Mar 28, 2018 at 3:09 PM, Jean Cérien
> wrote:
> >
> > Thanks for the help
> >
> > I've reproduced the issue on the test bed, with sipp to generate calls.
> >
> > The issue appears in the second call - Asterisk places a call to Kamailio
> > that s
does the
> INVITE and 200 message look like
>
>
> On Mar 28, 2018, at 9:04 AM, Jean Cérien wrote:
>
>
> Kamailio.
>
> Here is the situation. Call arrives from voip provider to kamailio, it
> dispatches to asterisk, asterisk answers, and initiates another call
> throu
vices
>
> On Mar 27, 2018, at 6:06 PM, Alberto Llamas
> wrote:
>
> Hi Jean,
>
> It might be something else. We do have an entire virtualized environment
> on Vmware with Asterisk, kamailios and another VoIP component without any
> issue with thousands of customers using i
Many thanks for this quick feedback. Is that your own hardware, or
something hosted ?
J.
On Tue, Mar 27, 2018 at 6:06 PM, Fred Posner wrote:
> On 3/27/18 5:48 PM, Jean Cérien wrote:
>
>>
>> Hello
>> We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with
Hello
We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple
of asterisk running on 2 physical hosts.
Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually
have no audio issues, but communication between the asterisk & kamailio for
sip sometime fails - I get a
Hello
I have setup 2 gateways in my dr_rules: 5,6 - so the 2nd one is tried if
the first fails.
The second one requires credentials.
When the first one fails: I execute the following code:
failure_route[ROUTEFAIL] {
if ( t_check_status("[345][0-9][0-9]") or (t_branch_timeout() and
!t_br
Hi
I've isolated to that line:
xlog("L_INFO","route(fixdigits) @@ $rm - E164 :
sip:"+$(fU{s.substr,1,0})+"@"+$fd);
(I was trying to evntually modify the from field). I guess the + werent a
good idea.
Kamailio version is the one from your repo (deb
http://deb.kamailio.org/kamailio50 stretch mai
Hi
I guess I'm a bit too creative when it comes to kamailio syntax just
edited the cfg file and got the following error upon startup:
kamctl start
INFO: Starting Kamailio :
/usr/sbin/kamctl: line 1915: 5294 Segmentation fault $OSERBIN -P
$PID_FILE -f $ETCDIR/kamailio.cfg $STARTOPTIONS
entries to the max dialog
> lifetime that you allow, as a safety backup.
>
> Cheers,
> Daniel
>
> On 22.01.18 13:59, Jean Cérien wrote:
>
>
> Hello
> Any suggestions ? still struggling with this,
>
> Rgds
>
> On Fri, Jan 19, 2018 at 3:09 PM, Jean Cérien
&g
Hello
Any suggestions ? still struggling with this,
Rgds
On Fri, Jan 19, 2018 at 3:09 PM, Jean Cérien wrote:
>
> Hello
>
> I am following up on the improper sip dialog that has been fixed by
> storing contact details into a htable.
>
> I am trying to delete the key at th
Hello
I am following up on the improper sip dialog that has been fixed by storing
contact details into a htable.
I am trying to delete the key at the end of the call, so the table doesnt
grow exponentially.
I've added a event_route[dialog:end] {} block, but I see it is processed
BEFORE the BYE i
Many thanks, I've managed to handle a call properly.
I really want to thank Daniel, and all that have helped on this subject.
Regards
J.
On Mon, Jan 8, 2018 at 11:42 PM, Joel Serrano wrote:
> Just a hint here, try setting $du and then t_relay...
>
> On Mon, Jan 8, 2018 at 11:
sip:number@asteriskip:5060";
if (!t_relay()) {
Why would the t_relay forward to the kamailio IP and not the asterisk ?
Rgds
J
On Mon, Jan 8, 2018 at 11:49 AM, Jean Cérien wrote:
> Many, many thanks !
