Freeswitch no audio ios. Apr 13, 2014 · It means what it says.

Freeswitch no audio ios But the bad way - ICE is the problem here. Previous message: [Freeswitch-users] JitterBuffer possible problem when configured with number of packets. I've created a new SIP profile for my domain [pbx01. Previous message: [Freeswitch-users] No audio in new bridge loopback/app after bridge timeout Tais Plougmann Hansen taisph at osd. We are sending the uuid_audio [uuid1] start command because we do not have any media audio when we do the bridge. com Mon Apr 5 10:28:17 PDT 2010. May be we are over looking something. conf. Previous message: [Freeswitch-users] No audio after bridging outbound calls Next message: [Freeswitch-users] Playback is not working Messages sorted by: Feb 9, 2018 · Hi all Let me say in advance, Thank you, as I have struggled all day trying to get this working. This does timeout and it does transfer but there is no audio when it goes to either voice mail or an ivr. so the OS catches the port not open and returns an ICMP 3:3 back to the MSS. webrtc:google-webrtc:1. It seems that this happens only if users are connected to my FreeSwitch using some specific italian internet providers. com Fri Oct 4 23:19:38 MSD 2013. We tried also sending a uuid_media off command but that has the consequence of dropping the First make sure your audio devices are configured properly in Zoiper. xml you can uncomment the start and stop ports and create firewall rules to allow these. [Freeswitch-users] No sound Cliconnect cliconnect at cliconnect. sampling-rate - choice of Oct 4, 2021 · I have made a mode-verto android client, using WebRtc; Pre-built library: org. > > I have tried using the playback, echo, and bridge actions. If you are using a laptop and expect to hear the sounds come out of the speakers, check to be sure that no audio headset (wired or bluetooth) is currently connected. Jul 13, 2022 · Describe the bug call between source A and destination B, once the call is stablish, B side, that is behind a SBC transfers the call internally, FS receives the multiples re-invites from the SBC (3pcc is enabled to proxy mode in sofia pr But when I call that contact, no > matter what action I specify in the dialplan, I don't hear any sound. When I initiate a call Video port was opened but later it was closed. Hello, I am having a problem with bridge_answer_timeout, also having email problems, hope this isn’t duplicated I included the dial plan below. call B and connect to Conf B. Next message: [Freeswitch-users] Attended transfer - no audio Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi, we have a Freeswitch server on a public IP and a few phones behind NAT. I've tried different NAT settings - no result. > > The FreeSWITCH server is behind a firewall, but I do not think it's a > NAT problem because then there would be no sound for outgoing calls made > with Feb 17, 2015 · If you want to hear audio in the second recording, you must make the recording LONGER than the first recording. Previous message: [Freeswitch-users] Oneway audio issues in freeswitch Next message: [Freeswitch-users] Oneway audio issues in freeswitch Messages sorted by: try blocking ICMP packets TO the MSS. I want to read the audio stream from FreeSWITCH and send it over to the Speech engine. there is no audio on inbound placed calls to the pbx for the remote extension(B leg?) . Regards, Tidiane > Date: Sat, 30 Oct 2010 04:21:10 +0200 > From: Prometheus001 at gmx. 10. 17 Registration: works; Caller Id: unknown; Call in/out: work; Call waiting: unknown; Transfer calls: unknown; Park calls: unknown; NAT Traversal: unknown (Initial Failure) Oct 22, 2024 · Making a call to a remote side that sends RTP in sessions in progress but when it answers there is no audio. org Subject: No input audio with FSClient I seem to have a problem with audio input when using FSClient. From pjsip to FreeSwitch(1st leg) it was video call but 2nd leg is audo call. If one of the legs. Hi All, FreeSWITCH Version 1. RTP ports need to be open for media to flow. In /etc/freeswitch/autoload_configs/switch. org Sat May 14 03:35:52 MSD 2011. xml Oct 31, 2020 · In my current project ,we are using webrtc to connect two mobile clients in android. dk Thu Jun 27 14:11:54 MSD 2013. We are using these api commands to do this: (we believe this is due to the uuid_audio start). 