sr-users December 2020

sr-users@lists.kamailio.org

67 participants
86 discussions

How to DIY/Setup An Open Source IP PBX Appliance/Server?

by Turritopsis Dohrnii Teo En Ming

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 -----END EMAIL SIGNATURE-----

3 years, 4 months

How to msg_apply_changes in a branch route?

by Anthony Alba

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

3 years, 4 months

vscode-kamailio-syntax - issue with empty comment line

by Chaigneau, Nicolas

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.

3 years, 4 months

Re: [SR-Users] [sr-dev] kamailio SIP and RTP proxy

by Henning Westerholt

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

3 years, 4 months

Kamailio Failover

by Rolly Casuga Jr.

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

3 years, 4 months

Kamailio behind HAProxy

by Joey Golan

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.

3 years, 4 months

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sr-users December 2020