> So the typical requirement I see is that I'll put in a handful of termination
> gateways for a given carrier, but the carriers usually (sometimes?) ask that
> if the call fails on one of the gateways (like a 503) to NOT send it to all
> the other gateways as well. In fact a "next_carrier" fun
Hi Brett,
On 04/02/2012 07:51 PM, Brett Nemeroff wrote:
Bogdan,
So the typical requirement I see is that I'll put in a handful of
termination gateways for a given carrier, but the carriers usually
(sometimes?) ask that if the call fails on one of the gateways (like a
503) to NOT send it to a
In ngrep traffic check no active rdp-session-id
but do not know how to solve
#
U +3.135110 IP-ASTERISK:5060 -> IP_OPENSIPS:5060
INVITE sip:100@ IP_OPENSIPS SIP/2.0
Via: SIP/2.0/UDP IP-ASTERISK:5060;branch=z9hG4bK3e684698;rport
Max-Forwards: 70
Well, you know, one is what we want to do , another we actually get.
I was rather asking if, making a sip capture (with ngrep) you see in
your call the RTPproxy insertion - check it in traffic, not in script.
Regards,
Bogdan
On 04/02/2012 10:05 PM, magnusadil...@gmail.com wrote:
hi, yes, rtp
On Mon, Apr 2, 2012 at 1:34 PM, Ali Pey wrote:
>
> How can I limit the number of calls sent to a gateway to its limit?
>
>
I've used dialog profiles to do this. Works well.
-Brett
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hi, yes, rtpproxy is active in invite 200
onreply_route[3] {
if ((isflagset(5) || isbflagset(0)) && status =~
"(183)|(2[0-9][0-9])" && has_body("application/sdp")) {
if (rtpproxy_answer()) {
log("L_INFO: rtpproxy_answer NAT");
Hi There,
How can I enforce a maximum number of channels for a gw or a carrier with
the DR module?
Let's say I have two carriers and each with two gateways. Each gateway can
only handle 150 channels/Calls.
How can I limit the number of calls sent to a gateway to its limit?
Regards,
Ali
Thanks Tijmen. Is there any documentation on the Radius configuration
needed?
On Mon, Apr 2, 2012 at 2:39 PM, Tijmen de Mes wrote:
> Hi,
>
>
> On 04/02/2012 02:05 PM, Remco . wrote:
>
>> Is there a way to log incoming calls using CDRTool?
>> (e.g. PSTN -> OpenSIPS -> User).
>>
>
> Yes, it will di
Make sure you have different parkinglots so one server is 700 to 750, the other
751 to 800 etc.
Either fork the call to all servers or write a config so you can route based on
number.
/O
2 apr 2012 kl. 18:36 skrev Bogdan-Andrei Iancu:
> Hi,
>
> But this park location - is it unique on your s
Bogdan,
So the typical requirement I see is that I'll put in a handful of
termination gateways for a given carrier, but the carriers usually
(sometimes?) ask that if the call fails on one of the gateways (like a 503)
to NOT send it to all the other gateways as well. In fact a "next_carrier"
functi
Hi Magnus,
attaching cfg files is useless, as no one will debug the script, but we
will help you to debug your script.
So, for the non-working case (PSTN to SIP) does your script force
RTPproxy in INVITE and 200 OK ?
Regards,
Bogdan
On 03/29/2012 01:52 AM, magnusadil...@gmail.com wrote:
I
Hi,
Is your phone actually registered with opensips ? if so, you can try
call it from another SIP phone.
Regards,
Bogdan
On 03/29/2012 12:10 PM, Petrut Dogaru wrote:
Hello,
I have downloaded the VM with opensips configured. I've copied the
opensips.cfg file and changed the paths to resemb
Hi,
SIP-T is transparent for a SIP proxy - only the SIP end points need to
be aware of the SIP-T extensions.
Regards,
Bogdan
On 03/30/2012 12:50 PM, [Digital^Dude] ® wrote:
Does Opensips have SIP-T support?
SIP-T: http://www.ietf.org/rfc/rfc3372.txt
Or does it plan to? If yes, whats the tim
After doing an "opensipsctl start", check if opensips actually started
or not:
ps auxw | grep opensips
Regards,
Bogdan
On 03/30/2012 07:00 PM, prasad kelkar wrote:
hello,
I used following paper to install opensips 1.4.4
I am getting folllowing problem in opensips1.4.4
opensipsctl start
I
But once again, the LB CAN do failover - if the call fails, in failure
route, simply do again LB to fail to another destination.
Regards,
Bogdan
On 03/29/2012 04:08 PM, [Digital^Dude] ® wrote:
Yes, dispatcher does both, but it doesn't have the LB logic... it
randomly forwards calls.
Is it poss
Hi,
But this park location - is it unique on your system ? I mean all your
Asterisk boxes do transfer all the parked calls to the same park location ?
And once the park is done, is the original Asterisk still involved in call?
I'm asking as I'm not so familiar on how the parking is done with A
Hi Miha,
On 03/29/2012 04:39 PM, Miha wrote:
Hi @Bogdan,
calls and registrantions. I will also ask for one explanation this.
well, take care and do not route the REGISTERs via LB, as LB module
works only for calls.
Now FS is connected directly to SBC. I am having some trouble finding
out ho
Hi Miha,
Well, in your script, when dealing with the initial requests, just look
at the source IP of the INVITEs - if from SBC, do the lb stuff,
otherwise route it back to SBC.
if (src_ip==11.22.33.44) {
# do LB
} else {
# send to SBC
}
Regards,
Bogdan
On 04/02/2
Hi Brett,
On 03/30/2012 09:46 PM, Brett Nemeroff wrote:
Hello List,
Upon looking at the enhancements of drouting in 1.8, I'm a bit
confused at all the new options for the gwlist fields. I found this
snippet on the list:
gateways : g1, g2, g3, g4, g5
carriers : c1 = g1=
Hi I have a Opensips server that handles registration and
loadbalancing, it load balances a few asterisk servers, I was having
issues with transfers that if each leg is on a different server it
would hangup, so Bogdan suggested I use dialogs and when opensips sees
a call on a extension, next time i
Thanks Vlad.
I'm using Dialog's CDR auto generation:
modparam("acc", "cdr_flag", 3)
The local_route acc_db_request insert 2 lines into DB - one for
"downstream" and another for "upstream" , however the dialog specific AVPs
usually inserted into the DB is lacking, and more importantly, the durati
Hi,
On 04/02/2012 02:05 PM, Remco . wrote:
Is there a way to log incoming calls using CDRTool?
(e.g. PSTN -> OpenSIPS -> User).
Yes, it will display any call if it is correctly setup. It is depending
on radius. If OpenSIPS is configured to write the accounting records to
the radius server, C
Is there a way to log incoming calls using CDRTool?
(e.g. PSTN -> OpenSIPS -> User).
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Hello,
If you want in-dialog pinging or BYE timeout to be used with
topology_hiding(), you can easily just create the dialog before, and
then apply topology_hiding() on top of the existing dialog, something like :
if (is_method("INVITE")) {
create_dialog("PpB");
Hi,
For example you could store some meaning-full information in the
dr_gateways attrs field, which will get populated in the gw_attrs_avp (
see [1] ) when you do do_routing() or use_next_gw() . You could use the
info in that AVP to store the profiles, like
set_dlg_profile("gateways","$av
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