July 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

test
by bob. chen 13 Jul '20

13 Jul '20

发送自 Windows 10 版邮件<https://go.microsoft.com/fwlink/?LinkId=550986>应用

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test
by bob. chen 13 Jul '20

13 Jul '20

发送自 Windows 10 版邮件<https://go.microsoft.com/fwlink/?LinkId=550986>应用

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test
by bob. chen 11 Jul '20

11 Jul '20

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Oneway Audio, can’t send RTP back to UAC
by Muhammad Naseer Bhatti 11 Jul '20

11 Jul '20

Hi, I can’t seem to be able to figure out why I can’t send RTP to the other side (UAC) of the switch (One way audio?) . I have FreeSWITCH Version 1.10.4-dev git 00113c4 with vanilla config (for testing) on Public IP address and the other switch is also on Public IP on same subnet. I receive call from SippySoft (UAC) and playing delay_echo application. The call flow is SIP UA (NATted) -> SippySwitch (Public IP) - FreSWITCH (Public IP) FreeSWITCH after answering the call says 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMA:8:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5594 Audio Codec Compare [PCMU:0:8000:40:64000:1]/[PCMU:0:8000:20:64000:1] 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5630 Audio Codec Compare [PCMU:0:8000:20:64000:1] is saved as a near-match 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:5701 Substituting codec PCMU@40i@8000h@1c 2020-07-09 21:48:37.902333 [DEBUG] switch_core_media.c:3839 Set Codec sofia/internal/04232152273(a)43.225.99.130 PCMU/8000 40 ms 320 samples 64000 bits 1 channels *2020-07-09 21:48:37.902333 [DEBUG] switch_core_codec.c:111 sofia/internal/04232152273(a)43.225.99.130 <04232152273(a)43.225.99.130> Original read codec set to PCMU:0* and then starts delay_echo() app. Isn’t Original read codec set to PCMU:0 is a bad thing? Perhaps the reason not able to send RTP back? On the other hand, I installed Asterisk 13.34.0, on the same machine, just to prove the point if there is network issue but things work just fine with Asterisk default config. Seem like there is either something not configured (default) in FreeSWITCH and I can’t seem to be able to find either. SDP in Asterisk and FreeSWITCH both seems to be the same. FreeSWITCH SIP Trace is here https://pastebin.freeswitch.org/view/0925119c and console log is here https://pastebin.freeswitch.org/view/69b5f68c Asterisk SIP Trace is here https://pastebin.freeswitch.org/view/f38abc97 Appreciate some input to figure out this problem. Thanks, Naseer

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Attended Transfer not routing to follow me
by Jason Benedict Low 11 Jul '20

11 Jul '20

Dear list, excuse me if the questions are already answered. I tried searching the list and google but i have no actual answer. I hope for your kind help here. Here my setup in freeswitch testing: 1. set up extension 2000 + follow-me (to external mobile nos 12345678) 2. set up extension 2001 + follow-me (to external mobile nos 87654321) 3. set up dialplan bind_meta_app - data = "4 b s execute_extension::att_xfer XML ${domain_name}" 4. set up dialplan att_xfer data = "user/${digits}@${context}" 5. set up DDI number 6512 3456 incoming dialplan transfer to extension 2000 data = "2000 XML ${somain_name}" what is working: 1. When the caller makes a call to DDI number 6512 3456, incoming call to DDI number 6512 3456 able to route to extension 2000 and ringing, also follow-me to external mobile nos 12345678 is ringing. 2. Both the extension and external mobile can answer calls. what not working: 1. When in-call, they tried to do an attended transfer by press *42001# - where 2001 is the extension 2001. 2. only extension 2001 is ringing but the follow-me seem to be not able to execute to ring to external mobile nos 87654321. any advice or suggestions how i can go about ? Thanks in advance. Have a great weekend to come! Best Regards, Jason *** Reclaim Your Freedom, use Free & Open Source Software - Be Open Standard Compliance. ***

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att_xfer and extension with follow-me
by Jason Benedict Low 10 Jul '20

10 Jul '20

Dear list, excuse me if the questions are already answered. I tried searching the list and google but i have no actual answer. I hope for your kind help here. Here my setup in freeswitch testing: 1. set up extension 2000 + follow-me (to external mobile nos 12345678) 2. set up extension 2001 + follow-me (to external mobile nos 87654321) 3. set up dialplan bind_meta_app - data = "4 b s execute_extension::att_xfer XML ${domain_name}" 4. set up dialplan att_xfer data = "user/${digits}@${context}" 5. set up DDI number 6512 3456 incoming dialplan transfer to extension 2000 data = "2000 XML ${somain_name}" what is working: 1. When the caller makes a call to DDI number 6512 3456, incoming call to DDI number 6512 3456 able to route to extension 2000 and ringing, also follow-me to external mobile nos 12345678 is ringing. 2. Both the extension and external mobile can answer calls. what not working: 1. When in-call, they tried to do an attended transfer by press *42001# - where 2001 is the extension 2001. 2. only extension 2001 is ringing but the follow-me seem to be not able to execute to ring to external mobile nos 87654321. any advice or suggestions how i can go about ? Thanks in advance. Have a great weekend to come! Best Regards, Jason *** Reclaim Your Freedom, use Free & Open Source Software - Be Open Standard Compliance. ***

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freeswitch databases
by Kakiman Expert 09 Jul '20

09 Jul '20

Hello, i'm using Freeswitch 1.10 and need some elements to understand DB. I have configured FS to use postgresql DB for "freeswitch database". In this DB I only see live events (calls, sessions, registrations...) and not past events. But with "fs_cli sofia status profile" command, I can see past events like calls-in, calls-out etc... So, where are these informations stored ? Is there other DB than "freeswitch" DB ? thanks regards Joe

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phrases with parameter # are no longer working
by John Covici 09 Jul '20

09 Jul '20

I am not sure when this started, but when a # parameter is encountered in a phrase I get the following in the log 2020-07-08 20:37:13.270216 [ERR] switch_ivr_play_say.c:292 Invalid SAY Interface [en]! . How can I fix this -- it used to work, not sure when, however. I am using v 1.10.3. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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Pre-announcement script
by Raúl Alexis Betancor Santana 07 Jul '20

07 Jul '20

Is there any sample dialplan to play an announcement to the destination of a call before bridge the call to the caller ? I've been looking at the fifo app, but that required the callee to call a number a be "waiting" ... what I am looking for is to play a record to the callee, then a pause and then bridge the incoming caller. Best regards

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mobile sip app getting strange notifications from fs
by John Covici 07 Jul '20

07 Jul '20

I have a freeswitch extension which I have the client for on my iphone. The client is Bria Mobile and I get notifications -- apparently from fs -- that say you have 0 new voicemail messages. How can I stop this from happening? I am using fs 1.10.3. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una covici(a)ccs.covici.com

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