Hello,
Kamailio version: 3.1.5
I have been trying to use presence for Event:Dialog. I have used PUA_Dialoginfo module to accomplish this task.
The PUA_dialoginfo module states that if "override_lifetime" is not used, the value of the expires is taken from dialog module.
I have tried the module without the "override_lifetime" which did not create the dialog in the presentity.
When used debug, I did see the xml being generated but finds the dialog "expires=0" and deletes the xml(please find the log below).
But when "override_lifetime" is set, the dialog in the presentity table is set until. But this has a problem,
modparam("pua_dialoginfo", "override_lifetime", 300)
1. if a call is still going on more than the override_lifetime the presentity is deleted.
2. the presentity information is available until the "override_lifetime" even after the call is hung-up.
How do I get this module working without the "override_lifetime" being used.
The Log:
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:242]: new_body:#012<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:user@mydemo.com">#012 <dialog id="112014dff048e71e" call-id="112014dff048e71e" direction="initiator">#012 <state>Trying</state>#012 <remote>#012 <identity>sip:user01@mydemo.com</identity>#012 <target uri="sip:user01@mydemo.com"/>#012 </remote>#012 <local>#012 <identity>sip:user@mydemo.com</identity>#012 <target uri="sip:user@mydemo.com"/>#012 </local>#012 </dialog>#012</dialog-info>#012
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:81]: freeing 1 columns
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:290]: publish uri= sip:user@mydemo.com
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x8314aa4
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:366]: qm_malloc(0x829bee0, 756) called from pua_dialoginfo: dialog_publish.c: dialog_publish(302)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314aa4), called from <core>: db_res.c: db_free_columns(86)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:406]: qm_malloc(0x829bee0, 756) returns address 0x82e84bc frag. 0x82e84a4 (size=900) on 1 -th hit
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314a8c alloc'ed from db_mysql: km_res.c: db_mysql_get_columns(78)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:51]: publ:
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x8318650
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:52]: uri= sip:user@mydemo.com
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318650), called from <core>: db_res.c: db_free_columns(95)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:53]: id= DIALOG_PUBLISH.112014dff048e71e
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8318638 alloc'ed from <core>: db_res.c: db_allocate_columns(148)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua_dialoginfo [dialog_publish.c:54]: expires= 0
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x8318684
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:403]: pres_uri=sip:user@mydemo.com
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8318684), called from <core>: db_res.c: db_free_columns(100)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:121]: core_hash= 504
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x831866c alloc'ed from <core>: db_res.c: db_allocate_columns(157)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [hash.c:171]: record not found
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:54]: freeing 1 rows
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:444]: insert type
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_row.c:97]: freeing row values at 0x8314b14
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:448]: UPDATE_TYPE and no record found
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x8314b14), called from <core>: db_row.c: db_free_row(98)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: DEBUG: pua [send_publish.c:454]: request for a publish with expires 0 and no record found
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: <core> [mem/q_malloc.c:450]: qm_free: freeing frag. 0x8314afc alloc'ed from <core>: db_row.c: db_allocate_row(114)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3289]: <core> [mem/q_malloc.c:428]: qm_free(0x829bee0, 0x82e84bc), called from pua_dialoginfo: dialog_publish.c: dialog_publish(357)
Oct 21 12:12:46 SIPTest /usr/local/sbin/kamailio[3288]: DEBUG: <core> [db_res.c:62]: freeing rows at 0x8314adc
Regards
Gnaneshwar
Hi All,
Kamailio generate a core at line below
if(_msg->contact!= NULL || _msg->contact->body.s!= NULL){
_msg is a sip_msg struct that my module receive from kamailio. I want
verify if on that request messagem have a contact header, but a core is
being generated when contact header isn't present on message.
Someone knows why this is happening?
Cheers
Hi folks,
I would like to know what to use for the following scenario. I want
users to be able to test their registration by sending REGISTER and
Kamailio sends them 200 OK if credentials are OK. However I don't want
the aor to be stored in location table. This REGISTER should be only
used for testing the settings. After some time I would allow the
registration to be stored and users will normally receive calls.
I tried to implement this by not doing save(location) when the flag
from load_credential is set. Unfortunately it is the function save who
sends 200 OK. When it is not being called, a user agent keeps sending
REGISTER until SIP triggers fail.
So is there any other common practice how to test registers?
Is there any flag that can prevent storing location?
Can I send 200 OK to these user in any other way?
Thanks
Efelin
Hello,
On 12/7/11 10:03 AM, Pavel Segeč wrote:
>
> HI,
>
> Thank you. Just as I sent my previous mail I look into kamctl command
> syntax and I found
>
> -- command 'rpid' - manage Remote-Party-ID (RPID)
>
> rpid add <username> <rpid> ......... add rpid for a user (*)
>
> rpid rm <username> ................. set rpid to NULL for a user (*)
>
> rpid show <username> ............... show rpid of a user
>
> How this work?
>
this is for adding/removing/showing the rpid column in subscriber table
-- if you look in database, there is a column named rpid for subscriber
table.
