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Playback file stopping abruptly from freeswitch 1.10.5 and onwards
by Shahnawaj Khan 02 Feb '22

02 Feb '22

Hi All, Recently we have moved from freeswitch 1.10.3 to 1.10.5. and we observed that while executing the following command through ESL Command Execute playback(random.wav) freeswitch is stopping the playback earlier than the file playback duration.In this setup we are using freeswitch to generate the call as well. Our setup looks similar to this. Freeswitch_1------>opensips------>Freeswitch_2 Freeswitch_1: a) Originate and park b) playback some file on park (Using command though ESL) Freeswitch_2: a) playback some file on park (Using command though ESL) In Attached logs freeswitch has stopped the playback after 3 seconds but the file is of 10 seconds. But in another scenario where we replace the freeswitch_1 with a manual call using a softphone, Freeswitch_2 plays the file for the whole 10 seconds. Please let me know if anyone has faced the same issue or any insight regarding it. We observed the same issue with 1.10.7 but working fine in our old 1.10.3 setup. Thanks & Regards, Shahnawaj

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SRTP calls randomly have no audio
by Gledi Musa 01 Feb '22

01 Feb '22

Hello, I'm using an instance of FreeSWITCH Version 1.10.7-release-19 in a Debian 11 VM, with OpenSSL 1.1.1k. The carrier only supports TLSv1, so I have hardcoded it in vars.xml and openssl.cnf to use only it, instead of TLSv1.2. On random calls there is no way audio, and without any changes, the next calls work, and then some other calls have no way audio. The logs don't throw any errors, but in the cases where there is no audio, I see that the DTLS negotiation doesn't go past HANDSHAKE, and stays like that for minutes, without throwing any errors. I have tried to set "legacyDTLS=1" as a channel variable, but that doesn't seem to have any effect. Any idea how to approach this?

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transfer_on_fail and continue_on_fail
by David Villasmil 31 Jan '22

31 Jan '22

Hello guys, I want to use transfer_on_fail to transfer the leg to another extension, i do this by setting the failure_causes and continue_on_fail to false (because of https://github.com/signalwire/freeswitch/blob/master/src/switch_channel.c#L… it will not work if set to true). that works, but i ALSO want to use continue_on_fail, how can i do this? i only want to be transferred if cause is busy. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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ReINVITE with codec change
by David Villasmil 31 Jan '22

31 Jan '22

Hello All, I’m testing a client and would like to test if it supports reINVITEs to change codecs… any suggestions on how to test that? Thanks all -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Re: [Freeswitch-users] Conference call based on RFC-4579
by Sergey Safarov 30 Jan '22

30 Jan '22

RFC-4579 is not fully implemented in FreeSwitch. Feature you a looking is not present in FreeSwitch.

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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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