July 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Mod_Opus :Add fmtp maxplaybackrate=8000 in OPUS codec offer.Version 1.6.8-15-99de0ad~64bit
by Jonathan Hunter 12 Jul '17

12 Jul '17

Hi Guys, We are currently testing OPUS and Im looking to lock down things to a maxplaybackrate of 8000HZ. This works fine if FreeSWITCH is the receiver (Offer Answer), however if I then send an invite out from FreeSWITCH (bridge the same call)it doesnt seem to add the fmtp maxplaybackrate=8000 paramater. As when FreeSWITCH answers I see; Media Attribute (a): fmtp:111 useinbandfec=1; maxaveragebitrate=14400; maxplaybackrate=8000; minptime=10 However when I send an Invite offer out I see; Media Attribute (a): rtpmap:111 opus/48000/2 Media Attribute (a): fmtp:111 minptime=10;useinbandfec=1 The OPUS module document seems to suggest we can add this, and if so how can I achieve this, as I have my opus.conf set to; <param name="use-vbr" value="1"/> <param name="use-dtx" value="0"/> <param name="complexity" value="10"/> <param name="maxaveragebitrate" value="14400"/> <param name="maxplaybackrate" value="8000"/> <param name="packet-loss-percent" value="15"/> <param name="keep-fec-enabled" value="1"/> <param name="use-jb-lookahead" value="1"/> <param name="advertise-useinbandfec" value="1"/> And I also set the SIP profile codec list and absolute codec offer to OPUS@8000h, and as I say works great when we answer an offer, but not when we originate. Am I misconfiguring or is this not possible? I know the RFC states the parameter "maxplaybackrate" is a unidirectional receive-only parameter that reflects limitations of the local receiver, however mod_opus documentation seems to suggest we can set it in the offer, so I should be able to add as; maxplaybackrate=8000 Is this possible? I just want to ensure I get narrowband end to end. Many thanks Jon

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compile FS with special lib dir
by Madovsky 12 Jul '17

12 Jul '17

Hi, is there any way to compile freeswitch with a different library target path? for example, I would like to compile mod_av referring the libav libraries to a special folder (not /usr/local/lib) thanks Franck

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How to add path information to mod_sms message
by Keith Whyte 12 Jul '17

12 Jul '17

Hi! I'm registering clients to freeswitch via a Kamailio proxy. Kamailio sends a Path: header to freeswitch, and calls are working fine, but I'm struggling to get chats working. I'm stuck trying to figure out how to send the path information back to Kamailio in a MESSAGE with mod_sms. Adding a "path" header to the event is ignored. I have a python chatplan, I have found that if I do something like: event = ESL.ESLevent("CUSTOM", "SMS::SEND_MESSAGE") event.addHeader("to", "extention@kamailio_server;received=sip:ip_addr:port") Then Freeswitch will send the MESSAGE to the kamailio server with a To line: To: <sip:extention@kamailio_server>;received=sip:ip_addr:port I guess I could go about having Kamailio parse this, but is this the way to go? It seems messy. Any ideas? Thanks!

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Got 603 Decline from freeswitch
by devang nathwani 11 Jul '17

11 Jul '17

Hello, Below is the fs_cli log Asterisk 192.168.201.227 Opensips 192.168.201.245 Freeswitch 192.168.201.248 Here is the call flow, Asterisk -> Opensips -> Freeswitch https://pastebin.freeswitch.org/view/2cd0875b What could be the issue?

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uuid_dump in dialplan
by Daniel Greenwald 11 Jul '17

11 Jul '17

Hello, Looking for a way to dump all variables to the log at the end of the dialplan? I know the info application can query specific variables but is there a way to perform a full uuid_dump on the channel and send it to the log just before or after hangup?

