Hi Freinds,
Has anyone done configuring the presence module only with DB.
In my requirement, I am going to have multiple servers and share the same
DB for presence info and
enable Publish/Unpublish and Subscribe/UnSubscribe for all the clients who
will be connecting to different Kamailio servers.
Also expecting the NOTIFY to happen without any issues.
I tried to get this done with modparam("presence", "subs_db_mode", 3)
But failed.
Please guide me for this.
Best Regards,
Roy.
Hey Guys,
I was wondering if I could get some guidance on my ideas.
I would like to use dispatcher ds_select_dst on ruri in a branch. Is
this scenario feasible or out of internals (winning branch maybe)
impossible?
The idea behind my scenario is to be able to dispatch separately
branches received from redirect server.
Thanks in advance for any tip!
DanB
Hi everybody,
I've been following Daniel's excellent tutorial at
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
and managed to get registrations and outbound dialing to run. The problem
is that whenever I try to call a local extension (directly or via trunk),
CLI reports
== Everyone is busy/congested at this time (1:0/0/1)
and plays the according VM message. "sip show peers" shows the extension as
online / the registration correctly passed along by Kamailio:
1001/1001 198.23.139.21 D A 5060 OK (1 ms)
198.23.139.21 is the server IP and used for both Kamailio (5060) and
Asterisk (5080). Asterisk only allows for UDP, Kamailio for for UDP and TCP
connections.
The server is connected directly to the internet and the only NAT is on the
client side. I have disabled NAT support both in Asterisk and Kamailio
because the clients support ICE and can connect using TURN to the Asterisk
echo-test just fine. Enabling NAT support (with and without rtpproxy) in
Kamailio doesn't solve the above problem (and doesn't seem to be able to
traverse all NAT situations in any case).
To be clear: The Asterisk configuration itself is working flawlessly. I
don't encounter the problem at all if I connect directly to the Asterisk
server, either in its original, standalone configuration or on port 5080
(after disabling the ACL allowing only the Kamailio IP and re-enabling the
password).
Please let me know if you need specific logs (I wouldn't want to needlessly
clutter the original description with useless information) to take a closer
look at the problem. If this is a known issue, a friendly pointer would be
much appreciated as well! I've searched for hours and hours and never
seemed to find anything helpful. Have a great day and with
Best Regards,
Michael
Daniel,
Good catch, I do not recall it but it could be I have used in first case
(when it worked) normal clone, and it second just the dept=1. I have
just rechecked now and it works checking out directly your commit, so
sorry for additional noise.
Thanks and have a good one!
DanB
Hi everybody,
I noticed (after checking the files in utils/kamctl/mysql) that kamailio
uses the MyISAM storage engine for its database tables in mysql, is there
some particular reason for this? Isn't the read/write concurency that
InnoDB provides desirable?
And on a side note, enabling 'low_priority_updates' (
http://dev.mysql.com/doc/refman/5.0/en/server-options.html#option_mysqld_lo…)
that prioritizes reads over writes in my mysql config can lead to any
potential issues with kamailio? Anybody tested this before?
Regards,
Lefteris Zafiris
Hey Daniel,
Thanks for the answer. Now if I think deeper the AVPs should be also
overwritten by sequential ds_select_dst so I guess it can end up nasty
(was kinda planning to use two dispatchers on the same transaction-one
to reach redirect server and second to process the answer in failure route).
Will anyway redesign my ideas ;).
Have a good one!
DanB
Daniel,
The command which failed for me it was checking out your commit in the
git sources:
Here is how I run it:
"""
root@IECDev:/usr/local/src/kamailio/kamailio.git# git checkout
74bf08ef30cf18fb609c2c067584f1515ede23da
fatal: reference is not a tree: 74bf08ef30cf18fb609c2c067584f1515ede23da
root@IECDev:/usr/local/src/kamailio/kamailio.git# git branch
* master
"""
DanB
Hello!
Radius pairs has two similar types: integer and ipaddr.
generate_avps
In modules/misc_radius/functions.c, function generate_avps()
we see that attribute checked for PW_TYPE_INTEGER (integer) type.
That incorrect: when rad attribute has PW_TYPE_IPADDR (for example
Framed-IP-Addr) then we raise error, but we must processing ipaddr same
as integer.
As result misc_radius cant sent queries with PW_TYPE_IPADDR at all:
error "attribute ## not integer" raised.
Need correct this same way:
@@ -142,7 +196,8 @@ static void generate_avps(struct attr *attrs, VALUE_PAIR* received)
do { \
if ((_len) != 0) { \
if ((_len) == -1) { \
- if (_attrs[_attr].t != PW_TYPE_INTEGER) { \
+ if (_attrs[_attr].t != PW_TYPE_INTEGER) \
+ if (_attrs[_attr].t != PW_TYPE_IPADDR) { \
LM_ERR("attribute %d is not of type integer\n", \
_attrs[_attr].v); \
goto error;
\
Daniel, what about my previous patch? May be resend with this corrections?
