Hi,
I wanted to achieve the following scenario with Kamailio and Asterisk/FreeSwitch
Kamailio as a trunk for Receiving incoming calls and forward to Asterisk and Make outgoing calls. My current working setup like this
VoIP Provider -> Kamailio -> Asterisk Clusters
Where Kamailio(NAT for public) and Asterisk in the same local network, So I had to do only one thing just create the SIP trunk in Asterisk for authenticating all the Kamailio Incoming calls and making outgoing calls like this
[kamailio]
host=172.18.1.1
type=friend
disallow=all
allow=ulaw
context=trunkinbound
This is IP based authentication, This is working fine since both servers are in the same local network or having the public IP's
What I want is, I want to register the Kamailio Trunk in an asterisk with Username Password authentication,
My current setup for making that,
VoIP Provider -> Kamailio(Public) -> Asterisk(Randomly Changing Public IP, Broadband Connection)
I want to register Kamailio SIP in asterisk trunk so that I can forward the incoming calls to asterisk via SIP extension
[kamailio]
host=172.18.1.1
user=user
secret=password
type=friend
disallow=all
allow=ulaw
context=trunkinbound
Let me know if anyone doesn't understand my concern
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Hello,
starting to continue the discussions here on mailing lists about some of
the approached topics during the last IRC meeting -- there will be
couple of them.
The one now is about finding and deploying a ranking/rating system that
could eventually help community members to make decisions easier in
regard to what components to use in their voip systems.
The need was exemplified by user verticelo with the dificulty to decide
what KEMI scripting language to use. While maintaining all the app_xyz
are expected to be easy, at least I know that for those I created, the
arguments were also from the perspective of the community. Like what
others are using, so one can expect good assistance, hints and share of
knowledge via community forums. This can be also extended to related
tools, like what people use for shared IP high availability of kamailio,
preferred database servers, ...
This email is to ask if those that didn't participate to IRC devel
meeting find such system useful and, if there is a positive feedback, is
anyone aware of some OSS that we can deploy on kamailio.org for such
purpose? It should be something that allows posting a topic (title and
short description) along with a list of answers/options (each can be
again like a title and short description) and provide a way to rate the
options (like, thumbs up, starts, ...), eventually allowing also
comments for each option. I guess it sounds a bit like stackoverflow ...
Being users related, the email is sent only to sr-users, let's keep the
discussion on this mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com
Kamailio Advanced Training - Mar 25-27, 2019, in Washington, DC, USA -- www.asipto.com
Hello,
I'm looking for a solution to integrate legacy devices to a SIP network.
More precisely, I need to forward to and receive Clearmode RTP traffic (see
[1]).
1. Do you know any Kamailio-compliant RTP engine (rtpproxy, rtp engine, ..)
that support this on a Debian host ?
2. Suggestion ?
[1] https://tools.ietf.org/html/rfc4040
Best regards
Hi there,
Well it's mostly rtpengine question but didn't know where should I send
it and probably the answer will be more or less useful for kamailio users.
By default if an offer has ICE candidates rtpengine leaves them
untouched and just inserts itself as an additional low priority
candidate and also rewrites c= line for the callee without ICE support.
Using ICE=force-relay rtpengine inserts itself as a "relay" candidate
what is great but doesn't touch c= line and callee without ICE have no
chance to send RTP to rtpengine.
Is it possible somehow to make a third behavior like force-relay +
rtpengine IP in c= param? It would allow any callee to send media to
rtpengine regardless of ICE support.
Hello Mojtava,
> Did you use FHOSS or IMS_DIAMETER_SERVER module?
No, I am using our own hss server.
>Anyway what mandatory parameters do you mean exactly?
According to RFC6733 section 5.3.1 -
https://tools.ietf.org/html/rfc6733#section-5.3.1
The Capabilities-Exchange-Request (CER), indicated by the Command
Code set to 257 and the Command Flags' 'R' bit set, is sent to
exchange local capabilities. Upon detection of a transport failure,
this message MUST NOT be sent to an alternate peer.
When Diameter is run over SCTP [RFC4960] or DTLS/SCTP [RFC6083],
which allow for connections to span multiple interfaces and multiple
IP addresses, the Capabilities-Exchange-Request message MUST contain
one Host-IP-Address AVP for each potential IP address that MAY be
locally used when transmitting Diameter messages.
