Benoit, Alex,
I appreciate the feedback, confirming my assumptions. Good news is the
Vendor finally responded positively and agreed to add in configuration
option to route on INVITE R-URI in next SDK release, expected later
this year. In the meantime I added 'uac_replace_to' function to my
proxy to re-write To: header with R-URI, so far so good.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Chasing the Azeotrope
>
> The To header is a cosmetic, purely logical commentary on the intended ultimate destination of the call, and no routing should ever be done on it. The RURI of the INVITE is the appropriate vehicle for that.
>
> When dealing with proxies, however, one will encounter from time to time the grim reality that there are user agents which expect the To URI and the INVITE RURI to be in alignment. For that scenario, you can use ‘uac’ module’s stateful rewriting functionally (uac_replace_to()) to modify To with Kamailio in a way that displays fidelity and esteem for protocol formalities.
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> > On May 21, 2019, at 9:32 AM, JR Richardson <jmr.richardson(a)gmail.com> wrote:
> >
> > Hey Folks,
> >
> > I could use some feedback to see if my mind is right on this topic.I'm
> > in a discussion with a software Vendor (respectfully unnamed) with
> > terminal software routing on To: header info. It's my contention any
> > modern SIP software should use INVITE header to retrieve DID info and
> > route from there.
> >
> > My reasoning is due to the To: header should contain the original
> > intended DID, Extension, Name, AoR or whatever else the original SIP
> > transaction wanted to contact, but being in a multi-carrier,
> > multi-hop, call forwarding environment, the INVITE header contains the
> > hop-to-hop true intention of where the call should route at any given
> > transaction. So in the case of terminal software the To: header could
> > likely be irrelevant where as the INVITE header is accurate.
> >
> > My most recent response from the Vendor suggest I put a SIP Proxy in
> > front of the terminal software and transform the To: header to
> > whatever I need for this application. Although I can do this, I'm
> > concerned with breaking RFC for CANCEL and BYE transaction sent back
> > from the terminal software that may not be recognized by upstream
> > origination carriers.
> >
> > Plus modifying To: header is widely frowned upon by most folks.
> >
> > Thanks.
> >
> > JR
Hello,
Kamailio SIP Server v5.2.3 stable release is out.
This is a maintenance release of the latest stable branch, 5.2, that
includes fixes since the release of v5.2.2. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.2.x. Deployments running previous v5.2.x
versions are strongly recommended to be upgraded to v5.2.3.
For more details about version 5.2.3 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2019/05/kamailio-v5-2-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hi,
My name is Mohit Chawla working on LTE Network integration with Kamailio IMS
server. We are facing issues in installation & Configuring of kamailio IMS
on CentOS 7 due to no documentation is available on web (or at least I am
not able to find one). Primarily I am interested in installing packages (not
building from source code as of now) and use it for further testing. I can
go with building the package from source if its more easier and quick.
Can you guys help me share any document/wiki which is helpful for my cause ?
I will be really grateful if someone can reply me by today.
Thanks and Regards,
Mohit
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Hello,
I am considering to release Kamailo v5.2.3 out of latest git branch 5.2
next week, likely on Wednesday, May 22. As usual, if you are aware of
issues not reported to the github bug tracker, fill a report there as
soon as possible to give a chance to be analyzed and fixed. Also, if
there was a fix for an issue that you were reporting, check to see if it
was backported to be sure it is not forgotten.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hello!
I was wondering if someone is using remove_hf in onreply_route and if yes
on which Kamailio version? I'm using Kamailio v5.2.1 and as soon as I add
remove_hf function to onreply_route Kamailio will fail to start with
following error messages:
ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free():
qm_find_free(0x7fc1dc044010, 27120); Free fragment not found!
ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc():
qm_malloc(0x7fc1dc044010, 27120) called from core: core/io_wait.c:
init_io_wait(524), module: core; Free fragment not found!
Thank you for any suggestions!