>
> I've posted the full dialog unredacted here: https://pastebin.com/EE9i
:5060 SIP/2.0
Am I understanding correctly ?
Rgds
J
On Mon, Jan 8, 2018 at 11:28 AM, Sebastian Damm wrote:
> Hi,
>
> On Mon, Jan 8, 2018 at 2:39 PM, Jean Cérien wrote:
>
>>
>> Thanks for this answer. The voip provider is not really eager to alter
>> its SBC as it
Thanks for this answer. The voip provider is not really eager to alter its
SBC as it considers that the contact field is not mandatory in the ACK. The
RFC states (section 8.1.1.8)
The Contact header field MUST be present and contain exactly one SIP
or SIPS URI in any request that can result in
traffic look
> like? Are Asterisk and Kamailio on different Servers?
>
>
>
> -Steve
>
>
>
> *From:* sr-users [mailto:sr-users-boun...@lists.kamailio.org] *On Behalf
> Of *Jean Cérien
> *Sent:* Tuesday, January 2, 2018 8:04 AM
> *To:* Daniel-Constantin Mierla
&
Hello
Any suggestion as to why the ACK is not forwarded to the Asterisk box and
stays on the kamailio server ?
REgards
J.
On Tue, Dec 26, 2017 at 11:49 AM, Jean Cérien wrote:
>
> Daniel
>
> Many thanks for your help in this holidays season.
>
> The ACK is now going out and
Daniel
Many thanks for your help in this holidays season.
The ACK is now going out and executions follows, which is good, however,
the ack is sent to the Kamailio box itself and not actually forwarded to
the asterisk (I've attached at the end of this message the initial invite,
received ACK and
11974b4571e0562e8e731df80f48dbc504915
>
> Apparently the t_realy() stopped script execution (by returning 0) when
> e2e ACK forwarding was successful. The patch should change that to return
> true.
>
> If all works fine while testing, then I will backport.
>
> Cheers,
> Daniel
lay is not displayed.
I hate to say this, but could this be a bug ?
Rgds
J.
On Fri, Dec 22, 2017 at 1:52 AM, Sergey Safarov wrote:
> Check that your installation have one NIC with only one default route on
> host.
> If not check that "mhomed=1" is enabled.
>
> Sergey
>
>
Hello
I am using kamailio 5.0.2, on a debian 9 system.
Everything was running fine, until one of our voip provider changed his
switch. Our kamailio is relaying between several voip providers and several
asterisk (only the signalisation, no rtp).
When we get an invite from this new switch, we sele
Ok - found it !!
I've managed to get the cseq incremented following the info in this post:
https://github.com/kamailio/kamailio/issues/679
Many thanks for your help,
J.
On Tue, Sep 26, 2017 at 3:34 PM, Jean Cérien wrote:
>
> Hi
>
> Still searching on that one. My guess
= "test";
$avp(apass) = "test";
uac_auth();
t_relay();
exit;
}
On Tue, Sep 26, 2017 at 11:02 AM, Jean Cérien wrote:
>
> Thanks - I've done some progress - I've hard coded temporarily the user,
> pass & realm, and authenticati
e a 401 ??? I've tried to contact them but no
answer
J.
On Tue, Sep 26, 2017 at 10:03 AM, Daniel Tryba wrote:
> On Tue, Sep 26, 2017 at 09:36:19AM -0400, Jean Cérien wrote:
> > I've inserted the following block on the failure route:
> > if (t_check_status(&q
Thanks Daniel for your usual help, it is really appreciated !
I've inserted the following block on the failure route:
if (t_check_status("401|407")) {
xlog("L_INFO","failure_route(ROUTEFAIL) @@ call to
uac_auth()\n");
uac_auth();
t_relay();
Hello
I'm using Kamailio 5.0.1
With the UACREG module, I am registering to a remote provider. Register
goes out, 401 back, Register goes out with nonce & co, OK
Later, when I send an invite, the provider issues an 401 Unauthorized. I
guess it expects me to resubmit an INVITE with the authenticat
Hello
I am using Kamailio 5.0.2, with module drouting to do prefix based routing.