--
WBR, Victor
JID: coyote(a)bks.tv
JID: coyote(a)bryansktel.ru
I use FREE operation system: 3.8.4-calculate GNU/Linux
Hello,
On 5/27/13 1:03 AM, Tony Turner wrote:
>
> (re-sent due to bad English)
>
> Hi,
>
> I have 3 registered test users, how can I configure Siremis to do the
> trunk to freeswitch using LCR or Carrierroute rather than using the
> code below. I am keen to be able to setup Inbound + Outbond trunks via
> Siremis. Do you know if there is a manual for Siremis or a how to /
> step by step?
>
> if($rU =~"^01") {
> $ru = "sip:" + $rU + "@__FREESWITCHIP__";
> route(RELAY);
> exit;
> }
>
Siremis is a web interface for the database use for kamailio, not for
kamailio configuration file. For using LCR or carrierroute, just read
the documentation of these modules and add to your configuration file.
LCR should be rather simple to setup, besides loading it and setting
module parameters, you have to use two new functions in the routing
logic of config file. Carrier route should be pretty much the same, just
that I haven't checked it lately.
> Currently with the above code if a user phones one of the other
> extensions it tries to route out to the PSTN network rather than the
> extension, is that because I have put the above code in the wrong
> place in the config so it never gets to the code to route to the
> extension? (routing to PSTN is fine)
>
> Or do I need an if else statement wrap checking if local user, please
> can you give me some idea of the code ...
>
If you don't have numbering rules that you can sort out if it is local
subscriber or not, the best is to check with subscriber table (see
uri_db or sqlops modules, both of them can help for this).
Cheers,
Daniel
>
> Thanks
>
> Tony
>
> *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com]
> *Sent:* 20 May 2013 16:19
> *To:* tony.turner(a)nodemax.com <mailto:tony.turner@nodemax.com>;
> Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] Kamailio + Siremis Outbound route
>
> Hello,
>
> if you want to send all calls that arrive to kamailio having the
> prefix 01 to freeswitch:
>
> if($rU =~"^01") {
> $ru = "sip:" + $rU + "@__FREESWITCHIP__";
> route(RELAY);
> exit;
> }
>
> Be sure calls are authenticated at that point and, if needed, the call
> is not actually coming from freeswitch.
>
> Cheers,
> Daniel
>
> On 5/20/13 11:33 AM, Tony Turner wrote:
>
> Hi
>
> Version Kamailio v4.0 + Siremis installed on Debian Wheezy via
> apt-get install
>
> I want to use Kamailio as a proxy edge register to our network.
>
> I have installed Kamailio and freeswitch.
>
> I can register on Kamailio but I can't route a call from my sip
> client from Kamailio to freeswitch and out to PSTN
>
> Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway
> --- Carriers
>
> If I register direct on Freeswitch I can route out to PSTN but I
> don't understand Kamailio routing.
>
> Can someone let me how I route say from SIP client registered on
> Kamailio to prefix 01% which goes out to Freeswitch
>
> Many Thanks
>
> Tony
>
>
>
> _______________________________________________
>
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>
> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>
>
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla -http://www.asipto.com
> http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
> *http://asipto.com/u/katu *
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hey Daniel,
Many thanks for so fast reaction. As I always say, opensource beats
proprietary in more than one way ;).
The changes you did work smooth for me, the new redirect comes with
proper enclosure characters for uri in branches:
"""
#
U 2013/05/28 10:44:53.201933 127.0.0.1:5070 -> 10.10.10.21:5060
SIP/2.0 302 iClass4-AP Redirect.
Call-ID: ab9c9470915850c7ec7c8922dc4eed95@0:0:0:0:0:0:0:0.
CSeq: 2 INVITE.
From: "dan" <sip:dan@iec.itsyscom.com>;tag=e65ed733.
To: <sip:dan@iec.itsyscom.com>;tag=46a6e639fa023622ac1ba4fea686e961.8db5.
Via: SIP/2.0/UDP
10.10.10.21;branch=z9hG4bKf5a1.88785e1b21984da781a03be04b99e6b9.0.
Via: SIP/2.0/UDP
10.10.10.154:5060;rport=5060;branch=z9hG4bK-313932-636d0be5350bb5038fc59a6d30111b9d.
Contact:
<sip:dan@10.10.10.141:3072;line=113rlmlo;rcv=sip:10.10.10.141:3072>;q=1,
<sip:dan@10.10.10.154:5060;transport=udp;registering_acc=iec_itsyscom_com;rcv=sip:10.10.10.154:5060>.
Server: iClass4-AP 4.0.0.
Content-Length: 0.
"""
Have a good one!
DanB
PS: Checking out specific commit fails for me after Juha's merge, could
be I am doing something wrong though.