May 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Transcoding / Call-Hold issue
by Carsten Bock 08 May '20

08 May '20

Hi, I am running a FreeSwitch 1.10 (built Debian Packages from the GIT-Branch due to some codec issues; AMR/AMR-WB is only available, if the respective libraries are present on the system). I am using FreeSwitch in the most simple configuration: FreeSwitch is acting as a (transcoding) B2BUA with only a single peer (Kamailio, which does the further distribution, Sofia Config attached), so my dialplan is straight forward and contains a single action: <action application="bridge" data="sofia/internal/${destination_number}@ ${loadbalancer}"/> Here is, what I've noticed and where I am unsure, how to fix this: 1) Call from A is sent to FreeSwitch, with Codecs AMR and AMR-WB 2) Call is leaving FreeSwitch towards B, offering Codecs AMR, AMR-WB, G722, G711, ... B-Party is selecting G722, as it does not support AMR or AMR-WB, 3) Call is connected to A on AMR/AMR-WB and to B using G722 - so far, everything is fine - Now the following happens: 4) A puts the call on hold, so a INVITE with according SDP is sent to FreeSwitch 5) FreeSwitch itself sends an INVITE to B indicating the recognized "Hold" with according SDP, however now this INVITE only contains AMR-WB, which was initially offered to B, but which was never responded or even indicated support for (!?!??!) I can hardly believe, I am the first one to see this. Does anyone know where or what to look for and what might have gone wrong here? Thanks, Carsten P.S.: The first mail didn't go through, as I added my Sofia Config.... without any notice. -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

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attendet transfer by REFER
by Yuriy Nasida 08 May '20

08 May '20

Hi guys, My callflow looks like in RFC doc (please look at screen). Transferee and Transfer Target stays behing FS. Should I do something special at FS side to implement such attendet transfer in case Transferor use re-invite/refer method like RFC describes ? I supposed - not, but weird things happens in the moment FS get REFER. I expected that FS will just connect Transferee and Transfer Target but... FS start new call (3th) instead of this.. I mean new call to connect Transferee and Transfer Target. But... They are at hold already and I do not need new call... I just need to connect them. In theory I can use lua and try to bridge these two sessions between each other but I am not sure it is correct way. Also I know about att_xfer feature but I can't use that because it require dtmf method. Please advice [image: Selection_972.png]

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English Sounds set, Male voice, provided Amazon's Polly TTS + Generator
by Avi Marcus 07 May '20

07 May '20

I've generated a full set of US-English sounds using the voice Matthew, with the fancy "neural" setting, for 16000 and 8000. https://github.com/avimar/freeswitch-sounds-polly has install instructions. Once you switch to a polly voice, you can use this script to generate new prompts that instantly match the rest of your prompts. I've tested with mod_say_en and it sounds decent. I think neural is optimized for actual sentences, and not single words. So perhaps digits and letters are best replaced with the more uniform Matthew-standard (included). I'll try to test it more another time. Feedback welcome. If people want 24000 I can generate as ogg and then convert with sox, I guess? Polly doesn't support above 24000hz. The script can read a .xml file but it's currently set a) to only do one section at a time and b) to request pcm at 16000 and use sox to downsample to 8000. If someone wants to create other sets, you can grab the phrases file from https://github.com/signalwire/freeswitch/tree/master/docs/phrase and pick a voice from https://docs.aws.amazon.com/polly/latest/dg/voicelist.html (you'll have to switch the XML/JSON importer, it defaults to english) -There's neural voices available for English (British) (en-GB), Portuguese (Brazilian) (pt-BR) (phrases <https://github.com/signalwire/freeswitch/blob/master/docs/phrase/phrase_pt_…>), and US Spanish (es-US). -There's regular voices available in lots of others languages, we have phrases for these: German (de-DE), French (fr-FR), Portuguese (European) (pt-PT), Russian (ru-RU), Chinese, Mandarin (cmn-CN). Email me or file an issue if you need help. -Avi Marcus BestFone

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Transcoding / Call-Hold issue
by Carsten Bock 05 May '20

05 May '20

Hi, I am running a FreeSwitch 1.10 (built Debian Packages from the GIT-Branch due to some codec issues; AMR/AMR-WB is only available, if the respective libraries are present on the system). I am using FreeSwitch in the most simple configuration: FreeSwitch is acting as a (transcoding) B2BUA with only a single peer (Kamailio, which does the further distribution, Sofia Config attached), so my dialplan is straight forward and contains a single action: <action application="bridge" data="sofia/internal/${destination_number}@ ${loadbalancer}"/> Here is, what I've noticed and where I am unsure, how to fix this: 1) Call from A is sent to FreeSwitch, with Codecs AMR and AMR-WB 2) Call is leaving FreeSwitch towards B, offering Codecs AMR, AMR-WB, G722, G711, ... B-Party is selecting G722, as it does not support AMR or AMR-WB, 3) Call is connected to A on AMR/AMR-WB and to B using G722 - so far, everything is fine - Now the following happens: 4) A puts the call on hold, so a INVITE with according SDP is sent to FreeSwitch 5) FreeSwitch itself sends an INVITE to B indicating the recognized "Hold" with according SDP, however now this INVITE only contains AMR-WB, which was initially offered to B, but which was never responded or even indicated support for (!?!??!) I can hardly believe, I am the first one to see this. Does anyone know where or what to look for and what might have gone wrong here? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 40 524 75 93-40 | M +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

