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[ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
by Giovanni Maruzzelli 22 Apr '21

22 Apr '21

Hello fellow VoIPers and RTCers, on GitHub there is an early release of sipnagios, opensource. check it out: https://github.com/gmaruzz/sipnagios sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values. mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!) Enjoy! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Problem getting custom voicemail script to run
by Alex Lake 19 Apr '21

19 Apr '21

FreeSWITCH Version 1.10.6-release-18 running on Debian 10. I need voicemail to run an alternative script when a message is left, and not just send an email. I've changed /etc/freeswitch/autoload_configs/switch.conf.xml to have: <param name="mailer-app" value="/usr/bin/php /usr/local/freeswitch/scripts/processvm.php"/> <param name="mailer-app-args" value=""/> But I don't believe it's even trying to use it. Instead I get entries in the log like this: 2021-04-19 16:49:04.272054 [DEBUG] mod_voicemail.c:2832 Deliver VM to 6378101(a)w.x.y.z 2021-04-19 16:49:04.872034 [DEBUG] switch_utils.c:1308 Emailed file [/tmp/mail.1618850944b82e] to [email(a)domain.com] 2021-04-19 16:49:04.872034 [DEBUG] mod_voicemail.c:3042 Sending message to email(a)domain.com What am I missing?! Can this behaviour be overridden?

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Lua API call delays the packet sending out
by VERMA Vineet (BCS/SPS) 19 Apr '21

19 Apr '21

Dears, I am facing the issue that when REST API triggers the freeswitch I am sending the reply with Following code: stream:write("HTTP/1.1 200 OK\r\n") if (accept == "application/json" or accept == "*/*") then stream:write("Content-Type: application/json\r\n") else stream:write("Content-Type: text/plain\r\n") end stream:write("\r\n") but it seems before packet it out , there is as deplay of 2-3 sec . can you please suggest how to solve this? I already tried stream:shutdown(); and stream:close() but did not helped. Thanks & Regards Vineet Verma This e-mail cannot be used for other purposes than BICS business use.

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Why doesn't this call get answered?
by Steven Schoch 17 Apr '21

17 Apr '21

This office has a bunch of Polycom SoundPoint IP 320 phones, and a single Grandstream HT801 (connected to a FAX machine for outgoing FAXes). I can make a call from the phones to outside numbers. I can make a call from the HT801 to local phones. But I can't call from the HT801 to outside numbers. The last important thing that happens in the failed call is this: 2021-04-13 17:28:15.260844 [DEBUG] sofia.c:7406 Channel sofia/external/<number> entering state [calling][0] The difference between the work and not work seems to be this: When I call from a phone to an outside number, it does this: Local SDP: v=0 o=FreeSWITCH 1618335260 1618335261 IN IP4 <external-IP> s=FreeSWITCH c=IN IP4 <external-IP> t=0 0 m=audio 25104 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv When it tries to call from the HT801 to an outside number, it does this: Local SDP: v=0 o=FreeSWITCH 1618327543 1618327544 IN IP4 <external-IP> s=FreeSWITCH c=IN IP4 <external-IP> t=0 0 m=audio 32552 RTP/AVP 0 8 102 9 101 103 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:103 telephone-event/48000 a=fmtp:103 0-16 a=ptime:20 a=sendrecv Is that why it doesn't answer? If so, how do I change it? I should mention that when I tried this at home, it worked, but when I attempted to install it here at the bookstore, it didn't. The Comcast router at my home is a little different; they use a Netgear router here; and I may have upgraded Freeswitch between the time it worked and now. -- Steve

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Send_Info not supported
by Ryan Delgrosso 15 Apr '21

15 Apr '21

Hi All, When using the uuid_send_info or uuid_send_message methods to send in-dialog messaging I get an OK then a message that the method is not supported. Example: freeswitch(a)fs9.domain.com> uuid_send_info fe044b54-f880-448a-89dc-c7d4db2d2399 test +OK Success 2021-04-15 15:52:33.282468 [DEBUG] mod_sofia.c:1908 sofia/webrtc/15729(a)domain.com send_info is not supported. The invite that established the session has the following allow/supported headers: Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound What am i missing here? What is Sofia looking for to permit this? Thanks much -Ryan