5. What could be the solution? The windows program from manufacturer, VMS, is still playing the stream. 20 - to a Google cloud Debian 8 instance. 168 Nov 14, 2023 · No audio is typically a firewall issue. Feb 13, 2015 · All WebRTC clients are inside local network, so ICE isn't needed here. Provide details and share your research! But avoid …. Previous message: When setting up a call, pass in an sdp offer with recvonly. Or to read it and then I can manually stream it. Next message: [Freeswitch-users] No audio/dtmf from softphone behind NAT Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] sofia loglevel all 0 sofia profile xx siptrace on replace xx with profile. com Wed Jul 30 09:56:56 MSD 2014. I've made a lot of tests and found that if call initiator is Web RTC client and there is some delay in answer (10-25 seconds) - audio is completely absent. c=IN IP4 192. Previous message: [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back. ivancev at gmail. 0. 10 built from source. com Mon Aug 25 10:39:45 PDT 2008. Verto Communicator runs in a web browser and speaks the Verto protocol to FreeSWITCH. In all other cases rtp is ok. In one instance, my server's network was the problem, and setting up the same exact FreeSWITCH (and even Asterisk) configuration resulted in two way audio. 71 (64-bit). 6. 77. After you update to head, of course. We are using socket signaling to exchange data between two mobile clients. If you have it enabled, disable it and see if that helps ( you may have echo though ). 261172 If audio isn't flowing, all I'm trying to say is it might not be a FreeSWITCH issue and you should make sure Chrome isn't the culprit before changing too many things in FreeSWITCH. eth1 file in network-scripts with proper route config [Freeswitch-users] Conference audio issues when Verto user sends video to audio-only conference Michael Jerris mike at jerris. On Fri, Apr 4, 2014 at 9:53 AM, Brian West <brian at freeswitch. Thanks! SIP - No audio or one way audio ( on Ios) « Back. [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 Jeff Pyle jpyle at fidelityvoice. org. Previous message: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio Next message: [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio I'd start by getting a debug log of the call and posting it on pastebin. 8-dev~64bit ( 64bit) Trace logs May 15, 2014 · [Freeswitch-dev] STUN Binding Request failed causing no audio when join to conf Sasa Ivancev sasa. Audio_stream is built on ixwebsockets, statically linked c++ lib, compiled together with the module. net > To: freeswitch-users at lists. c:4918 Audio Codec Compare Technically speaking, mod_audio_stream is a simpler but yet effective, and less dependent (no external libs required), while audio_fork is built on libwebsockets. However with or without > that although I can hear audio from the server, audio to the server isn't > arriving (doesn't appear in recordings), and dtmf doesn't get received > either. com Tue Mar 17 17:50:55 MSK 2015. In these cases I have no audio. s=FreeSWITCH. Previous message: [Freeswitch-users] No Audio with Skypopen (Oz Mortimer) Next message: [Freeswitch-users] No Audio with Skypopen Messages sorted by: If audio isn't flowing, all I'm trying to say is it might not be a >> FreeSWITCH issue and you should make sure Chrome isn't the culprit before >> changing too many things in FreeSWITCH. My phone and lightning to hdmi will have audio for any other app like YouTube etc. v=0. Audio is not muted (a=sendrecv), and there is active in and out audio RTP. ac. The best way to diagnose this is to look at a packet capture, sometimes referred to as a pcap trace. 3 jail broken with checkra1n), however I get no audio from the phone speakers, or if I have the phone screen mirroring through lightning to hdmi. o=FreeSWITCH 1540768194 1540768195 IN IP4 192. The problem is caused that it sends in the 200 OK SDP a different port number and crypto Apr 20, 2016 · [Freeswitch-users] No audio Freesswitch Deepika Yadav deepikay at iiitd. com Wed Sep 23 22:38:11 MSD 2015. 8. 