Practically, setting it and loading with load_credentials is a way to
use it in the config file. In config then you can use append_hf() add
header as you need, or if it is rpid specific format, see siputils
module for other options:
http://kamailio.org/docs/modules/stable/modules_k/siputils.html
Cheers,
Daniel
> pavel
>
> *From:*Daniel-Constantin Mierla [mailto:miconda@gmail.com]
> *Sent:* Tuesday, December 06, 2011 11:14 PM
> *To:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Cc:* Pavel Segeč
> *Subject:* Re: [SR-Users] Remote Party ID
>
> Hello,
>
> On 12/6/11 2:50 PM, Pavel Segeč wrote:
>
> Hi,
>
> I'm looking for some recommendations or guidlines. My leading IT
> department which is providing PSTN connectivity requires to include
> Remote Party ID in SIP Messages in a case where From URI is not in
> telco format (I prefer email like style of addressing). How to simply
> assign to an user its Remote Party ID which will be used when PSTN
> calling will occure? On what should I focus on? To use user
> preferences with AVP pairs? or something another?
>
> since it is about one attribute associated with the caller, a
> recommended way is to add a new column in subscriber table, say it is
> named 'rpid'.
>
> Then, in config file, set the load_credentials parameter of auth_db to
> load the rpid value and store in an avp. After you authenticate the
> caller, the avp will be set to the rpid value, see:
>
> http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2528175
>
> Using this option, practically you use same db query that loads the
> user password for authentication to fetch the rpid value.
>
> Cheers,
> Daniel
>
> --
> Daniel-Constantin Mierla --http://www.asipto.com
> http://linkedin.com/in/miconda -- http://twitter.com/miconda
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Ladies and Gentlemen,
I'm happy to announce the official release of sip provider CE v2.4, an
easy-to-use, free and open-source soft-switch built on top of Kamailio
and Sems. As always, we provide a dead-simple installer to start from
scratch, upgrade scripts from v2.2 and Virtualbox and VMware images to
test-drive a turn-key installation.
http://www.sipwise.com/news/announcements/spce-v2_4-release/
The SPCE shows the capabilities of what can be created with flexible and
powerful software like Kamailio and Sems, when the proper provisioning,
billing, configuration and packaging glue is being put around that. The
SPCE v2.4 is currently deployed around the world acting as an SBC in
front of old legacy 3rd party switches to provide SIP service to mobile
clients, as a Class4 peering concentrator to provide ENUM/PSTN access to
other soft-switches, and as standard Class5 soft-switches for all kinds
of access networks.
It's ideal for new Kamailio/Sems users to have a head-start over
beginning completely from scratch, because the SPCE comes with customer-
and admin-web-interfaces, SOAP/XMLRPC-APIs, provisioning and rating
engines etc. No need to tinker with all the various configuration files,
unless you want to. :)
You can find the full release announcement with the changelog highlights
over here:
http://www.sipwise.com/news/announcements/spce-v2_4-release/
Thanks,
Andreas
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Interested in working with one of the biggest Voice over IP networks in
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Best regards,
Henning
--
Henning Westerholt - Head of IT Operations Internet Access & Communications
1&1 Internet AG, Brauerstraße 48, 76135 Karlsruhe, Germany
Hello,
On 12/6/11 2:50 PM, Pavel Segeč wrote:
>
> Hi,
>
> I'm looking for some recommendations or guidlines. My leading IT
> department which is providing PSTN connectivity requires to include
> Remote Party ID in SIP Messages in a case where From URI is not in
> telco format (I prefer email like style of addressing). How to simply
> assign to an user its Remote Party ID which will be used when PSTN
> calling will occure? On what should I focus on? To use user
> preferences with AVP pairs? or something another?
>
since it is about one attribute associated with the caller, a
recommended way is to add a new column in subscriber table, say it is
named 'rpid'.
Then, in config file, set the load_credentials parameter of auth_db to
load the rpid value and store in an avp. After you authenticate the
caller, the avp will be set to the rpid value, see:
http://kamailio.org/docs/modules/stable/modules_k/auth_db.html#id2528175
Using this option, practically you use same db query that loads the user
password for authentication to fetch the rpid value.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
I think the most straight-forward way is to load the RPID in user preferences as AVP,
and print the associated AVP from SER script. I guess there is an example of such
in the OOB script -- have you tried to look there?
-jiri
On 12/6/11 2:50 PM, Pavel Segeč wrote:
> Hi,
>
>
>
> I'm looking for some recommendations or guidlines. My leading IT department
> which is providing PSTN connectivity requires to include Remote Party ID in
> SIP Messages in a case where From URI is not in telco format (I prefer email
> like style of addressing). How to simply assign to an user its Remote Party
> ID which will be used when PSTN calling will occure? On what should I focus
> on? To use user preferences with AVP pairs? or something another?
>
>
>
> thank
>
>
>
> palo73
>
>
>
>
>
> _______________________________________________
> 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
Hi,
I'm looking for some recommendations or guidlines. My leading IT department
which is providing PSTN connectivity requires to include Remote Party ID in
SIP Messages in a case where From URI is not in telco format (I prefer email
like style of addressing). How to simply assign to an user its Remote Party
ID which will be used when PSTN calling will occure? On what should I focus
on? To use user preferences with AVP pairs? or something another?
thank
palo73