Hello,
I'd like to use a < select > as described here:
http://www.kamailio.org/wiki/cookbooks/5.4.x/selects#contactparams_s
I'm trying to extract a parameter from the Contact URI.
What I've tried to do:
xlog("L_INFO", "contact.uri: $sel(contact.uri)\n"); # OK
xlog("L_INFO", "contact.uri.params: $sel(contact.uri.params)\n"); # null ??
This does not seem to work... see output here :
3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri: sip:+33145678901;tgrp=YC031;trunk-context=YC90@10.31.22.2:5099
3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri.params: <null>
What am I doing wrong ?
Maybe this is the wrong approach for what I'm trying to do ?
Regards,
Nicolas.
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Hello,
I am planning to release next week (likely on Monday 21st December) the next minor version out of branch 5.3, version 5.3.8.
Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker).
Cheers,
Henning
--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com<https://gilawa.com/>
What's the best way of sending a message to all other sessions (and the
receiver) of the same sender?
For example: If user1(a)example.com is logged in to Kamailio from multiple
devices and from one of the devices user1 sends a SIP message to
user2(a)example.com. I want the message forwarded not only to
user2(a)example.com but also to all other user1(a)example.com sessions.
What's the easiest way to achieve this?
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Hi,
Can't figure it out when to use RTPproxy, i am forwarding everything to
asterisk in seem to work without RTPproxy.
The scenario is like this:
Softphone -> INTERNET -> Kamailio - > Asterisk (private ip)
Now it seems to work without any rtpproxy but is it ok?
Hi All,
I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual
IP) functionality, and have run into a rather peculiar issue.
*Setup:*
Caller ------------ KamailioVIP(*Primary *and *Secondary* Kamailio)-------
Callee
HA provided by Keepalived
DMQ is used between the Primary and Secondary Kamailio for dialog sync
*Issue Seen:*
*Suppose the Primary Kamailio has been brought down* and the Secondary one
is actually active and tied to the VIP.
When a call is fired from the Caller, it traverses through the Secondary
Kamailio and lands onto the callee. The dialogs are updated properly. At
this point, the Primary Kamailio is brought up, and dialog state is synced
due to the DMQ module.
The Keepalived will now attach the VIP to the Primary Instance. If the
caller hangs up the phone at this point, the BYE sip message traverses
through the Primary Kamailio instance to the callee and the call is
cleared, but there are two issues here.
1. The Primary Kamailio throws an error in *ds_load_remove() *saying
that it cannot find the load for the specific call id. This is apparently
due to the fact that the dispatcher data is not synced between the two
modules but dialog data is. So dialog wise, things are fine. Can this be
fixed somehow?
2. The above is still as grave a problem as the dialogs are cleared. BUT,
if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* kamailio even
well after the call is cleared, the *DLGLOAD* shows 1. Since we are using
the *Call Load based distribution* for the dispatching, this is effectively
one call stuck on the dispatcher, which leads to resource leak.
Is this a known issue, and if so, do we have a way to circumvent this?
Regards,
Harneet Singh
--
"Once you eliminate the impossible, whatever remains, no matter how
improbable, must be the truth" - Sir Arthur Conan Doyle
Hello,
(cross-posting to kamailio public groups, a few may get this message
duplicated).
Fosdem 2021 is going to be online, meaning that it opens the possibility
for everyone to do a presentation on the RTC Devroom, no longer being
required to travel to Brussels. Therefore I am sending here the link to
CFP in case someone would be interested to present about Kamailio or
something else RTC-related:
* https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md
New speakers are always welcome! Only have in mind that the event is
targeting developers and source code development, therefore your
proposals have to be made from this perspective.
Note also that submission deadline is 20th of December, very short time
frame from now.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Here is my setting:
modparam("dmq", "server_address", "sip:0.0.0.0:5090" )
modparam("dmq", "notification_address", "sip:0.0.0.0:5090")
modparam("dmq", "multi_notify", 1)
modparam("dmq", "num_workers", 2)
modparam("dmq", "worker_usleep", 0)
modparam("dmq", "ping_interval", 90)
I have tried localhost and I believe other options but to no avail.
I also get the following message:
ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial
nodelist from sip:0.0.0.0:5090
How do I set up DMQ for adding and removing any Kamailio instance to/from
the cluster and any Kamailio instance could serve any request coming from
any client? ie. all Kamailio instances keep the current state in DMQ.
I also have dmq_usrloc enabled.
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Dear Team,
I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog.
but can not able to capture in kamailio.cfg.
how can I prepared ivr using kamalio?
Please find the attached syslog.
Regards
Amit Pal
I am sending JWT tokens via SIP URL through JSSIP for authentication before
websocket connection is made.
However, I am getting errors like:
ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri <
*/sip?jwt=..*
I have the following setting in kamailio config:
modparam("xhttp", "url_match", "^/sip") // only match this path
Is this a bug or some configuration issue? How do I resolve it?
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this email from your system. Email transmission cannot be guaranteed to be
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