Hello,
Using 4.0.4:
I'm getting the following error (removed the values part of the query):
3(2931) ERROR: db_unixodbc [dbase.c:133]: db_unixodbc_submit_query(): rv=-1.
Query= insert into Dialogs
(HashEntry,HashID,CallID,FromURI,FromTag,ToURI,ToTag,CallerSock,CalleeSock,StartTime,State,Timeout
Carsten you're the man!
it was a simple case of add_contact_alias(); vs fix_nated_contact();
thanks for the insights guys
Kelvin Chua
On Fri, Nov 29, 2013 at 12:24 AM, Carsten Bock wrote:
> Hi,
>
> i see. However in the request from Twinkle, there is something else wrong:
> For subsequent re
Hi,
1. After adding t_newtran() as shown below, some how this got fixed i.e.
ACK got routed.
However, we are disputed in considering t_newtran() to have caused the fix.
if (is_method("ACK")) {
if (t_check_trans()) {
t_relay
Hi,
i see. However in the request from Twinkle, there is something else wrong:
For subsequent requests, the User-Agents should send it to the URI in
the Contact-Header.
It is done correctly for the ACK
Contact in 200 OK: Contact: .
URI in ACK: ACK sip:kelvin@1X5.6X.1X6.2X3:62280 SIP/2.0
But it i
just to illustrate further, now i used twinkle to be on the termination
side and generate the BYE
(notice twinkle does not send a routeset)
U 2013/11/28 10:53:14.919998 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060
INVITE sip:+18X88X441X1@2X8.1X1.3X.2X SIP/2.0.
Record-Route: .
Record-Route: .
Accept: ap
1. nope, i only have 1 instance of record_route in my config:
if (is_method("INVITE|SUBSCRIBE")) record_route();
2. i already have:
modparam("rr", "enable_full_lr", 1)
Kelvin Chua
On Fri, Nov 29, 2013 at 12:03 AM, Carsten Bock wrote:
> Hi Kelvin,
>
> 1) it seems you do record_route() for I
Hi Kelvin,
1) it seems you do record_route() for INVITE and ACK? If you do it for
the ACK as well, why? Doing record_route() should only be necessary
for INVITE (& SUBSCRIBE).
2) You could also test, if
modparam("rr", "enable_full_lr", 1)
would help. In this case, the first Record-Route-Header s
i am using only record_route() and my setup is dialog-aware. (dialog)
it's worth noting that i am doing serial fork, but this particular call is
the first branch.
On Nov 28, 2013 11:38 PM, "Carsten Bock" wrote:
> Hi Olle,
>
> actually it's an extension, i wanted to do sooner or later. If
> Kamai
Hi Olle,
actually it's an extension, i wanted to do sooner or later. If
Kamailio is Dialog-Aware (e.g. dialog or dialog_ng is loaded),
Kamailio "knows" what the route-set should be and could check and
optionally enforce the route-set.
By the way: In the Record-Route of the "200 OK" it's "lr" and
hi olle,
if it is indeed the case, then we would all be at the mercy of buggy
clients (or buggy carriers)
On Nov 28, 2013 11:25 PM, "Olle E. Johansson" wrote:
>
> On 28 Nov 2013, at 16:23, Alex Balashov wrote:
>
> > On 11/28/2013 10:22 AM, Kelvin Chua wrote:
> >> hi alex,
> >> is there a way fo
On 28 Nov 2013, at 16:23, Alex Balashov wrote:
> On 11/28/2013 10:22 AM, Kelvin Chua wrote:
>> hi alex,
>> is there a way for kamailio to force lr=on before relay? this carrier is
>> driving me nuts
>
> No, Kamailio (as a SIP proxy) cannot modify the Route set after the dialog is
> set up, in
On 11/28/2013 10:22 AM, Kelvin Chua wrote:
hi alex,
is there a way for kamailio to force lr=on before relay? this carrier is
driving me nuts
No, Kamailio (as a SIP proxy) cannot modify the Route set after the
dialog is set up, in a sequential (in-dialog request) furthermore.
--
Alex Balashov
hi alex,
is there a way for kamailio to force lr=on before relay? this carrier is
driving me nuts
On Nov 28, 2013 11:17 PM, "Alex Balashov" wrote:
> Kelvin,
>
> As BYEs are not generated by Kamailio, only relayed, the setting does not
> affect them. That setting only affects initial Route set pl
Kelvin,
As BYEs are not generated by Kamailio, only relayed, the setting does
not affect them. That setting only affects initial Route set placement
in the initial INVITE to which the RR header is added.
-- Alex
On 11/28/2013 10:14 AM, Kelvin Chua wrote:
hi carsten,
thanks for the insight
Hi Kelvin,
the BYE is generated by the client (CSipSimple_m0-16/r2330) and
Kamailio has no impact on the generated Route-Headers of the client.
Kind regards,
Carsten
2013/11/28 Kelvin Chua :
> hi carsten,
> thanks for the insight, that's possible.
> enable_full_lr is already set. are BYEs unaffe
hi carsten,
thanks for the insight, that's possible.
enable_full_lr is already set. are BYEs unaffected by that setting?
On Nov 28, 2013 10:44 PM, "Carsten Bock" wrote:
> Probably the other Server is validating the Route headers against the
> last set of Record-Route headers:
> In ACK:
> Record-R
Probably the other Server is validating the Route headers against the
last set of Record-Route headers:
In ACK:
Record-Route: .
Record-Route: .
In the BYE:
Route: .
Route: .
Probably their server does not like, that first you have "lr=on" and
later it's just "lr".
Kind regards,
Carsten
2013/11/
U 2013/11/28 09:06:09.411796 6X.2X1.4.1X5:5060 -> 2X8.1X1.3X.2X:5060
INVITE sip:+18X88X441X1@2X8.1X1.3X.2X SIP/2.0.
Record-Route: .
Record-Route: .
Accept: application/sdp.
Allow: INVITE,ACK,CANCEL,BYE.
Via: SIP/2.0/UDP 6X.2X1.4.1X5;branch=z9hG4bKcff3.da8920b2.0.
Via: SIP/2.0/UDP 6X.2X1.4.9X;branch
Per my understanding, it all depends of all other services running on your
machine. If you have dedicated kamailio server running nothing but kamailio
then you can set as much RAM you want. However make sure to keep base
minimum RAM for operating system, e.g. 512 MB (or even less if you are
running
Hello,
I have such infrastructure:
Asterisk1 (provider 1) - IP1
Asterisk2 (provider 2) - IP2
Softphones on client computers with configured two IPs to two Asterisk servers.
User has to manually switch between each server.
I need to change this that user will have only one account configured (eg
Hi,
this may make your config a little more tricky:
You could instruct the RTPProxy/Mediaproxy-ng to "Trust" the IP in the
SDP for the Originating call to PSTN and the Freeswitch (add "r" the
flags).
Have you verified/checked, that rtpproxy_manage() is really called for
both the 183 and the 200 OK
Hi Naveen
We use Siremis for our web interface to configure kamailio.
http://siremis.asipto.com/
I hope that helps.
All the best.
Will Ferrer
On Wed, Nov 27, 2013 at 9:16 PM, Naveen Chand wrote:
> Dear Sir,
>
>
>
> well i would like to know,is there any web interface for kamailio to
> config
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