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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

3 2

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

2 1

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

4 10

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

1 0

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

2 1

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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Re: [Freeswitch-users] Voicemail on AWS
by David Villasmil 31 Oct '19

31 Oct '19

Why not just uploading the files once it’s been recorded? What’s wrong with that? On Sat, 26 Oct 2019 at 13:45, H Yavari via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: H Yavari <hyavari(a)rocketmail.com> > To: freeswitch-users(a)lists.freeswitch.org, "per(a)wgtwo.com" <per(a)wgtwo.com> > Cc: > Bcc: > Date: Sat, 26 Oct 2019 12:34:39 +0000 (UTC) > Subject: Re: [Freeswitch-users] Voicemail on AWS > As @Per said, there is problem with wave files. > I know that S3 doesn't support appending to file, so I think FS makes some > headers for wave file and then write to it, and on S3 we have just wave > headers. > > If someone has any idea, please share. > > Regards, > HY > > On Wednesday, September 25, 2019, 10:21:59 AM GMT+3:30, H Yavari < > hyavari(a)rocketmail.com> wrote: > > > Hi Per, > > Thank for your hint. > So you didn't use http_cache right. Do you have the HA for your system > also? is it ok with this structure? > > Regards, > H.Yavari > > On Tuesday, September 24, 2019, 09:52:57 PM GMT+3:30, Per Modin < > per(a)wgtwo.com> wrote: > > > On 2019-09-23 20:10 (CEST), H Yavari via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> wrote: > > Date: Sun, 22 Sep 2019 09:32:56 +0000 (UTC) > > From: H Yavari <hyavari(a)rocketmail.com> > > To: "freeswitch-users(a)lists.freeswitch.org" > > <freeswitch-users(a)lists.freeswitch.org> > > Subject: Voicemail on AWS > > X-Mailer: WebService/1.1.14303 YMailNorrin Mozilla/5.0 (Windows NT 10.0; > > Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/77.0.3865.75 > > Safari/537.36 > > > > I am looking for some hints how we can make an voicemail server for > > our company users to have their own voicemail box, then we save all > > voice files on AWS S3. I see mod_http_cache, but afraid of it's only > > solution or there is better one. > > Hi, > > we used to run on S3, where we mounted the recordings with s3fs-fuse[1] > which we ran successfully on 1.6.20. > > HOWEVER, when we updated to 1.8.5 we ran into problems with the wave > files, the media stream was not saved and we ended up with empty media > containers, with only some meta data of around 250 bytes ended up being > saved. This issue affected about half of all recordings, and we found no > correlation in regards to codecs used etc. If this was an issue with > configuration, s3fs-fuse or Freeswitch we didn't investigate. > > We solved this by moving the storage from S3 to EFS, and are currently > running it in production without issues in that regard. > > [1]: https://github.com/s3fs-fuse/s3fs-fuse > > -- > Per Modin > Software Engineer > Working Group Two > m: +46724452037 > w: https://wgtwo.com e: per(a)wgtwo.com > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com > > > > ---------- Forwarded message ---------- > From: H Yavari via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org > > > To: freeswitch-users(a)lists.freeswitch.org, "per(a)wgtwo.com" <per(a)wgtwo.com> > Cc: > Bcc: > Date: Sat, 26 Oct 2019 05:45:13 -0700 (PDT) > Subject: Re: [Freeswitch-users] Voicemail on AWS > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > https://freeswitch.com -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 5

overwrite the Caller-ID-Number
by Alexander Haugg 31 Oct '19

31 Oct '19

Hi, There is a bug in the Unify Openscape Business PBX. The P-Asserted-Identity is wrong (have the same values like the Diversion Header). The incoming SIP call will routed into the mod_callcenter application and show the wrong caller number. How can I set the correct information into the Caller-ID-Number in the dialplan? I was trying <action application="set" data="effective_caller_id_number=${newNumber}"/> <action application="set" data="outbound_caller_id_number=${ newNumber }"/> <action application="set" data="origination_caller_id_number=${ newNumber }"/> But in the call center events I can see, the caller-id-number have the original wrong value. Or is there a possibility to force the from hader as base for the caller id informations? Thanks a lot Alex

4 7

avmd stops when bgapi uuid_broadcast starts
by ivan381eu 31 Oct '19

31 Oct '19

What I need is to start intro sound file playback on call answer and detect beep sound from callee side What I tried is: 1. start avmd 2. bridge call 3. on answer start intro sound message playback using bgapi BUT when background playback starts avmd seems to be paused and continues after playback is completed. gateway dialplan is: <extension name="mytelco"> <condition field="destination_number" expression="^987(\d+)$"> <action application="set" data="effective_caller_id_number=46xxxxxxxx"/> <action application="set" data="record_link=${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="record_session" data="/var/www/html/freeswitch/recordings/${record_link}"/> <action application="lua" data="/etc/freeswitch/scripts/beep_detection-from-call-v0.1.2.lua $1 ${record_link}"/> <action application="hangup"/> </condition> </extension> file beep_detection-from-call-v0.1.2.lua #!/usr/local/bin/lua DNID = argv[1] if session:ready() then session:sleep(1000) session:execute("avmd_start", "inbound_channel=1,outbound_channel=0,sample_n_continuous_streak=10,sample_n_to_skip=0,debug=1,report_status=1") session:execute("bridge", "{execute_on_answer=lua /etc/freeswitch/scripts/on_answer-v0.3.lua " .. uuid .. "}sofia/gateway/mytelco/" .. DNID) session:execute("avmd_stop","") end --if file on_answer-v0.3.lua #!/usr/bin/lua uuid_original = argv[1] api = freeswitch.API() reply = api:executeString("bgapi uuid_broadcast " .. uuid_original .. " /usr/share/asterisk/sounds/en/dennis-ivan.gsm both") freeswitch.consoleLog("INFO","==============================================") freeswitch.consoleLog("INFO","API reply is:" .. reply) freeswitch.consoleLog("INFO","==============================================") -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

2 1

tcp-keepalive stuck at 30 seconds
by Nathan Stratton 30 Oct '19

30 Oct '19

<param name="socket-tcp-keepalive" value="true"/> <param name="tcp-keepalive" value="10000"/> Even tho, tcp-keepalive is set to 10 seconds (10000 ms), wireshark shows WSS TCP Keep-Alive every 30 seconds. ><> nathan stratton

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