April 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dynamic payload negotiation
by Babak Yakhchali 30 Apr '19

30 Apr '19

Hi I'm trying to make calls between a custom softphone as client and freeswitch using opus as codec. when client calls freeswitch everything is working fine. but when I originate a call from freeswitch there is no audio from sofphone but client hears fs. After capturing packets I see that when client sends invite it is using 107 as its payload type and freeswitch sends 107 in reply. But when fs sends invite it is using 102 and client is sending 107. What is the standard? I think the softphone is doing something wrong because testing with jitsi works fine. thanks

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context of transfer call
by Bruce Marriner 29 Apr '19

29 Apr '19

I'm having the issue that when calls come in from the outside they hit the public context which sends the call to a phone, and then user at the phone tries to transfer them to another extension, then call is dropped. That's because, the transfer gets put in the "public" dialplan where the call came in from. I've read that other's have had this issue, and surely this is a normal use-case to transfer inbound calls, so I'm hoping this is a solved issue :) I read some things about using force_transfer_context but I can't seem to figure out how to use it. I though maybe it was a variable so I tried adding it as a variable in the user directory but that didn't help :( Can someone help explain how to address this issue so inbound calls can be transferred? Thanks. :) Ideally, I'd like FreeSWITCH to use the context of the phone/user that's doing the transfer.

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Sending periodical audio announcement to callee
by Arik Halperin 29 Apr '19

29 Apr '19

Hello, I would like to play an audio recording to the callee of a call every few seconds. What is the best way to do this? Thanks, Arik Halperin

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Record_ession and SIP INFO record: on
by Allan Kristensen 28 Apr '19

28 Apr '19

Hello, I'm having some issues getting dialplan command "record_session" and SIP INFO messages to work together. If recording is started using record_session in the dialplan, at the beginning of the call and I then send a SIP INFO message with "record: on" i get the response "Already recording", which is as expected. To avoid this situation, we then send the recording state in response header, so SIP client knows if recording is already started at the begining of the call. Now if I try to stop the recording using SIP INFO message, I get the error message "Nothing to stop"? I'm guessing it has something to do with recording bug being in a different session of the call. Is there anyway to adjust this behaviour? (One could argue that just because I send a SIP INFO to stop the recording does mean that it should be allowed) Thanks,

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originate with SIT tones
by John Burke 26 Apr '19

26 Apr '19

Hello: Does the originate command support monitor_early_media_fail like the following: originate {ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}sofia/gateway/test/1000 All examples online that I have seen are for the bridge application, but I am not having much luck with originate. I made several test calls on FS1.8.5 with 480/620 tones in the 183, but originate progresses normally without firing an event or hanging up the call (not sure which one is expected). Anyone have any examples, configs, etc? Thanks, John

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No Video in conference video mux mode with VP8 but H264 works fine
by Faisal Hanif 26 Apr '19

26 Apr '19

I have a conference with video-mode mux and I have proper mixed video with H264 but with VP8 only empty canvas sent by FreeSWITCH while video-mode passthrough send proper video with VP8. Any idea what's wrong or missing here? Regards, Faisal

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Re: [Freeswitch-users] P-Asserted-Identity in display updates for Polycom's
by Shaun Stokes 26 Apr '19

26 Apr '19

Polycom's are on current firmware '5.7.4'. We use FreeSWITCH internally with Kamailio as our internet facing registrar but FreeSWITCH is grabbing the IP address of Kamailio and using it in the PAI which is sent to the device. I'm going to do some more digging to see if there's anyway we can get FreeSWITCH to use the domain\context instead of the IP address, if not then we should be able to fix this with-in Kamailio before the UPDATE is sent to the device. Thanks, Shaun ________________________________ From: Nathan Neulinger <nneul(a)mst.edu> Sent: 25 April 2019 14:17:19 To: FreeSWITCH Users Help; Shaun Stokes Subject: Re: [Freeswitch-users] P-Asserted-Identity in display updates for Polycom's Are you running current polycom firmware on the devices? Sounds from the commit like it was changes after polycom fixed bug on their side. -- Nathan ________________________________ From: Shaun Stokes <shaun.stokes(a)itec-support.co.uk><mailto:shaun.stokes@itec-support.co.uk> Sent: Thu, Apr 25, 2019 7:10 AM CDT To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org> Subject: [Freeswitch-users] P-Asserted-Identity in display updates for Polycom's After FreeSWITCH 1.8.5 (and 1.8.3) we've noticed that display updates have changed, on Polycom's (UA/5) for example this now includes the host portion such as the IP address which breaks directory contact association and is confusing for the user to see on the display: "Name" <sip:Number:Host:Port> Previously in FreeSWITCH 1.8.2 on Polycom's and on most other devices we've tested this doesn't include the host portion and works perfectly: "Name" <Number> This is a result of this change: 9e8ae18f95e<https://freeswitch.org/stash/projects/FS/repos/freeswitch/commits/9e8ae18f9…> How do we stop FreeSWITCH from including the IP address in the PAI on display updates for Polycom's? Other than manually reversing this change in our environment. Thanks, Shaun _________________________________________________________________________ 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<mailto:sales@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users https://freeswitch.com -- ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu<mailto:nneul@mst.edu> Missouri S&T Information Technology (573) 341-6679 System Administrator - Architect (573) 612-1412 System and Desktop Infrastructure Team Manager

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VoiceMail to Email has blank WAV file
by Sean Devoy 26 Apr '19

26 Apr '19

Hi All, I have been trying to get VoiceMail to email working in the latest production version (1.8) on Ubuntu. I am using sendmail and I have gotten as far as sending the email with an attachment. The attachment is a wav file of the correct length but has no audio. It is for example 5 seconds of silence. In the voicemail folder for this extension is a wav file with audio just fine. The logfile says: 2019-04-26 04:01:46.842091 [DEBUG] switch_utils.c:1210 Emailed file [/tmp/mail.155625130604d1] to [sdevoy(a)bizfocused.com] 2019-04-26 04:01:46.842091 [DEBUG] mod_voicemail.c:3042 Sending message to sdevoy(a)bizfocused.com<mailto:sdevoy@bizfocused.com> I have checked to protection on the /tmp folder. Any ideas what is wrong? Thanks, Sean

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ESL API command to get the status of Freeswitch
by Ramesh Kandasamy 25 Apr '19

25 Apr '19

Hi everyone, I am looking for a way to get the status of a Freeswitch from a remote server through ESL API command. I tried using SendRecv and Api methods of ESLconnection class to send the 'status' command. But I don't receive the response. However, after further reading through the confluence, I realized that it is probably because, 'status' command output doesn't include with a header named "content-type" that has a value of "api/response" or "command/reply" and both SendRecv and Api methods look for this. So, the question is what other ESL methods shall be used to get the status of freeswitch? Thanks Ramesh

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Bridge question
by Sean Devoy 25 Apr '19

25 Apr '19

Hi, My customer wants to ring the local extension, then 15 seconds later connect their cell phone. I have the leg_delay_start working fine. What I don't know is how to pass the internal Caller ID to one and the external caller id to the other? Thanks, Sean

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