1. When I call do_routing and the prefix is not known, the r-uri remains
unchanged. Is there a way to detect that the prefix is unknown, other than
r-uri remains unchanged ?
2. I have the following parameters
modpa
m not really satisfied with this
method,
Regards,
J
On Thu, Jun 15, 2017 at 4:24 AM, Daniel Tryba wrote:
> On Wed, Jun 14, 2017 at 05:28:35PM -0400, Jean Cérien wrote:
> > I have the registration working, but when my kamailio receives an
> INVITE, I
> > am a bit lost regarding
ocs/modules/5.0.x/modules/uac.html#idp38102932
>
> Cheers,
> Daniel
>
> On 14.06.17 18:17, Jean Cérien wrote:
>
>
> Thanks, I went for the permission + dispatcher, and so far it looks good.
>
> One last thing I need the kamailio server to explicitly register
>
>- carrierroute
>
> - drouting
>
> Cheers,
> Daniel
>
> On 12.06.17 21:21, Jean Cérien wrote:
>
>
> Hello
>
> I am looking at the following configuration:
>
> 1/ voip providers, with different external sip servers, will send calls to
>
Hello
I am looking at the following configuration:
1/ voip providers, with different external sip servers, will send calls to
kamailio, that will dispatch them to asterisk servers, with load balancing
& failover. The IP of the asterisk servers are known and fixed.
2/ the asterisk servers will al
collides with the above
> statement.
>
> Try removing the class=4 and give it a try. If it works, you can set
> specific 4xx codes afterwards(if needed).
>
> ---
> Stefan
>
> On Tue, May 30, 2017 at 4:49 PM, Jean Cérien
> wrote:
>
>>
>> Hello
>>
Hello
I am working with Kamailio 5.0 dispatcher module. I am aiming at having two
or more gateways receiving the traffic with load balancing, and when one
fails, that the traffic is distributed among others. Currently, testing
with 2 gw.
I see the options message going out (and not being answered
the destination to:
>
> sip:192.168.2.20
>
> --fred
>
> On 05/24/2017 03:56 PM, Jean Cérien wrote:
> >
> > Thanks for the help.
> >
> > Adding sip is not changing anything, however, kamctl dispatcher dump
> fails:
> >
> > {
> > "jsonr
/udp_server.c:475]: udp_rcv_loop(): probing packet received from
192.168.2.200 56595
On Wed, May 24, 2017 at 3:52 PM, Daniel-Constantin Mierla wrote:
> Helo,
>
> destination has to be a full sip address, so you have to add 'sip:' in
> front of the ip addresses.
>
> C
Hello
I am working with Kamaiio 5.0 on the dispatcher, with the default
dispatcher.cfg file (
https://github.com/kamailio/kamailio/blob/master/src/modules/dispatcher/doc/dispatcher.cfg
)
I've added via mysql a gateway:
mysql> select * from dispatcher \G;
*** 1. row **
mailio-knowledgeable...
>
> I think the problem is the linphone config, i sometimes use it and have
> never seen that...
> On Wed, May 17, 2017 at 8:45 PM Jean Cérien wrote:
>
>>
>> Thanks for the help.
>> I have reverted to the default config file (https://github.co
mil <
david.villasmil.w...@gmail.com> wrote:
> There isn't an ACK received, check in kamailio side to make sure it is
> received. This is most probably a nat issue.
> On Tue, May 16, 2017 at 11:20 PM Jean Cérien
> wrote:
>
>>
>> Hello
>>
>> I am getting sta
Hello
I am getting started with Kamailio (or restarted, used it briefly years
ago), with the final objective to do load balancing.
For the time being, I am just trying to have one asterisk and one kamailio,
on the same box. I have setup a box with an asterisk 11.3, and kamailio
4.4. I've taken th
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