14 (64bit) + Chrome Version 46. com Thu May 15 02:39:12 MSD 2014. The range can be less also. Previous message: [Freeswitch-users] RECORD_STEREO should not make both channels the same Nov 13, 2019 · about this problem，something message : freeswitch 1. When I call > the contact in Google Talk, I don't see any obvious problems in the > console output, but I hear nothing. log. 6. Vice versa. org] On Behalf Of Mirko Brankovic Sent: Monday, August 15, 2016 10:30 AM To: FreeSWITCH Users Help <freeswitch-users at lists. Previous message: [Freeswitch-users] FS behind NAT encounter no audio Next message: [Freeswitch-users] FS behind NAT encounter no audio Messages sorted by: But when I call that contact, no > matter what action I specify in the dialplan, I don't hear any sound. EDIT: Debi/Ubu has dropped the live555 library, that VLC used to play RTSP. Is there any way in the native API to get the audio stream. > > If, for example, I call from a softphone registered on Iptel. I believe the reason for this is that the audio buffer from the first recording is still associated to the recording session and it is full of silence. ESL is a powerful module in Freeswitch where you can able to get all the events of freeswitch application and play with. We tried also sending a. Dummy Soundcard for Amazon linux server. i'm able to establish connection without the audio though. com> Sent: 09 January 2017 16:58:53 To: freeswitch-users at lists. Unable to ping EC2 Server. i had this exact same problem a few months ago. to test freeswitch webrtc with chrom + jssip, using the latest git version 1. org with a > second account to the Iptel. Nov 29, 2023 · However, I encounter an issue when attempting to make calls to external destinations, such as GSM numbers. com Wed Jul 11 20:04:01 UTC 2018. In [Freeswitch-users] No audio when calling in via SIP phone Iqbal Abdullah iqbal. internal sip endpoints i'm able to establish 2 way audio but not when i call out. b. disconnect in order to launch the next call. Previous message: [Freeswitch-users] no audio after hold Next message: [Freeswitch-users] no audio after hold Messages sorted by: Apr 4, 2014 · What exactly is your use case? I can assume by the line numbers that you’re not running the same code. Previous message: [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 [Freeswitch-users] No sound during a video call Anthony Minessale anthony. com Wed Sep 9 21:14:53 MSD 2015. There is no such setting, "live555 stream transport" or anything like it. org account that was registered from > FreeSWITCH, it answers the call but no sound. Make sure the input device and output device are selected properly. It implements the WebRTC specification for audio and video streaming. com Thu Feb 18 01:01:30 MSK 2016. 1b+git~20130423T194907Z~e1c325dcb5 (git e1c325d 2013-04-23 19:49:07Z) Error : while error its not accepting calls o=Sonus_UAC 902944909 Hello, I have a problem (probably with nat) with calls that are answered after 30 seconds. I know the mic is providing sound and the FreeSWITCH server is operating OK because I can use X-Lite to perform the same test and I get the audio feed played back to me. have any media audio when we do the bridge. xml (if you're on a Linux machine). In these cases, there is no voice transmission for both the sender and the receiver. Nov 5, 2015 · [Freeswitch-users] verto no audio? robert mundkowsky rfmundkowsky at yahoo. Previous message: [Freeswitch-users] No sound during a video call Next message: [Freeswitch-users] Getting 503 Service Unavailable while enable record_session Messages sorted by: Sep 26, 2013 · Thanks for contributing an answer to Stack Overflow! Please be sure to answer the question. 8b to build and run under windows, when calling an example ivr(e. 3. king at quentustech. com], I've set the ext-rtp-ip and uuid - Freeswitch channel unique id; wss-url - websocket url ws:// or wss:// mix-type - choice of "mono" - single channel containing caller's audio "mixed" - single channel containing both caller and callee audio "stereo" - two channels with caller audio in one and callee audio in the other. Sep 26, 2015 · I am using Microsoft speech with FreeSWITCH. Paul _____ From: Paul Mateer <paul. [Freeswitch-users] no audio after hold Nick 'tarantul' Novikov tarantul at gmail. org Fri Sep 25 15:17:48 MSD 2015. It happens about every 300 calls。 