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PHP code in voicemail.tpl
by Uday kumar 11 Jul '17

11 Jul '17

Hello All, I am trying to use Google Transcribing to Transcribing Voicemail. Is possible to add PHP code in voicemail.tpl or run php file like voicemail.php on place of voicemail.tpl? I tried to add PHP code in tpl file but didn't worked also tried with voicemail.php after changing in voicemail.conf.xml but didn't worked. I am using sendmail to send voicemail on email. Email going when extension getting voicemail with recording. If I can run PHP file or php code in template then i will implement google API and add response in email body. I am using CURL for dialplan, directory and configuration. I tried mailer-app=/usr/bin/php /usr/local/www/mailer_app.php but email not working because not getting To email id. Also is possible to send ${voicemail_file_path} to php mailer? cat /tmp/voicemailtoemail.txt To: From: FreeSWITCH mod_voicemail <0001(a)XX.XX.XX.XX> Subject: Voicemail from +919377579349 +919377579349 Mailer Error: You must provide at least one recipient email address. Please advice. -- Thanks & Regard Uday Mobile:- +91-9377579349

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call limit between multiple freewitchs using mod_db
by devang nathwani 10 Jul '17

10 Jul '17

I have two freeswitch servers load balancing is happening through opensips at front I am using mod_db of freeswitch to have centralized database for account based call limit over both freeswitch I have configured db.conf.xml settings with <param name="odbc-dsn" value="freeswitch:umn:pwd"/> However, when i set the limit per account freeswitch wont restrict call attempt as it should be. What could be the possible issue?

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mod_odbc_query
by Madovsky 10 Jul '17

10 Jul '17

Hi Folks, is it a normal behavior that mod_odbc_query accept only odbc-dsn like |<||param| |name||=||"odbc-dsn"| |value||=||"freeswitch:freeswitch:secret"||/>| |and not| |<||param| |name||=||"odbc-dsn"| |value||=||"pgsql://xxxxxx"||/>| | | |?| | | |thanks| | | |Franck |

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Generate a custom log alert on failed SIP Registration
by Joseph Waite 08 Jul '17

08 Jul '17

Hi Guys Wondering if it is possible, and if so how, to create a custom log ALERT entry on a failed SIP Registration? Basically I want to output an ALERT to the log file with “Auth Failed ${ip of register} I want to drop logging level down to Alert as this is going to be a high volume SBC, but I want to be able to use fail2ban to block on failed registers. Any help would be much appreciated. Regards

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One-way audio on first leg of attended transfer
by Richard-Freeswitch 08 Jul '17

08 Jul '17

I'm experiencing one-way audio on the first leg of an attended transfer. The originator of the transfer hears audio but the destination user does not. The one clue I have so far: the FS log shows a PGRES_FATAL_ERROR and the Postgresql log says 'ERROR: duplicate key value violates unique constraint "calls_pkey"', and the channels record update also fails as a result of the calls update failure. Any ideas on the cause of this issue? Log lines from fs_cli are pasted below. Thanks! 2017-07-07 16:15:28.819969 [NOTICE] switch_rtp.c:1275 Auto Changing audio stun/rtp/dtls port from 111.111.111.111:54590 to 10.10.10.10:54590 2017-07-07 16:15:28.829991 [DEBUG] switch_pgsql.c:415 Query (insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('f292dc4c-e1af-47b9-914d-cf8e7953e033','2017-07-07 16:15:28','1499544928','14fbd555-a49d-45c2-81c1-6a149446d6db','659c9953-685e-4e74-bd89-2da2f0973d79','freeswitch')) returned PGRES_FATAL_ERROR 2017-07-07 16:15:28.829991 [DEBUG] switch_pgsql.c:415 Query (update channels set read_codec='PCMU',read_rate='8000',read_bit_rate='64000',write_codec='PCMU',write_rate='8000',write_bit_rate='64000' where uuid='659c9953-685e-4e74-bd89-2da2f0973d79') returned PGRES_FATAL_ERROR 2017-07-07 16:15:28.829991 [ERR] switch_pgsql.c:656 Error executing query: ERROR: current transaction is aborted, commands ignored until end of transaction block

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