the full contact field with the correct port (2049). Sur
Hi Jason,
Try forcing rport before saving user location:
http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#force_rport
Cheers,
DanB
On Thu, Jul 31, 2008 at 12:19 PM, Jason Penton
<[EMAIL PROTECTED]>wrote:
> Hi *
>
> not sure if someone has encountered this before but I have noticed
Hi *
not sure if someone has encountered this before but I have noticed a problem
with Snom phones registering with Kamailio.
It seems that the Snom phones register with a contact port of 2049.
However, when Kamailio does the usrloc lookup and the corresponding t_relay,
a packet dump shows me th
Hi Henning and Daniel
Indeed, the command you gave me helped. As I interpret it, the update
pua was the last transaction, that does not "free" the last transaction.
Acc. MySQL tutorial "[...] This can happen, for example, if you are
using mysql_use_result() and try to execute a new query before yo
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