Message Format
<CER> ::= < Diameter Header: 257, REQ >
{ Origin-Host }
{ Origin-Realm }
1* { Host-IP-Address }
{ Vendor-Id }
{ Product-Name }
[ Origin-State-Id ]
* [ Supported-Vendor-Id ]
* [ Auth-Application-Id ]
* [ Inband-Security-Id ]
* [ Acct-Application-Id ]
* [ Vendor-Specific-Application-Id ]
[ Firmware-Revision ]
* [ AVP ]
Kamailio 5.1 with loaded cdp, cdr_avp, ims_icsf modules and default
configuration described here
https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/icscf
sends Host-IP-Address but Kamailio 5.2 with the same configuration
doesn't.
Any idea how to fix it?
Regards,
Pavel Siderov
>Hello,
>The CER/CEA messages are used just for connections between two peers
>diameter interface. Did you use FHOSS or IMS_DIAMETER_SERVER module? In
>both cases, you should not have any issue. Anyway what mandatory parameters
>do you mean exactly?
>WIth Best Regards. Mojtava
><https://www.kamailio.org/docs/modules/5.1.x/modules/ims_diameter_server.html>
>On Tue, Mar 12, 2019 at 8:17 PM Pafel <pafels at gmail.com <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote:
>
>>* Hello,
*>>>>* I am testing ims setup with kamailio 5.2 using the default configuration
*>>* provided here:
*>>* https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/
<https://github.com/kamailio/kamailio/tree/master/misc/examples/ims/>
.
*>>* Тhere is an issue I am facing with cdp avp in request sent to hss. There
*>>* are some missing mandatory parameters in capabilities exchange request like
*>*> 257 host-ip-address for example. Any idea how to add it via icscf.xml or
*>*> kamailio.cfg ? The same parameter is passed to hss with kamailio 5.1 using
*>*> default ims configuration. I read cdp and cdp_avp documentation but
*>*> couldn't understand if it is possible to add it.
*>>>>* Regards,
*>>* Pavel Siderov
*>>* _______________________________________________
*>>* Kamailio (SER) - Users Mailing List
*>>* sr-users at lists.kamailio.org
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
*>*> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
*>>
Hi,
Just curious if siremis works would work on Kamailio 5.2?
The webside might not be updated:
Installation
Siremis installation is straightforward, being done mostly through a web
wizzard. In just a few steps and less than 10 minutes you have your Siremis
web site set up and ready to grant you easy administration of your Kamailio
SIP server.
You need to install first a compatible version of Kamailio SIP Server
<https://www.kamailio.org/>.
Siremis v5.1.x (devel)
*Last release: v5.1.0 (devel version)*
*Compatible with: Kamailio 5.1.x, 5.0.x*
Thanks!
Joel.
Hi All,
I started playing with async kamailio feature, in particular using
"async_route" function provided by ASYNC module
(https://www.kamailio.org/docs/modules/5.1.x/modules/async.html#async.f.asyn…).
My scenario is the following:
- A calls B
- B answer the call
- inside onreply_route[reply] I catch the 200OK message and I suspend
the transaction for X seconds.
- after X seconds the transaction is resumed and the call continues as
expected
Everything looks good, but I encountered an issue trying to drop reply
re-transmission messages.
My desire is to drop all the messages received from B (basically 200OK
re-transmissions) meanwhile the transaction is suspended.
I tested "t_is_retr_async_reply" function provided by TM module
(https://www.kamailio.org/docs/modules/5.1.x/modules/tm.html#tm.f.t_is_retr_…),
but I'm not able to get it worked as expected:
- t_is_retr_async_reply() function can be used only inside named
'onreply_route' (no default 'onreply_route' or standard 'reply_route')
- drop() function, on the contrary, doesn't allow to drop 200OK
messages inside named 'onreply_route' (but only in default
'onreply_route' or standard 'reply_route').
Do you have any suggestion on how I can achieve my task?
Thank you very much for your support.
Marco
Hello,
I was wondering that is there a command / procedure available in
Kamailio that would do dialplan.dump on a certain dialplan and then
export/write the output to another sql database where another
dialplan table is present? This would be a sort of backup procedure
that I'm planning.
Cheers,
Olli
