- FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon Hotel is filling up! ACT NOW and save $300
by Anthony Minessale 17 Jul '17

17 Jul '17

Use the code CCJulySM2017 and save $300 Help support the FS Community and have fun in the process! -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Re: [Freeswitch-users] RTP Auto Switch behaviour on EC2
by Sean Ingham 17 Jul '17

17 Jul '17

Hi Brian, That makes sense since the EC2 has no network interface for the public IP that FS can bind to. So what is RTP auto switching actually doing when it says: Auto Changing audio port from 172.31.x.x:31196 to 52.17.x.x:31196 Is it just amending udp packet headers but not actually rerouting anything? Thanks, Sean >"Freeswitch always sends RTP traffic from it's private IP", Because thats >all it CAN do, the public IP isn't directly bound on the system so it >can't send from the public IP. >/b On Mon, Jul 17, 2017 at 4:33 AM, Sean Ingham <sean at missionlabs.co.uk <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >* Hi, *>* I'm running FreeSwitch on an AWS EC2 with config as per the wiki EC2 page. *>>* external_rtp_ip & bind_server_ip are both set to the box's public AWS *>* Elastic IP - as is ext_rtp_ip on all profiles. *>>* I'm running a WebRTC gateway also in AWS. I can see that SDPs from *>* Freeswitch to the gateway all include Freeswitch's public IP as desired, *>* however when the call is answered Freeswitch always sends RTP traffic from *>* it's private IP. Then a second or 2 later Freeswitch sees incoming RTP from *>* the gateway to it's public IP address and RTP auto swtiching behaviour *>* kicks in, resulting in a line like this in FS logs: *>>* switch_rtp.c:6954 Auto Changing audio port from 172.31.x.x:31196 to *>* 52.17.x.x:31196 *>>* As it happens, the gateway is able to receive media from either private or *>* public IP as the box is in the same subnet, but in around 30% of cases at *>* the point this switching occurs the call audio drops. (I'm guessing the *>* gateway has issues stitching together the rtp streams from 2 different *>* sources.) *>>* I've tried experimenting with the disable_rtp_auto_switch parameter, but *>* seems to me a cleaner solution would be to have Freeswitch send RTP from *>* it's public IP in the first place, and I can't understand why it's not *>* doing that given it's negotiated to use the public IP in it's SDP. *>>* Can anyone provide any explanation for the behaviour I'm currently seeing, *>* or suggest how I can get FS to set the initial outgoing RTP port correctly? *>>>* Thanks, *>* Sean. *>>* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> *>>* Official FreeSWITCH Sites *>* http://www.freeswitch.org <http://www.freeswitch.org/> *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> *>* http://www.cluecon.com <http://www.cluecon.com/> *>>* FreeSWITCH-users mailing list *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> *>* http://www.freeswitch.org <http://www.freeswitch.org/> *> -- *Brian West*brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com Book a phone call (CST) <https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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RTP Auto Switch behaviour on EC2
by Sean Ingham 17 Jul '17

17 Jul '17

Hi, I'm running FreeSwitch on an AWS EC2 with config as per the wiki EC2 page. external_rtp_ip & bind_server_ip are both set to the box's public AWS Elastic IP - as is ext_rtp_ip on all profiles. I'm running a WebRTC gateway also in AWS. I can see that SDPs from Freeswitch to the gateway all include Freeswitch's public IP as desired, however when the call is answered Freeswitch always sends RTP traffic from it's private IP. Then a second or 2 later Freeswitch sees incoming RTP from the gateway to it's public IP address and RTP auto swtiching behaviour kicks in, resulting in a line like this in FS logs: switch_rtp.c:6954 Auto Changing audio port from 172.31.x.x:31196 to 52.17.x.x:31196 As it happens, the gateway is able to receive media from either private or public IP as the box is in the same subnet, but in around 30% of cases at the point this switching occurs the call audio drops. (I'm guessing the gateway has issues stitching together the rtp streams from 2 different sources.) I've tried experimenting with the disable_rtp_auto_switch parameter, but seems to me a cleaner solution would be to have Freeswitch send RTP from it's public IP in the first place, and I can't understand why it's not doing that given it's negotiated to use the public IP in it's SDP. Can anyone provide any explanation for the behaviour I'm currently seeing, or suggest how I can get FS to set the initial outgoing RTP port correctly? Thanks, Sean.

