November 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch 1.10.X mod_limit db configuration
by Volodymyr Fedorov 11 Nov '19

11 Nov '19

Hi I need some help from community, I have a question regarding mod_limit db configuration in modern freeswitch release where we have new db drivers. I tried to use following construction: <configuration name="db.conf" description="LIMIT DB Configuration"> <settings> <param name="odbc-dsn" value="mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;"/> </settings> </configuration> When I try to load mod_db I`m getting following error: 2019-11-11 09:22:11.813986 [ERR] switch_xml_config.c:267 Invalid value [mariadb://Server=localhost;Database=freeswitch;Uid=freeswitch;Pwd=fsgreat!;] for parameter [odbc-dsn] freeswitch@loopkam> Item name: [odbc-dsn] Type: string (optional) Syntax: dsn:username:password Help: If set, the ODBC DSN used by the limit and db applications I have following version installed: FreeSWITCH (Version 1.10.1 -release-12-f9990221e6 64bit) is ready Thanks for advices! -- Best regards, Volodymyr

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ERR has no write codec. [INCOMPATIBLE_DESTINATION]
by David Dean 10 Nov '19

10 Nov '19

Hi I wonder if anybody can help. We are getting an error creeping in on a production system. It is happening randomly which of course makes it difficult to track down. A call to a SIP registered endpoint (VoIP handset) randomly fails, it rings, then when it is answered the call fails. It could be to any of the end points on our system, it doesn't happen every call, in fact it only happens on prepositionally very few calls. If we restart the Freeswitch service the frequency of calls effected reduces to nothing and then builds up again. Logs show (I have replaced our IP address with {IP ADDRESS}) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.857442 [DEBUG] mod_sofia.c:646 SOFIA EXCHANGE_MEDIA freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [ERR] switch_core_media.c:15418 sofia/internal/5033305@{IP ADDRESS}:38532 has no write codec. freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [NOTICE] switch_core_media.c:15419 Hangup sofia/internal/5033305@{IP ADDRESS}:38532 [CS_EXCHANGE_MEDIA] [INCOMPATIBLE_DESTINATION] freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.897441 [DEBUG] switch_ivr_bridge.c:1881 sofia/internal/5033305@{IP ADDRESS}:38532 skip receive message [UNBRIDGE] (channel is hungup already) freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 2019-11-08 10:01:02.837454 [DEBUG] sofia.c:7301 Remote SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=5033305 38480 137 IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=Mapping freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 51.148.123.105 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 38480 RTP/AVP 0 101 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Local SDP: freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 v=0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 o=FreeSWITCH 1573174674 1573174675 IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 s=FreeSWITCH freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 c=IN IP4 178.128.173.117 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 t=0 0 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 m=audio 32570 RTP/AVP 0 101 13 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:0 PCMU/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:101 telephone-event/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=fmtp:101 0-16 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=rtpmap:13 CN/8000 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=ptime:20 freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 a=sendrecv freeswitch.log.2019-11-08-10-15-53.1:7ea8711f-40c3-4cfc-b927-fc99fc9b7437 Any ideas would be appreciated. David J Dean

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fs_rpt.pl radio interface and mod_echolink
by Stephen Miller 07 Nov '19

07 Nov '19

Hello everyone! I am wondering if there still exists a copy of fs_rpt.pl and mod_echolink for FreeSWITCH. I have a few applications that I would like to explore... and really eventually switch off from Asterisk. Any help would be greatly appreciated! Thank you! -- Steve Miller KC1AWV

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Fax T38 problems in proxy media
by Jose Fco. Irles Durá 06 Nov '19

06 Nov '19

Hi, I have sended another email to the list with attachments. I think that the list not accepts attachments, sorry. I have several FreeSWITCH's (latest version from deb packages: 1.10.1 -release-12-f9990221e6) and I have problems with faxes in proxy media (in bypass-media works). In the sofia.conf.xml I have setted the param t38-passthru=true If the flow is: 1. Initial INVITE 2. Re-Invite from one of the legs with t38 sdp Result: works But if the flow is: 1. Initial INVITE 2. Re-invite from one of the legs changing the media (at answer) 3. Re-invite from one of the other legs with t38 sdp Result: not works I tried with "soa_enable=false" without luck. FreeSWITCH seems that hangs the sip transaction. This capture is the first reinvite (in the flow, the "showing" line): https://pastebin.com/raw/X4zrUr4k This capture is the reinvite with the t38 negotiation sended by the receiver (Grandstream ATA): https://pastebin.com/raw/KCXD51aH And this with the response (200ok) accepting the t38 negotiation): https://pastebin.com/raw/ifqtPN8L How I can debug the problem? Best regards

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mod_perl on 1.10.1
by ik 06 Nov '19

06 Nov '19

Hello, I installed under Centos 7 fs 1.10.1 out of the repos - A clean new installation an a clean server. The system replaces an old system from 1.4.x without changing my code, that is written in Perl. Once a week it looks like the loading of the perl script using `perlrun` stop loading the modules properly, getting error that some variables such as %Config were not found. Restarting FS fixes the issue for a week or so. I could not find an open bug on it, but before reporting, I wish to know if there anything else I should be looking for when such issue found? Thank you Ido

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Creating RESTful API xml for dialing outbound numbers into a conference
by Spencer Angerbauer 05 Nov '19

05 Nov '19

Hello, I am semi new to FreeSwitch but have worked in Asterisk for many years. I am trying to create a simple originate api call from the webapi module. I have been able to successful create an internal user extension 1001 and connect to 1002 in a default conference room. (See below call format which was successful. http://XX.XX.XX.XX:8080/webapi/originate?user/1001%201002%20XML%20default <http://xx.xx.xx.xx:8080/webapi/originate?user/1001%201002%20XML%20default> My question is, How do I take this same command structure to connect multiple external numbers (not internal extensions) into a specific or unique conference room via my SIP trunk line? I have tried multiple variations but am having difficulty finding anything online that shows some good examples of JSON restful webapi’s for originating an outbound call to 2 or more numbers and connecting them into a conference room... I was able to successfully execute the webapi call below for a simple originate api call to an external number and connect but am unsuccessful and bridging these two api commands into what we need. http://XX.XX.XX.XX:8080/webapi/originate?{origination_caller_id_number=1801… <http://54.215.223.18:8080/webapi/originate?%7Borigination_caller_id_number=…> Please let me know if you have any resources or API references to simply connect 2+ external calls into a unique conference via the webapi calls. Thank you so much in advance! -Spence

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mod_hash
by Max Solaris 05 Nov '19

05 Nov '19

Hi, I found this module and seems to be very useful, however I have a question. I can see methods supported talks about insert / delete a single key by realm, and this is fine. But, is there a way to delete a complete realm in order to rebuild it? I’m thinking in storing a custom actions LCR in memory and this module fits my needs, but I believe delete a single key each and rebuild a new realm is not so easy do deal with. Thanks Max

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