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Re: [Freeswitch-users] Switching voice for mod_say_en per caller?
by Avi Marcus 04 May '20

04 May '20

UPDATE: For the heck of it, I tried excluding the sound-prefix="$${sound_prefix}" in en.xml. Everything seems to work just fine for both languages and no error. So... am I missing something? -Avi

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Switching voice for mod_say_en per caller?
by Avi Marcus 04 May '20

04 May '20

To switch the default sound path and audio for both playback and mod_say_X, it seems you modify 2 things: <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_language=en"/> But I'm trying to test a new english voice (amazon's neural Matthew, digits/ivr/currency available here <https://github.com/avimar/freeswitch-sounds-polly/>) Changing the sounds_prefix worked for the audio files, but not for mod_say_en. I found that if I modified /usr/share/freeswitch/lang/en/en.xml: <language name="en" say-module="en" sound-prefix="*$${sound_prefix}*" tts-engine="cepstral" tts-voice="callie"> ... to not evaluate on startup/original, but the current value, then it worked. <language name="en" say-module="en" sound-prefix="*${sound_prefix}*" tts-engine="cepstral" tts-voice="callie"> But now I'm getting this error anytime I use mod_say_en -- even though it's working. EXECUTE sofia/internal/1000@YYYY say(en NUMBER ITERATED 1000) 2020-05-04 20:19:28.963796 [CRIT] switch_channel.c:1436 Invalid data (${sound_prefix} contains a variable) Is this a mistaken error? Was I supposed to modify this somehow else? Thanks. -Avi Marcus

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mod call center tier behaviour
by abubacker 04 May '20

04 May '20

I am working in mod_callcenter Before my call center configuration is like this /referene : callcenter.conf.xml/ /<agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1003@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1003@default" queue="support@default" level="1" position="3"/>  <tier agent="1003@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers> </configuration> / Tier list : /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 +OK/ When I renamed the 1003@default as 1006@default / <agents> <agent name="1000@default" type="callback" contact="[call_timeout=10]user/1000" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1001@default" type="callback" contact="[call_timeout=10]user/1001" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> <agent name="1006@default" type="callback" contact="[call_timeout=10]user/1006" status="Available" max-no-answer="0" wrap-up-time="0" reject-delay-time="1" busy-delay-time="1" /> </agents> <tiers>  <tier agent="1000@default" queue="support@default" level="1" position="1"/> <tier agent="1001@default" queue="support@default" level="1" position="2"/> <tier agent="1006@default" queue="support@default" level="1" position="3"/>  <tier agent="1006@default" queue="secondary" level="1" position="3"/> <tier agent="1001@default" queue="secondary" level="1" position="2"/> <tier agent="1000@default" queue="secondary" level="1" position="1"/> </tiers>/ After that I have reload the mod_callcenter http://pastebin.freeswitch.org/14810 The 1003@default is not overwritten , but 1006@default has added newly why 1003@default is still exist /freeswitch@FMS-Queue> callcenter_config tier list support@default queue|agent|state|level|position support@default|1000@default|Ready|1|1 support@default|1001@default|Ready|1|2 support@default|1003@default|Ready|1|3 support@default|1006@default|Ready|1|3 +OK/ Is it a bug or Is it a default behaviour ? -- Best regards, N.Abubacker , Associate system engineer , bk systems pvt ltd , Ph : 9144-43902701 Disclaimer:http://www.bksystems.co.in/email-policy

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Proxy when SIP Request method is MESSAGE
by João Vitor Arruda 04 May '20

04 May '20

Hello, I currently have FS being used as a B2BUA for voice calls (INVITE SIP Request method) but I also need to process some MESSAGE type SIP requests. I'm wondering if there is a way to configure FS to completely proxy those messages.without acting like a B2BUA. In my initial try I got a "chatplan" configured to handle those messages with a lua script that use: freeswitch.Event("info", "SMS::SEND_MESSAGE"); This kinda working but the problem i'm having is that FS will reply "202 Accepted" to the original sender before it even forward it to the final destination which breaks our ability to really confirm that the message was accepted by the final destination.

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Proxy when SIP Request method is MESSAGE
by João Vitor Arruda 02 May '20

02 May '20

Hello, I currently have FS being used as a B2BUA for voice calls (INVITE SIP Request method) but I also need to process some MESSAGE type SIP requests. I'm wondering if there is a way to configure FS to completely proxy those messages.without acting like a B2BUA. In my initial try I got a "chatplan" configured to handle those messages with a lua script that use: freeswitch.Event("info", "SMS::SEND_MESSAGE"); This kinda working but the problem i'm having is that FS will reply "202 Accepted" to the original sender before it even forward it to the final destination which breaks our ability to really confirm that the message was accepted by the final destination. Joao Arruda

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Announcing SaraPhone, SIP WebRTC Open Source business phone
by Giovanni Maruzzelli 01 May '20

01 May '20

My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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