May 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

isfocus parameter in Contact header
by Ramesh Kandasamy 06 May '19

06 May '19

Hi all, I am relatively new to Freeswitch and I am looking for a way to add isfocus parameter in the Contact header of 200 OK (INVITE response) from the freeswitch. I found channel variable sip_contact_params that can be set in dialplan as below, but it doesnt seem to help. Am I missing something or doing something wrong here? <action application="answer"/> *<action application="set" data="sip_contact_params=isfocus"/>* <action application="conference" data="..." Please advise. Thanks Ramesh

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Wiki is MUCH faster after the down day
by mario_fs 06 May '19

06 May '19

I didn’t see a comment here about the wiki update but after the down day I noticed the wiki web site averages triple it’s previous speed here on the west coast. Whatever was done worked very well. Mario G

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Abuse of files.freeswitch.org
by Brian West 06 May '19

06 May '19

FreeSWITCHers, I have found over 2000 IP Addresses that are abusing the files repo, If you happen to be one that has been blocked please email me directly to resolve your access. This isn't about resources, it isn't about us needing mirrors, its blatant abuse. The few that have come forward and resolved their issues had no idea they were doing this, They have various scripts that have gone off the rails and they are investigating and resolving the issue. AKA When Automation Goes Wrong! Thanks, Brian -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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'To' number modifcation
by devang nathwani 06 May '19

06 May '19

Hello Team, I have idea to modify from user, contact user from dialplan variable as well as channel variables but can we modify to number by any variables? The scenario is like this, My switch's device registered as trunk to destination switch, which is having dynamic ip(not having fqdn) So I am routing the DID number from my switch to that device which is registered as trunk to destination switch, but I need that device as bridge only to connect my switch with destination switch and send the call to another destination number. How can I achieve this by any variables? Thanks in advance.

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Firewall mysteriously starts blocking calls to port 5060
by Chad Phillips 05 May '19

05 May '19

Recently I reconfigured my firewall (via Shorewall) to block all inbound traffic to port 5060, except for whitelisted IP addresses from my inbound DID providers. After setup, we ran tests and everything worked fine for all incoming calls across all providers. Then a few hours later, calls from one of our providers started being blocked. All calls from our other providers continued coming through fine. Upon restarting our firewall service, the blocked calls from the single provider started coming through again. Between our successful tests and the start of the issue, there were zero changes made to the server. So why would my firewall suddenly start blocking inbound traffic from a whitelisted IP that it was previously letting through??

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what's the difference between bridge() and session.execute('bridge') in javascript
by Sanfy D 04 May '19

04 May '19

hi all, I write a javascript to bridge two endpoints. The senes is like the following, A(web+sip.js+wrtc) <--->Freeswitch(sofia+javascript) <---> Twilio sip trunking <---> B(pstn phone). I tried two methods in Freeswitch javascript to call from A to B: 1. create new sessionB then bridge A to B: (1) sessionA.answer() and ring back to A, (2) create new session to B via sip bridge, then break sessionA ringback and park sessionB, (3) bridge(sessionA, sessionB) 2. bridge A to B directly without create sessionB: (1) sessionA.answer(), (2) sessionA.execute('bridge','sofia/gateway/32d1440c-d13c-44b3-94da-1589daf6e81b/'+sessionA.getVariable('callee_id_number')); The second one works well. But in the first one, A always loses some audio from B. I use tcpdump on my Freeswitch server to check the connection and find that some UDP packets from Freeswitch to A are lost, but the packets from Twillo to Freeswitch are not lose. It seems that some UDP packet are lost while the Freeswitch relays the udp packets to A. So, what the difference is between the two methods? And how can I fix this problem? I need to use the first method to make the call. Thank!

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issue with Ring Groups playing a message to B-Leg only
by kaleem rehman 04 May '19

04 May '19

Hi All, i have an incoming SIP number, i want callee to hear a message on pick up to tell them what line call is on. on the incoming call if i bridge it to extension directly i can achieve this successfully, method i used is as below; <action application="set" data="group_confirm_file=lua streamfile.lua 16mit-pcm-signed.wav"/> <action application="set" data="group_confirm_key=exec"/> then i use transfer method <action application="transfer" data="2000 XML test.com"/> where 2000 is extension number when i change 2000 to 1000 which is RING GROUP and ring group rings ext 2000 i dont hear the message. i have added these two line to ring group as well (xml below for ring group) but i do not hear the audio. <extension name="ring group" continue="" uuid="5b0eaa67-d282-4299-8cc7-e696e3599b24"> <condition field="destination_number" expression="^1000$"> <action application="ring_ready" data=""/> <action application="set" data="ring_group_uuid=27cd2bf7-0f1e-4546-938e-536afd1a34c8"/> <action application="set" data="group_confirm_file=lua streamfile.lua 16mit-pcm-signed.wav"/> <action application="set" data="group_confirm_key=exec"/> <action application="lua" data="app.lua ring_groups"/> </condition> </extension> regards, Kaleem

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Building on win 10 using visual studio community edition failse
by Babak Yakhchali 04 May '19

04 May '19

Hi I'm trying to build fs 1.8.5 using vs community edition. Is it possible or I need other editions of vs? build fails with: C:\Program Files (x86)\Microsoft Visual Studio\2017\Community\Common7\IDE\VC\VCTargets\Microsoft.CppCommon.targets(209,5): error MSB6006: "cmd.exe" exited with code -1073741515 Done building project "libvpx.2017.vcxproj" -- FAILED tried with both win 10 sdk 10.0.17763.0 and 8.1 thanks

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Common Variable for CDR in Call Transfers
by Chhorm Chhatra 04 May '19

04 May '19

Hi, If you do not mind, may I know is there any way to group CDR records together? Currently, I would like to associate all CDR records made by Leg A regardless of how many time it has been transferred. Is there any common variable, tool or API to do this? Any help would be greatly appreciated. Thank you. Best Regards Chhorm Chhatra

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Conference RTP issue
by Sergey Safarov 03 May '19

03 May '19

Hi all I have broken audio for conference call for first 4-7 seconds of call. That is paging call using FS conference. Caller make call to conference and then FS conference dials about 28 phones with "Alert-Info: <intercom>" header (auto answer for Yealink phones). If you look into RTP stats using Wireshark, then you can see high jitter value for RTP stream from FreeSwitch. Oh jitter graph value is variable. Is anybody observe similar issue? How i can troubleshoot high jitter issue on FS side? call example is attached

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