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real time configuration management (extensions)
by robert mundkowsky 17 Jul '17

17 Jul '17

What are the best practice for real-time configurationmanagement of extensions in a live FreeSWTICH system without downtime? We currently use FreeSWITCH XML configuration files andreload the dial plan via fs_cli. Would using a database make configurations more dynamic? If we move to using a database, I am guessing PostgreSQLis the way to go? Or is MySQL possible? Robert

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Can't connect to freeswitch.org, any issues??
by Joel Serrano 16 Jul '17

16 Jul '17

Hi guys, Don't know if it's just me, but I cannot reach freeswitch.org via ipv4/ipv6 or http/https. (I tried from multiple different sources, all fail) Is there any maintenance or anything going on? Thanks! Joel.

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mod_sms: is possible to not forward message?
by Sergey Safarov 16 Jul '17

16 Jul '17

I created chatplan <?xml version="1.0" encoding="utf-8"?> <include> <context name="public"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="stop" data=""/> </condition> </extension> </context> </include> Required simple accept message and do nothing. But really FreeSwitch send forward message. Tested current master and 1.5 Is it bug? Or i do some thing wrong?

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how to send simpe message
by Sergey Safarov 15 Jul '17

15 Jul '17

Could you advice how to send simple message? As I knows this may be done only from lua or esl script https://freeswitch.org/confluence/display/FREESWITCH/mod_sms#mod_sms-Sendin… But provided example can send only to local endpoin When i try send message to remote host thne got error like this 2017-07-15 08:59:46.408247 [INFO] switch_cpp.cpp:1377 chat console 2017-07-15 08:59:46.408247 [ERR] sofia_presence.c:200 Chat proto [sip] from [sip:1004@my.local.host] to [1019(a)my.remote.host] Hello from Seven Du! Have fun! Invalid Profile my.remote.host Please advice how to specify profile name in event?

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FreeSWITCH HA and BDR, wiki
by Matthew Nichols 15 Jul '17

15 Jul '17

After going through https://freeswitch.org/confluence/display/FREESWITCH/High+Availability, it appears out of date/misleading. Here are the things I have found: Switchname needs to be different or they end up clobbering each other. Call recovery still works. $${domain} needs to be set the same on both boxes for voicemails and registrations to work well after recovery. As far as I can tell the entirety of /var/lib/freeswitch can be shared with something like GlusterFS (assuming sqlite is not being used). Especially with PostgreSQL BDR auto-create-sql needs to be false and auto-clear-sql needs to be true, otherwise information is never cleared after a crash. Also I see no documentation on the wiki on actually using FreeSWITCH with PostgreSQL BDR. I would be willing to start a page with what I have discovered so far. The only online documentation I have found is https://gist.github.com/DigiDaz/1cfe3d5d32080a8e3d75a20bb5bc4fb5, which doesn't have the primary keys set up quite right (calls and channels need to add hostname to the primary key or call recovery doesn't work quite right) and doesn't include voicemail. I am very willing to setup the required edits but I don't have write access so I would need to go through one of you who does. Matthew Nichols

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Channel Var Codec
by Andre DeMattia 15 Jul '17

15 Jul '17

Hi what Channel variable should I use to see what Codec the A leg allows? variable_rtp_use_codec_string or variable_ep_codec_string or something else? In the CDR I see this <rtp_use_codec_string>G722,PCMU,PCMA,GSM</rtp_use_codec_string> This looks like what I'm looking for but I need it in the dialplan before the call goes out so I can direct the call where I need it to go. Thanks

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Are you planning on creating a "1.6 -> 1.8" upgrade notes document?
by Nathan Neulinger 14 Jul '17

14 Jul '17

I remember moving from 1.4 to 1.6 there were several breaking behavior changes. Are you planning on doing anything to document specific known "you have to change X to Y" and "this behavior has changed" sort of document for sites upgrading from 1.6.x to 1.8.x? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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