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Using both builtin and online grammars with play_and_detect_speech
by mayamatakeshi 12 Apr '21

12 Apr '21

I know it is possible to start play_and_detect_speech using multiple builtin grammars like this: builtin:speech/transcribe\nbuiltin:dtmf/digits And I know it is possible to specify an inline grammar like this (for hints): inline:<speech-context>...</speech-context> But would it be possible to use both a builtin grammar and an inline grammar? I tried like this but it didn't work: builtin:dtmf/digits\ninline:<speech-context>...</speech-context> Was anyone able to use something like this?

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Pathing Problem?
by William Colls 11 Apr '21

11 Apr '21

I am trying to build a the latest production version from source. Everything looks OK when I I do the bootstrap/configure/make/make install for freeswitch, sandsp, and sofia-sip sequence, but when I try to start freeswitch I get an error: error while loading shared libraries: libspandsp.so.3: cannot open shared object file: No such file or directory The file does exist in /usr/local/lib., owned by freeswitch:freeswitch as a link to libspandsp.so.3.0.0 also owned by freeswitch:freeswitch. Building on a Pine64-LTS, with Arbian's implementation of Debian Buster. Any suggestions greatly appreciated. Thanks for your time. William

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Send call to different place on busy vs no answer
by Steven Schoch 09 Apr '21

09 Apr '21

I'm setting up a pbx. Incoming calls to the main number will ring on 3 "front desk" phones. If all phones are busy, I want to play a message saying: "All operators are busy. Please call back in a few minutes or send email." However, if the call rings for 20 seconds with no answer, it should transfer to voicemail. My dialplan has: <action application="set" data="continue_on_fail=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="${group_call(operator@${domain_name})}"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="???"/> The last application should be to do one thing (play a message) if the bridge failed because all phones were busy, and do something else (transfer to voicemail) if there was no answer after 20 seconds. Is this something that can be done in the dialplan? If not, do I use a script? -- Steve

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FreeSWITCH didn't send any rtp packages to callee in ipv6 calls
by 王聡 09 Apr '21

09 Apr '21

Hi all, We’re preparing to upgrade our fs server for ipv6 supporting, but it is not going well on test. The test proceeded on FreeSWITCH 1.10.6 and LinphoneSDK 4.5.0, under IPv6Only networks. As the result, the callee couldn’t hear any voice from caller, while caller works fine. The tcpdump showed that freeswitch didn’t send any rtp packages to callee, like this: [2001::14] is FreeSWITCH, [240d::68ae] is the caller, and [240d::b342] is the callee, which missed the rtp stream from server. We tried to exchange the device of caller and callee, but result didn’t change (callee couldn’t hear any voice from caller). FreeSWITCH Console also showed no error and RTP port is set correctly. Environment: FreeSWITCH Version 1.10.6-release+git~20210325T131609Z~1ff9d0a60e~64bit (git 1ff9d0a 2021-03-25 13:16:09Z 64bit) Linphone iOS 4.4.0 with Linphone SDK 4.5.0 / iOS 13.7 Any suggestion would be appreciated. Regards. —————————————————— 王聡 cong.wang.itsherpa(a)gmail.com 〒810−0073 福岡市中央区舞鶴2-3-6 赤坂プライムビル 2F 株式会社アイティーシェルパ

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Anyone else get junked email this week?
by mario_fs 07 Apr '21

07 Apr '21

Just curious, did anyone else notice flood of junk mails to your address used for the freeswitch mailing list? My email is ONLY used for this emailing list and nowhere else, it is secure on a Mac. None other of my 50+ email addresses get junk. For the second time, I got 9 junk emails to this address today. I am just wondering if anyone else gets these. Yes you can setup and filter only for this list, but then people on this list I who may want to contact me can’t if there not on a list. Mario G

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