January 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

expand-vars-in-tag-body
by Gregor Nanger 28 Jan '22

28 Jan '22

Hi! I am fiddling with mod_httapi. Does anyone know what is difference between: <permission name="expand-vars-in-tag-body" value="true"> <permission name="expand-vars" value="true"> New FS installation doesn't have expand-vars anymore in config file.

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Dialplan apps clarification transfer & execute_extension
by Piccinin Francesco 28 Jan '22

28 Jan '22

Hi guys, I really don't understand well mod_dptools: transfer. In the documentation I read: "Immediately transfer the calling channel to a new context. If there happens to be an xml extension named <destination_number> then control is "warped" directly to that extension. Otherwise it goes through the entire context checking for a match." I have this instruction in my dialplan: <action application="transfer" data="extension_C XML calls"/> and a dialplan file like this: <include> <context name="calls"> <extension name="test_1" continue="true"> <condition> <action application="log" data="ERR ****** i'm in Test1"/> </condition> </extension> <extension name="test_2" continue="true"> <condition> <action application="log" data="ERR ****** i'm in Test2"/> </condition> </extension> <extension name="extension_C"> <condition> <action application="log" data="ERR ****** i'm in extension_C"/> </condition> </extension> </context> </include> isn't Freeswitch supposed to go stright to "extension_C"? in logs I can see that it executes all the extensions as they appears in the calls context. Maybe is there anything I'm missing? Thanks Regards Francesco Piccinin Telecomunicazioni Infrastrutture Insiel S.p.A. via San Francesco d'Assisi, 43 34133 - Trieste (TS) Italia tel +39 040 3737752 mob +39 335 1329440 francesco.piccinin(a)insiel.it<mailto:francesco.piccinin@insiel.it> www.insiel.it<http://www.insiel.it> AVVISO DI RISERVATEZZA Informazioni riservate possono essere contenute nel messaggio o nei suoi allegati. Se non siete i destinatari indicati nel messaggio, o responsabili per la sua consegna alla persona, o se avete ricevuto il messaggio per errore, siete pregati di non trascriverlo, copiarlo o inviarlo ad alcuno. In tal caso vi invitiamo a cancellare il messaggio ed i suoi allegati. Grazie. CONFIDENTIALITY NOTICE Confidential information may be contained in this message or in its attachments. If you are not the addressee indicated in this message, or responsible for message delivering to that person, or if you have received this message in error, you may not transcribe, copy or deliver this message to anyone. In that case, you should delete this message and its attachments. Thank you.

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Conference call based on RFC-4579
by H Yavari 27 Jan '22

27 Jan '22

Hi to all, I am trying to use FS to handle a 3-way call conference with REFER. A calls B, then puts B on hold and then calls C, then puts C on hold too, and calls conference room. After that A sends REFER to FS for B and C to move them to conference room. FS handles first REFER and transfers B leg to conference room correctly, but unexpectedly FS sends a BYE to A party for conference call too and terminated A party from conference with this :Reason: SIP;cause=601;text="ATTENDED_TRANSFER" I put this config in conference: <param name="conference-flags" value ="rfc-4579"/> but did not help and FS does not any 'isfocus' to OK contact header for A party.So is there any other config? or FS does not really handle Ad hoc conferencing based on RFC 4579. Thanks for your help. Regards,Hossein

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Delay Hangup on attended transfer
by Peter Steinbach 27 Jan '22

27 Jan '22

Hello, we are integrating MS Teams into our FS environment, and we have some problems with attended transfer - A calls B (Teams User) - B AttXfer to C (Teams) Call leg ob B is stopped immediately on attXfer, and this makes it difficult to handle the call in MS Teams, we would therefore like to delay this hangup. So is there any way to delay the hangup of leg B? I tried with api_hangup_cause, but this happed after Freeswitch has been sent it's BYE packet. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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SRTP internal calls failing for Push registered softphones
by Rasheed Kalapurackal 27 Jan '22