Here is a call log（freeswitc Mar 16, 2023 · Thanks for that Adrian, strangely all of our calls this morning began to be effected by the 1 way or no audio issue intermittently. Previous message: [Freeswitch-users] No audio after transfer Next message: [Freeswitch-users] No audio after transfer Messages sorted by: (we believe this is due to the uuid_audio start). Jul 30, 2014 · [Freeswitch-users] No audio when calling in via SIP phone William King william. pristine:libjingle:11139@aar and FreeSwitch but only got success to make uni- Apr 4, 2014 · Its a webrtc call with no media flowing. I have a pcap and the freeSWITCH logs, and I can easily reproduce this. Package version or git hash FreeSWITCH Version 1. We were able to connect peer to peer Feb 23, 2017 · Originate a call to A and connect to Conf A on answer. If I dial test tone or test music feature code, audio from server is being routed properly, so the issue is with the connection from a softphone located outside VMs LAN. Log from caller INVITE sip:1001@10. [freeswitch at localhost fs]$ grep 43945544-ba03-11e3-8df9-7b004e0e7c32 fs. org> Subject: Re With any external speaker connected via Bluetooth I get full audio running cores through retroarch on my iPhone X (13. We need that. 168. Previous message: [Freeswitch-users] Port changes in multiple 183s cause no audio after 200 After update both Ubuntu and Debian, VLC stopped playing the Sunba rtsp stream. The other end is not sending stun like it expects. This can either be a network issue or the client has disconnected unexpectedly. 1 43945544-ba03-11e3-8df9-7b004e0e7c32 2014-04-02 09:08:13. If the microphone is enabled, but still there is one-way audio, or no audio at all, please try to disable or [Freeswitch-users] No audio after bridging outbound calls Brian West brian at freeswitch. org> Subject: Re . Audio driver is the default Coreaudio, mute is off and volumes are up. Expected behavior When FreeSWITCH responds with an answer of sendonly it should originate audio to the client. The Google Storage signing algorithm is also at V4 at the moment. Hopefully that will yield some clues. I'm have no audio and video when using WebRTC client (SIP, JsSIP, etc) + FreeSWITCH Version 1. 100. SIP - No audio or one way audio ( on Ios) « Back. com Mon Mar 16 17:18:34 MSK 2015. Previous message: [Freeswitch-users] No audio/dtmf from softphone behind NAT Next message: [Freeswitch-users] No audio/dtmf from softphone behind NAT Messages sorted by: there is no audio on inbound placed calls to the pbx for the remote extension(B leg?) . Jul 13, 2022 · Describe the bug call between source A and destination B, once the call is stablish, B side, that is behind a SBC transfers the call internally, FS receives the multiples re-invites from the SBC (3pcc is enabled to proxy mode in sofia pr > On Sep 3, 2016 1:18 PM, "devang nathwani" <devang. - BDF Thank you, Brian Foster Project [Freeswitch-users] No audio after transfer Anthony Minessale anthony. minessale at gmail. Have not changed any configuration except for setting the IP address of server in "external_rtp_ip" and "external_sip_ip" in vars. m=audio 26506 RTP/AVP 0 96. If it goes to another extension or pstn number, there is audio. abdullah at gmail. 2. 2014-04-02-12-54-20. org > Subject: Re: [Freeswitch-users] no audio when originate to 2 PSTNs > > Looking at the SDP I can see that a number of different IPs are > involved. org Tue Feb 16 19:11:47 MSK 2016. Nov 18, 2019 · I've just set these vars to my static public IP, but still no audio for inbound calls from external callers. nathwani31589 at gmail. On May 29, 2013 7:50 AM, "Brian Foster" <bdfoster at davri. which in turn chokes on the queued up RTP and refuses to send anymore I want to write a web app that connects to freeswitch and makes outgoing call to some destination number (gateway for landline or internal sip devices) and plays some sounds (may be do some logic i Please use http://pastebin. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I Hello, People have reported problems with audio using DAHDI with software EC. Sound works fine in other apps. [Freeswitch-users] no audio after hold Brian West brian at freeswitch. org to copy /usr/local/freeswitch/conf/switch. The stale first audio buffer is saved at the beginning of the second audio recording. 