Hey Folks,
I could use some feedback to see if my mind is right on this topic.I'm
in a discussion with a software Vendor (respectfully unnamed) with
terminal software routing on To: header info. It's my contention any
modern SIP software should use INVITE header to retrieve DID info and
route from there.
My reasoning is due to the To: header should contain the original
intended DID, Extension, Name, AoR or whatever else the original SIP
transaction wanted to contact, but being in a multi-carrier,
multi-hop, call forwarding environment, the INVITE header contains the
hop-to-hop true intention of where the call should route at any given
transaction. So in the case of terminal software the To: header could
likely be irrelevant where as the INVITE header is accurate.
My most recent response from the Vendor suggest I put a SIP Proxy in
front of the terminal software and transform the To: header to
whatever I need for this application. Although I can do this, I'm
concerned with breaking RFC for CANCEL and BYE transaction sent back
from the terminal software that may not be recognized by upstream
origination carriers.
Plus modifying To: header is widely frowned upon by most folks.
Thanks.
JR
--
JR Richardson
Engineering for the Masses
Chasing the Azeotrope
Hey guys,
Is there a numeric mapping for $snd(proto)?
Thanks
--
Andy Chen
Sr. Telephony Lead Engineer
achen@ <achen(a)thinkingphones.com>fuze.com
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contents to any other person. Thank you.*
Dear Kamailio Users
I might need a bit help with starting.
We are a TSP and at the moment we use a commercial carrier grade voice
switch, which is very VERY inflexible and where bugs we find either
don't get fixed, or only with a large delay.
As everywhere else we embrace opensource and enjoy the possibility to
quickly fix bugs ourselves (we have a software development department)
we are evaluating Kamailio.
Basically, if we have a centralized SIP routing engine, which can record
CDRS for Billing and do some Number Translations (lookup on
external service or database) for (location based emergency numbers,
call forwarding services, ported numbers, call spam blacklisting) we
could do way more than our actual $$$ carrier switch is capable of.
Kamailio sounds like perfect for this task.
First step:
Get a machine up and running with Kamailio and Siremis, done, that was
easy.
Next:
Connect his SIP wise via what I would call a 'SIP carrier
interconnecton', ip based authenticated trunk between our carrier
switch and Kamailio so we can route some number ranges from our carrier
switch to Kamailio and then continue to test with subscribers,
subscriber trunks and all else we need on Kamailio.
I'm quite fluent in Asterisk (our Voicemail and Announcement Services
are based on Asterisk) and SIP and have come across many different PBX,
but after reading parts of the documentation, and trying myself to
understand how Kamailio talks to other SIP endpoints I am a bit at a
loss.
I would have expected that I would configure our carrier switch as
'remote SIP gateway' or whatever it would be called.
But I start fearing, Kamilio works completely differently. So I might
need a bit of pointing the right direction to get started.
I did try with the 'CarrierRoute Management' or 'Dispatcher' functions,
but I only find very little documentation on this.
So how do I get started? How do I, for starters, tell Kamailio what the
Hostname or IP of my Carrier switch is and how do I set a 'default'
call route to this?
Kind regards
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
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CH-4133 Pratteln Fax +41 61 826 93 01
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Hi,
Any quick thoughts. I am using a REDIS "GET" to retrieve a JSON obj as a
string from REDIS and then the JANSSON get value to extract a key value
from the string and pop it into an AVP ($avp(val)) and wish to ensure the
value is an integer in the AVP. New to this, be kind :-)
Steve
Hello everyone,
I install and setup an IMS Network by kamailio-5.2.2 . I have p-cscf,
i-cscf and s-cscf in three different systems . my Hss is Fokus.
Registration and call between subscribers of network work successfully .
now I want to define IP Trunk in my network which can pass external calls
through my IMS network. I don't know how to define IP Trunk and changes in
which element(pcscf, icscf or scscf) should be done.