27 Jan '22

Hello All , I configured for SIP TLS + SRTP and i was trying to make calls between 2 registered softphones. One phone is a bria android softphone and it is push registered from the bria's push server (FSTest2 -4002) . The other phone is a softphone running on windows desktop (FSTest1 - 4001) . In this scenario it was noticed that the call from 4001 to 4002 is failing because , the call invite being sent to the push server does not have SRTP in it.. So it return unsupported media and is failing. Following are the log files for this scenario. https://pastebin.com/J2PkFVJa If i keep the phone 4002 in the foreground of the mobile and hence it is directly registered with freeswitch instead of a push registerd , it is noticed that the call is working fine and the call from 4001 to 4002 will ring the bria softphone 4002 because In this case the call INVITE being sent from the SIP server/PBX to the bria softphone contains SRTP, Following are the logs for this scenario. https://pastebin.com/1849GVQk Our freeswitch version is 1.10.7-release+git~20211024T163933Z~883d2cb662~64bit . I am trying to find a solution to this issue for last 2 days without any luck. Any help is highly appreciated. Thanks and Regards Abdul Rasheed

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OPUS - is FS ignoring incoming FAC packets?
by David Villasmil 27 Jan '22

27 Jan '22

Hello all, Wondering if anyone's seen this scenario: With 15% packet loss, FEC enabled. - sip -> pstn: -- We're seeing FEC out and in * client (OPUS) --> FS --> (PCMU/A) B-leg - Result: Choppy Audio * client (OPUS) <-- FS <-- (PCMU/A) B-leg - Result: Very good audio * client (OPUS) --> FS --> client (OPUS) - Result: Very good audio * client (OPUS) <-- FS <-- client (OPUS) - Result: Very good audio It's like FS is ignoring the incoming FEC when transcoding. I have a small jitterbuffer 20:100:20 Any ideas? David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Call is not terminated due to media_timeout
by Mirjana Jovanovic 26 Jan '22

26 Jan '22

Hello, I have the following issue: 1. An audio call is started from a client application 2. Connection is established 3. Client crashes (or is terminated manually through task manager) 4. The call is never terminated because of MEDIA_TIMEOUT I would expect the call to be dropped after the configured threshold for media timeout. I have traced the call to verify that no packages have been received from the client, after the client application has been terminated. The value for the channel variable media_timeout has been set, and also an action to be executed on media timeout (execute_on_media_timeout) has been defined. However, it seems that call status never changes to MEDIA_TIMEOUT, and therefore the call is never dropped. Are there additional/related properties that need to be set for this to work? Thanks

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sched_broadcast
by David Villasmil 22 Jan '22

22 Jan '22

Hello all, So i'm trying to get a sched_broadcast's task id so that i can later cancel it if needed. Problem is sched_broadcast returns: +OK Message scheduled sched_hangup doesn't return an id either (this is all in the xml dialplan) Anyone knows what i'm doing wrong? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Looking for an issue...
by Brian West 20 Jan '22

20 Jan '22

FreeSWITCHers, I've been tracing an issue down for bit, Many on this list have experienced the situation, The end result is the session still shows in "show channels" but uuid_kill says "-ERR No such channel!", A gcore will still show the session in the session table when using our gdbinit, If you've experienced this problem before please let me know. Any feedback may help me narrow down the root cause. Thanks, Brian West -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: https://www.facebook.com/signalwireinc?src=email] <https://www.facebook.com/freeswitch> [image: https://twitter.com/freeswitch] <https://twitter.com/freeswitch>

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Add new line ending after mod_event_socket JSON event
by Mahmood Alkhalil 20 Jan '22

20 Jan '22

Hi FS Community, I'm trying to add a '\n' new line after the JSON event sent from FS on the mod_event_socket when the event type is JSON, I need to do so as I'm writing a JAVA event parser and it would make the code much simpler if I can add this new line after the event. I'm not able to know where to edit exactly in the mod_event_scoket file (I have no experience in C lang at all) but I think it is somewhere around line 1274. If someone could help with this, I would really appreciate it. (the java socket lib am writing uses Netty DelimiterBasedFrameDecoder on '\n' nut it breaks as after the JSON event there is no new line and Content-Length: is directly written after the JSON event) Mahmood Alkhalil.

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