- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch on Kubernetes
by Mahmood Alkhalil 02 Jun '22

02 Jun '22

Would like to hear from anyone who managed to run Freeswitch on Kubernetes ? how did u setup the pod specs ? how did you expose the pod to public access ?

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User Variables from directory not accessible
by Örn Arnarson 02 Jun '22

02 Jun '22

Hello guys, I have a bit of a conundrum; I recently upgraded from FreeSWITCH 1.6.12 to 1.10.6. I copied most of the configuration over as-is and most everything works as expected. However, any variables set in the user directory that were previously accessible in the dialplan, such as accountcode and other custom variables that I set are no longer accessible from the dialplan when users originate calls. I can access them via the user_data function as a workaround, but I am struggling to understand why I need to do this. That is, I can access them like so: ${user_data(${caller_id_number}@${domain_name} var accountcode)} But not like so: ${accountcode} In the previous version (and on another FreeSWITCH installation of the same version), the latter is enough. Can anyone think of a reason why this would be the case? I have another FreeSWITCH instance running the same exact version where I can access directory variables without a problem, so it must be a configuration issue or a module that I am missing or something. Regards, Örn

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Programatically constructing switch_r_sdp
by Aditya Uppuluri 01 Jun '22

01 Jun '22

Hello Team, Is there a way to programmatically copy the SDP from one leg to the other? I have a situation, where i have two incoming legs and want to bridge them without answering them in freeswitch. I want to know if i can copy the SDP from one leg to the other programmatically (Not via XML but via ESL inbound) I Found that there is a way to do it in XML dialplan like this: <action application="set"> <![CDATA[switch_r_sdp=(sdp here) ]]> </action> How to achieve something similar using ESL inbound? Regards, Aditya

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Intermittent RTP gaps when using playback app
by Shaun Stokes 30 May '22

30 May '22

Hi All, We're experiencing intermittent RTP gaps from FreeSWITCH (roughly every 1 in 3 calls), when using the playback app in the dialplan on FreeSWITCH '1.10.7', when listening to the call it sounds like jitter. We've tried using sleep 1000 (after answer), 'jitterbuffer_msec=100:200:20', 'rtp_rewrite_timestamps=true', and 'send_silence_when_idle=400', but the problem remains the same. The problem only occurs when the server is under load, but calls not using the playback app are not affected. Here is an example. [cid:c78434ce-b599-4717-a140-4f250cc2be59] Has anyone come across this problem before, are there any solutions, or could this be a bug? Thanks, Shaun

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freeswitch.org confluence\wiki is down
by Shaun Stokes 28 May '22

28 May '22

Hi All, The freeswitch.org confluence\wiki has been down all day, the following error is displayed when accessing the site. 503 SERVICE UNAVAILABLE CONTACT TEAM(a)SIGNALWIRE.COM OR CALL +1.650.382.0000 I assume there's ongoing maintenance but haven't seen any announcements. Thanks, Shaun

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Build a WebRTC Gateway for Asterisk with Freeswitch (WebRTC to SIP/RTP).
by Manuel Montes Reinhardt 26 May '22

26 May '22

Hello all, I've been looking for documentation on how to build a WebRTC Gateway with Freeswitch (WebRTC to SIP/RTP) but I haven't found anything which could help me. I found something on this maillist: https://lists.freeswitch.org/pipermail/freeswitch-users/2020-April/133228.h… "FreeSwitch for WebRTC to SIP/RTP gateway?" so it seems it could be done but there is no documentation. I'm trying to build a gateway which could translate webrtc into SIP in order to let some users connect their extensions to a PBX based on Asterisk that does not support WebRTC. I'm looking for documentation but also a Company who could help us in the initial configuration. Thank you very much. Regards, Manuel

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Changed mod_xml_cdr
by Gregor Nanger 23 May '22

23 May '22

Something changed in XML that mod_xml_cdr produce with version 1.10.7. When transfer inbound call to extension caller_id_number variable is now extension number instead of number that call inbound. Here is example: *Before 1.10.7* <cdr core-uuid="4649fdb9-37f3-4930-a358-2e04837eb596" switchname="WORK-GREGZ"> <channel_data> <state>CS_REPORTING</state> <direction>outbound</direction> <state_number>11</state_number> <flags>0=1;2=1;22=1;40=1;42=1;91=1;96=1;110=1;112=1;113=1;114=1;115=1</flags> <caps>2=1;4=1;5=1;7=1</caps> </channel_data> <call-stats /> <variables> * <caller_id_number>16006554321</caller_id_number>* *Version 1.10.7* <cdr core-uuid="6733f6e5-2d81-453c-a8c4-e2220df4755b" switchname="WORK-GREGZ"> <channel_data> <state>CS_REPORTING</state> <direction>outbound</direction> <state_number>11</state_number> <flags>0=1;2=1;22=1;40=1;42=1;92=1;97=1;112=1;114=1;115=1;116=1;117=1</flags> <caps>2=1;4=1;5=1;7=1</caps> </channel_data> <call-stats /> <variables> * <caller_id_number>003</caller_id_number>* This kind of change breaks CDR import logic. I am asking if someone else noticed this? Is there some kind of bug or feature request that changed this? BR, Gregor

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Re: [Freeswitch-users] Error "Cannot create outgoing channel"
by Bote Man 21 May '22

21 May '22

I read on here that this is typically caused by a mis-match in CODEC. I see that you have many voice codecs, so perhaps it is a video CODEC? I copied your log to the FreeSWITCH pastebin which highlights the log lines when set to "FreeSWITCH" language. https://pastebin.freeswitch.org/view/a57f10d5#L97 2022-05-19 09:22:03.455283 [DEBUG] mod_sofia.c:866 Disabling proxy mode due to call answer with no bridge After the SDP lines I see a second routing attempt which immediately fails, but I see no outbound Leg-B set up. Do you add another party to the conference bridge via a script? That might be where the problem is. John Boteler Bote Communications -----Original Message----- From: FreeSWITCH-users <freeswitch-users-bounces(a)lists.freeswitch.org> On Behalf Of PICHARD, GUILLAUME via FreeSWITCH-users Sent: Friday, 20 May, 2022 18:19 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Error "Cannot create outgoing channel" Hello, I’m using MicroSIP with FreeSwitch. I can see presence so I guess my MicroSIP is working properly I can text message but I cannot make a call (audio or video). The error in FreeSwitch seems to be: 2022-05-19 09:22:03.475283 [ERR] mod_conference.c:1590 Cannot create outgoing channel, cause: INCOMPATIBLE_DESTINATION (full log there: https://justpaste.it/2h7si) Do you have any idea what might be wrong ? Thank you, Regards, Guillaume. _________________________________________________________________________ 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

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Error "Cannot create outgoing channel"
by PICHARD, GUILLAUME 19 May '22

19 May '22

Hello, I'm using MicroSIP with FreeSwitch. I can see presence so I guess my MicroSIP is working properly I can text message but I cannot make a call (audio or video). The error in FreeSwitch seems to be: 2022-05-19 09:22:03.475283 [ERR] mod_conference.c:1590 Cannot create outgoing channel, cause: INCOMPATIBLE_DESTINATION (full log there: https://justpaste.it/2h7si) Do you have any idea what might be wrong ? Thank you, Regards, Guillaume. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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sched_* lost on REFER?
by David Villasmil 19 May '22

19 May '22

Hello all, to create conferences, in the scenario: a->b hold a->c merge we refer the b leg to a conference in the same box. This channel previously had scheduled tasks and it seems those are being lost on refer. Is this correct and if not, anyway i can fix this? Many thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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