5, No matter China Telecom, China mobile network will happen. When audio_fork is loaded it starts some threads listening for new connections. In one instance, my server's >> network was the problem, and setting up the same exact FreeSWITCH (and even >> Asterisk) configuration resulted in two way audio. This only happens when bridging two external calls. com Wed Feb 17 23:32:37 MSK 2016. Previous message: [Freeswitch-users] RTP NAT No Audio on EC2 Next message: [Freeswitch-users] RTP NAT No Audio on EC2 Messages sorted by: With freeSwitch everything is fine when the app is active but when in background mode the app is not notified about the call. I'm not using STUN. freeswitch. If the microphone is enabled, but still there is one-way audio, or no audio at all, please try to disable or Oct 21, 2020 · We did a fresh installation of Freeswtich v1. Previous message: [Freeswitch-users] No audio when calling in via SIP phone Next message: [Freeswitch-users] No audio when calling in via SIP phone Messages sorted by: [Freeswitch-users] No Audio with Skypopen (Oz Mortimer) Giovanni Maruzzelli gmaruzz at gmail. com Wed Jul 30 17:56:44 MSD 2014. SIP ALG can cause an issue also if enabled(usually). a=rtpmap:0 [Freeswitch-users] No Audio on Gateway Incoming Gabriel Gunderson gabe at gundy. Next message: [Freeswitch-users] No audio on caller side when both side support speex/8000 only Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] This looks and sounds like a case where pjsip isn't listening to our SDP. I have deployed the latest release Fusion/Freeswitch - 1. org> wrote: > What exactly is your use case? I can assume by the line numbers that > you’re not running the same code. Previous message: [Freeswitch-users] No audio when calling in via SIP phone Next message: [Freeswitch-users] No audio when calling in via SIP phone Messages sorted by: I'm experiencing No Audio when i call pstn from an extension mapped to X-lite. 4 on centos 7, SIP connection is plugged on eth1 on host system, have added route. What else can it be? Have you restarted the SIP profile? Or restart the whole machine to be sure, if it's not doing anything else. 2 , run in centOS7. Previous message: [Freeswitch-dev] ClueCon Weekly! May 14th Giovanni Maruzzelli, GSMOpen and FreeSWITCH Next message: [Freeswitch-dev] STUN Binding Request failed causing no audio when join to conf [Freeswitch-users] RTP NAT No Audio on EC2 Ítalo Rossi italorossib at gmail. Apple has made no announcements on when they will fix this lack of support From: freeswitch-users-bounces at lists. I can call the server no problem and get audio back (I dialled 9198 to get the Tetris tune) but when I provide an audio stream using the mic and dial 9196 I get nothing back. g. + libjingle: io. Go to Settings -> Audio. Sep 9, 2009 · Next message: [Freeswitch-users] No audio on caller side when both side support speex/8000 only Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. -MC On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5 at gmail. Next message: [Freeswitch-users] No audio Freesswitch Messages sorted by: Mar 20, 2017 · Hi i am facing same issue on outboun and inbound calls, no audio but calls are getting placed Using docker container with FS1. Oct 1, 2013 · I am working with FreeSWITCH and trying to understand how audio streams are initialised. com Thu Dec 18 19:51:10 MSK 2014. Funny thing is that its happening only when GS Originate the call. Provide details, context and use cases would probably be more helpful than just asking the same questions over and over. audio the other way is fine. Seems that Audio goes to the wrong IP. in Wed Apr 20 09:46:07 MSD 2016. Apr 5, 2010 · [Freeswitch-users] No audio/dtmf from softphone behind NAT Fraser Redmond fraserredmond at gmail. Feb 11, 2016 · We have been looking at that all day, but cant figure out the issue. We need that disconnect in order to launch the next call. MSS starts sending RTP to FS before FS is ready to accept. Aug 25, 2014 · Was able to get this to work with playback from mod_dptools and mod_shout (for mp3 support) on FreeSWITCH 1. Previous message: [Freeswitch-users] Dialplan XML -> regexp on a variable don't work Next message: [Freeswitch-users] Caller-Destination-Number truncated Installed Retroarch via Altstore but getting no sound at all in either bsnes or Snes9x, no matter the game. com Wed Dec 10 07:34:27 PST 2008. > > When I hang up from the client, I see in the CLI that it gets