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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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Call Drop ( we think only when originated by Mitel phone)
by Sean Devoy 30 Apr '19

30 Apr '19

FreeSWITCH Version 1.8.5-6-31281a0bf1~64bit (-6-31281a0bf1 64bit) The server is offsite and not NATed, the phones are NATed. My customer got some Mitel 6867i phones. They are working fine 95% of the time. We have narrowed it down we think but this may be anecdotal based on limited data. It appears to only happen when a call is originated from these Mitel phones. Cisco phones at the same site are rock solid. The customer reports "dropped calls". I got a Mitel 6867i and tested. After 44 test calls, I got a dropped call within seconds of being answered. Another instance where user actually gave me enough information to track it down showed the exact same thing - hangup after 7 seconds. Here is a log extract at the point where I answer the test call. sofia/external/4104440002 is the Mitel originating phone. 15:52:04.041308 [NOTICE] sofia.c:8429 Channel [sofia/internal/314] has been answered 15:52:04.041308 [DEBUG] switch_channel.c:3781 (sofia/internal/314) Callstate Change EARLY -> ACTIVE 15:52:04.041308 [DEBUG] switch_core_codec.c:248 sofia/external/4104440002(a)66.241.96.237 Restore previous codec PCMU:0. 15:52:04.041308 [DEBUG] switch_ivr_originate.c:3798 Originate Resulted in Success: [sofia/internal/314] 15:52:04.041308 [DEBUG] switch_channel.c:2047 (sofia/external/4104440002(a)66.241.96.237) Callstate Change RING_WAIT -> ACTIVE 15:52:04.041308 [DEBUG] switch_ivr_bridge.c:1782 (sofia/internal/314) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 15:52:04.041308 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/314) Running State Change CS_EXCHANGE_MEDIA (Cur 4 Tot 1109) 15:52:04.041308 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/314) State EXCHANGE_MEDIA 15:52:04.041308 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA 15:52:04.101306 [DEBUG] switch_rtp.c:7550 Correct audio ip/port confirmed. 15:52:07.501320 [NOTICE] sofia.c:1079 Hangup sofia/external/4104440002(a)66.241.96.237 [CS_EXECUTE] [NORMAL_CLEARING] 15:52:07.501320 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/external/4104440002(a)66.241.96.237] 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:825 sofia/external/4104440002(a)66.241.96.237 ending bridge by request from write function 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:917 BRIDGE THREAD DONE [sofia/internal/314] 15:52:07.521295 [NOTICE] switch_ivr_bridge.c:1034 Hangup sofia/internal/314 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 15:52:07.521295 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/314) State EXCHANGE_MEDIA going to sleep 15:52:07.521295 [DEBUG] switch_core_state_machine.c:584 (sofia/internal/314) Running State Change CS_HANGUP (Cur 4 Tot 1109) 15:52:07.521295 [DEBUG] switch_core_state_machine.c:847 (sofia/internal/314) Callstate Change ACTIVE -> HANGUP 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/314 skip receive message [UNBRIDGE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_ivr_bridge.c:1884 sofia/external/4104440002(a)66.241.96.237 skip receive message [UNBRIDGE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_core_state_machine.c:849 (sofia/internal/314) State HANGUP 15:52:07.521295 [DEBUG] mod_sofia.c:443 sofia/internal/314 Overriding SIP cause 480 with 200 from the other leg 15:52:07.521295 [DEBUG] mod_sofia.c:449 Channel sofia/internal/314 hanging up, cause: NORMAL_CLEARING 15:52:07.521295 [DEBUG] switch_core_session.c:2887 sofia/external/4104440002(a)66.241.96.237 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 15:52:07.521295 [DEBUG] switch_core_session.c:2887 sofia/external/4104440002(a)66.241.96.237 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Any ideas why this is happening? Anyone know any secrets about Mitel phones other then use TCP instead of UDP? Any ideas on how to debug this further since it happens so rarely? Thanks, Sean

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execute_on_answer
by Joli Martinez 30 Apr '19

30 Apr '19

Hello, I am trying to find more info on the application "execute_on_answer". I have a script that calls an IVR then dials a users extension. I need to wait for the calling party extension to answer before executing the rest of my script. I don't seem to find much documentation on this application. Is there another way to test if the calling party has been answered? thanks,

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dynamic payload negotiation
by Babak Yakhchali 30 Apr '19

30 Apr '19

Hi I'm trying to make calls between a custom softphone as client and freeswitch using opus as codec. when client calls freeswitch everything is working fine. but when I originate a call from freeswitch there is no audio from sofphone but client hears fs. After capturing packets I see that when client sends invite it is using 107 as its payload type and freeswitch sends 107 in reply. But when fs sends invite it is using 102 and client is sending 107. What is the standard? I think the softphone is doing something wrong because testing with jitsi works fine. thanks

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