Ahh, I thought Asterisk is in the public internet, but actually you
connect to a provider (vendor), which seems to run Kamailio as well.
Using information from 100 trying is too late, as the INVITE was already
sent... so one more question before trying to propose a solution. Do you
have to registe
Hello,
this looks like an issue on the SQL query syntax. What version of mysql
server do you run there?
Cheers,
Daniel
On 14/01/16 22:38, Spencer Thomason wrote:
> Hello,
> I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing
> some strange date related errors:
> ERROR: d
I think I couldn't explain myself clearly.
I want to see the following media lines in resulting SDP:
m=audio 10076 RTP/AVP 97 98
c=IN IP4 rtpproxy1.example.com
a=rtpmap:97 speex/8000
a=fmtp:97 vbr=on
a=rtpmap:98 telephone-event/8000
m=video 20007 RTP/AVP 96
c=IN IP4 rtpproxy2.example.com
a=rtpmap:
Hi,
We're running a system with Kamailio running in front of Asterisk just
handling registrations and forwarding everything else to Asterisk. But
we're having an issue during hangup on incoming calls. If the initiator
hangs up, the call completes successfully. But if one of our phones hangs
up, th
I am stumped as my DID provider can't route via SIP URI to PBXes but they can
to IPTel. I can call my own PBX, and from my own PBX to the PBXes URI without
issue. When I route the DID via URI to IPTel and set voicebox it works and
routes to the voicebox. When I add a forward for the PBXes URI it
Hello,
I’m attempting to deploy a homer sip capture server on SPARC and I’m seeing
some strange date related errors:
ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on
query: Incorrect datetime value: '2052-06-12 16:21:06' for column 'date' at row
1 (1292)
Is there some
Do you want to change/update the SDP once the call is established ?
shouldn't it just work like this:
if(has_totag() && is_method("INVITE")) {
if(search_body("video") {
set_rtpproxy_set("1");
unforce_rtpproxy();
set_rtpproxy_set("2");
offer_rtpproxy("$avp(myflags)");
Hi,
My CSCF environment has an application server acting as a B2BUA. It's
originating a new session, and creating a new terminating session back to
the S-CSCF (two separate sessions per-call). When I perform a re-INVITE
with a diversion to transfer the call to another URI it is all good until
th
Hi Daniel,
I haven’t yet but I will try to test shortly.
Here is the value of the block variable:
(gdb) p /s block
$1 = (unsigned char *) 0x76e5ded4 “"
Thanks,
Spencer
On Jan 13, 2016, at 11:07 PM, Daniel-Constantin Mierla
mailto:mico...@gmail.com>> wrote:
Hello,
hmm, they are multip
Yes it is possible, but is there an easy way to workaround the issue using
Kamailio.
Because I have the port because vendor is sending that info in Trying:
2016/01/13 20:10:15.842055 VENDOR-IP:5060 -> PRIVATE-IP-KAMAILIO:5060
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP PUB
Not really up to date with all Asterisk features -- do you know if you
can append a custom header to a SIP response that is going to be
generated by Asterisk? Eventually the reply for an OPTIONS request.
Cheers,
Daniel
On 14/01/16 17:19, Nelson Migliaro wrote:
> Yes, I manage all devices, even th
Yes, I manage all devices, even the internet router but it does not allow
static pat.
2016-01-14 16:07 GMT+01:00 Daniel-Constantin Mierla :
> Do you control the Asterisk? If yes, depending on Asterisk capabilities of
> building replies, you may be able to do some automation to detect the
> extern
Hi Sammy,
I'm not sure it will work.
Assume the following scenario:
Kamailio receives INVITE with audio only SDP and selects rtpproxy-1.
After a while video is enabled with REINVITE.
Now SDP has video and kamailio selects rtpproxy-2 according to your
solution.
In this case, most probably video and
Do you control the Asterisk? If yes, depending on Asterisk capabilities of
building replies, you may be able to do some automation to detect the
external port.
Cheers,
Daniel
On Thu, Jan 14, 2016 at 3:47 PM, Nelson Migliaro
wrote:
> There is not a public Kamailio, only one Kamailio behind NAT,
There is not a public Kamailio, only one Kamailio behind NAT,
Right now the configuration is:
Asterisk <-> Kamailio (Private IP + advertise public IP + RTP Proxy ) <->
Internet router (public IP + symmetric na) <-> Internet
Regards,
2016-01-14 15:43 GMT+01:00 Daniel-Constantin Mierla :
> Is t
Is the kamailio behind nat communicating with another kamailio on a public
IP?
Cheers,
DAniel
On Thu, Jan 14, 2016 at 1:33 PM, Nelson Migliaro
wrote:
> Thank you Daniel for your answer,
>
> As you mention, there is a symmetric nat and router does not allow a
> static NAT.
>
> By sniffing traffi
Hi,
Yes thats possible as I think you can do a search in SDP body for "video"
if found then select the rtpproxy instance else select other one.
Regards,
Sammy
On Jan 14, 2016 06:39, "Koray Vatansever"
wrote:
> Hi everybody,
>
> Is there a way to use different rtpproxies for different media types
Thank you Daniel for your answer,
As you mention, there is a symmetric nat and router does not allow a static
NAT.
By sniffing traffic I can see the port is using new but in case it change,
how can automate the process of advertising the correct port?
Cheers!
-- Forwarded message -
Hi everybody,
Is there a way to use different rtpproxies for different media types?
I want to use one rtpproxy set for audio, and use another one for video.
Is this possible in kamailio?
Regards,
Koray
___
SIP Express Router (SER) and Kamailio (OpenSER
>
> So if I set ds_probing_mode to 0 ("only the gateways with state PROBING
> are tested"), do I have to set state PROBING manually at startup for all
> gateways? Or are gateways set to probing by default?
in ds_probing_mode=0 not any PROBING by default. if you just want to test
your node after s
Hi Dmitri,
Am 13.01.2016 um 16:08 schrieb Dmitri Savolainen:
> 1. Have you set modparam("dispatcher", "ds_probing_mode", 1)? If so, it
> will be moved to PROBING mode after ds_ping_interval
Yes, I have it set to 1.
So if I set ds_probing_mode to 0 ("only the gateways with state PROBING
are tes
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