Hi,
I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only
Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER,
How can I achieve that?
Thanks,
Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server?
Good day from Singapore,
After reading recent reviews, I gather that Asterisk is the gold
standard when it comes to open source VoIP systems and it is the most
famous open source PBX out there.
Article: Compare the Top 10 Best Open Source PBX Software of 2020
Link:
https://www.voipreview.org/business-voip/best-open-source-pbx-software
Article: Top 10 Free Open Source PBX Software Solutions
Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/
The following is an excerpt from Wikipedia:
"Asterisk is a core component in many commercial products and
open-source projects. Some of the commercial products are hardware and
software bundles, for which the manufacturer supports and releases the
software with an open-source distribution model.
AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software
to realize all telephony functions.
AstLinux is a "Network Appliance for Communications" open-source
software distribution.[15]
FreePBX, an open-source graphical user interface, bundles Asterisk as
the core of its FreePBX Distro[16]
LinuxMCE bundles Asterisk to provide telephony; there is also an
embedded version of Asterisk for OpenWrt routers.
PBX in a Flash/Incredible PBX and trixbox are software PBXes based on
Asterisk.
Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer
PBX, fax, instant messaging and email functions, respectively, before
switching to 3CX.
Issabel is an open-source Unified Communications software which uses
Asterisk for telephony functions. It was forked from the open-source
versions of Elastix when 3CX acquired it.
*astTECS uses Asterisk in its VoIP and mobile gateways."
Link: https://en.wikipedia.org/wiki/Asterisk_(PBX)
I would like to DIY/setup an IP PBX appliance/server using free open
source projects.
Which free open source project, mentioned in the list and links above,
would you recommend to DIY my IP PBX appliance/server?
Should I buy a desktop computer or get one of those appliances listed in
the link below to serve as my IP PBX appliance/server?
Link:
https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero…
Please also refer me to very good, detailed and well explained
guides/tutorials/manuals on setting up open source IP PBX
appliances/servers.
Lastly, please recommend a cheap and affordable IP phone (suggest brand
and model) to go along with my DIY open source IP PBX appliance/server.
Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020
Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore.
Thank you very much.
-----BEGIN EMAIL SIGNATURE-----
The Gospel for all Targeted Individuals (TIs):
[The New York Times] Microwave Weapons Are Prime Suspect in Ills of
U.S. Embassy Workers
Link:
https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html
********************************************************************************************
Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's
Academic
Qualifications as at 14 Feb 2019 and refugee seeking attempts at the
United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug
2019) and Australia (25 Dec 2019 to 9 Jan 2020):
[1] https://tdtemcerts.wordpress.com/
[2] https://tdtemcerts.blogspot.sg/
[3] https://www.scribd.com/user/270125049/Teo-En-Ming
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Is there a way to obtain the effect of msg_apply_changes in a branch route?
I want to do:
route[BRANCHMANAGE] {
rtpengine_manage()
msg_apply_changes()
// further mangle the SDP with textops for
// obtuse UAs
}
The reason for this, is that after forking, I have some UAs that are
extremely picky about SDP, and I need to mangle the body to make them
happy. (4000 char limit, 68 attribute limit on SDP body - ever seen
that? Compared with the absolutely gigantic OFFERs from WebRTC
signalling libraries...)
Now rtpengine_manage() and textops are working on separate copies of
the msg body and the results don't stack correctly.
Cheers
Anthony Alba
Thanks for the link.
One issue I've noticed:
If you have an empty comment line (just # on a single line), then the next line is wrongly highlighted as comments.
For example:
#
loadmodule "db_postgres.so"
> i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode.
> its great!
This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.
Hi Patrick,
better to contact our sr-users list with the usage related questions, added to CC.
Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further.
Feel free to ask again on sr-users after you have got more details.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-dev <sr-dev-bounces(a)lists.kamailio.org> On Behalf Of Patrick Leybag
Sent: Wednesday, November 25, 2020 6:26 AM
To: sr-dev(a)lists.kamailio.org
Subject: [sr-dev] kamailio SIP and RTP proxy
Hi, Can someone help me?
I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work.
Any help is much appreciated. Thank you in advance
Hi,
I am having a problem on fail overing my master kamailio server and slave
kamailio server.
I use kamailio as a load balancer for my asterisk servers. And it is
working with calls, but when I added failover with keepalived to the slave,
there is no audio during calls.
Thank you,
Rolly
Hello,
I have a kamailio server running behind HAProxy with proxy protocol v2 enabled.
In Kamailio I have set the parameter tcp_accept_haproxy=yes and loaded tcpops module.
UEs are registered using TLS and kamailio sees that the message has received from their real ip address + port and not HAProxy ip + port.
When UE A calls UE B, kamailio is trying to reach UE B using his real ip address and port instead of HAProxy IP address + port.
I know I can get the tcp ip and port of HAProxy using $tcp(c_si) and $tcp(c_sp) but I can’t make it work.
What is the right way to do this? How should I use these variables properly in order to establish the call successfully?
Thanks,
Joey.