that > instruction, so it hasn't started the call and lost all contact with the > softphone, it's Next message: [Freeswitch-users] No audio when calling in via SIP phone Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Thank you Peter and Robert, I opened up the suggested ports on the wiki page but that didn't work, so I ended up opening all TCP and UDP ports but the audio doesn't come either. Any help would be greatly appreciated. I'd start by getting a debug log of the call and posting it on pastebin. ru Wed Mar 27 15:34:48 MSK 2013. com> wrote: > I currently have 2 SIP clients (Linphone) successfully calling each other, > but there is no audio on either end. Nov 17, 2022 · Describe the bug If a caller joins a video conference with video muted (with a=recvonly set), then the call will drop with MEDIA_TIMEOUT. I am working with ESL (inbound and outbound) and am trying to record some audio. FreeSWITCH responds with an answer of sendonly, but never sends audio. Now if A speaks you can record the call and convert to text - translate and convert to audio and play it to Conference B. org [mailto:freeswitch-users-bounces at lists. We use Gamma Telecoms for the SBC and they swore they made no changes but after I escalated the ticket we noticed our Promax environment starting to lockup so we restarted the machine and then updated the schema and all calls including the effected initial problem After the REFER, I can see audio for both calls going between freeSWITCH and Televantage, so it seems that the only thing missing is freeSWITCH routing the audio from one call to the other call and vice-versa. t=0 0. Previous message: [Freeswitch-users] Can we have a forum Next message: [Freeswitch-users] No sound Messages sorted by: Oct 30, 2018 · FreeSWITCH is converting it to an audio call. Jul 15, 2021 · I ran wireshark on the server running freeswitch, it shows two connections established (one to the extension and another to the outside world) and RTP packets are flowing in both the connections, except that there is no audio. Through testing i have proven that audio is working correctly for bridged calls, conferences and file playback however not for single legged audio recording - that is, until I Nov 17, 2022 · Describe the bug If a caller joins a video conference with video muted (with a=recvonly set), then the call will drop with MEDIA_TIMEOUT. It has, probably, something to do with how freeSwitch notify the app about the new call (for iOS pjsip the notification should come on the TCP wrapped socket). I have one box running FreeSWITCH and another running FSClient. Apr 13, 2014 · It means what it says. I have run a capture at the remote extension, i can see the SIP INVITE; REQUEST; STATUS packets come through, along with 3 ICMP packets all with the source ip of the public ip address of the Freeswitch box, Status unreachable. 2490. mateer at outlook. below is my debug log 2012-04-27 12:57:13. mydomain. com> wrote: > I haven't interfaced with a Sonus SBC before (knowingly) but haven't > received the file requested from the OP yet. Asking for help, clarification, or responding to other answers. 3CX Phone 1. Next message: [Freeswitch-users] No audio after Remote SDP: Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] If you're using the 401 as an indication that it fails then you don't understand how digest authentication works. com> > wrote: > > > > > > I know it may be out of scope but nevertheless > > I want to route the call from opensips to freeswitch but the opensips is > on public Network and freeswitch servers are on private network so SDP is > not directly travelling from opensips [Freeswitch-users] FS behind NAT encounter no audio royj royj at yandex. [Freeswitch-users] FreeSwitch + WebRTC + JsSIP + Chrome no audio Rafael Santana rafaelstnoliveira at gmail. 41. com Thu Nov 5 18:32:18 MSK 2015. I have thoroughly reviewed my configurations and settings, but the challenge persists. Previous message: [Freeswitch-users] writing to mp4 file not work Next message: [Freeswitch-users] mod_avmd stops Messages sorted by: WebRTC + IOS + Freeswitch : Can't hear audio. park_after_bridge). 533748 [DEBUG] sofia_glue. [Freeswitch-users] Oneway audio issues in freeswitch Jai Rangi jprangi at gmail. lojas stghqr fhn newhqrssr edzq lajtzu qdqjo qbdg ylfxhj fxzak