How to play constant background music on one leg during a call
I'd like an incoming caller to hear constant (low volume) background music while speaking with the called party. The called party (leg B) should not hear the music. The music will come from another extension, for example, the tetris tone_stream extension
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$5 Trial Mode SingalWire
Hi can anyone please let me know that i am using trial mode of singalwire without deposit but haven't received single call from couple of months so my concern is that deposit is necessary to start working smoothly??
π£ January FreeSWITCH Office Hours
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+sip.instance
Hi, I use FreeSWITCH Version 1.10.11-release~64bit (-release 64bit) and FusionPBX and have a problem to set FreeSWITCH to take in account Contact parameter: +sip.instance="<urn:uuid:XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXX>" in REGISTER try multiple-registrations
NAT Problems between External Profile and Internal Profile (in a Container)
Hi all! This is my first question to this board. Please let me know if I am breaking any taboos or form. I have an installation of Freeswitch/FusionPBX. The wrinkle which makes this a bit of a bear is that the install is running inside of a Podman Container.
Fax2Email setup - Freeswitch is not accepting T38 in reinvites
Hi, Origination carrier sent this invite in SDP v=0 o=NN 497024071 1 IN IP4 192.10.16.121 s=- c=IN IP4 55.55.55.55 t=0 0 m=audio 5000 RTP/AVP 8 0 Freeswitch sent 100 trying followed by 200OK then origination sent this v=0 o=NN 497024071 2 IN IP4 192.10.16.121
How to disable symmetric RTP in freeswitch.
I'm having trouble sending video to FreeSWITCH from a source port that is different from the one specified in the SDP. Even though I enabled disable_rtp_auto_adjust, FreeSWITCH still sends video RTP to the port defined in the INVITE SDP. However, unless
Freeswitch 481 Response to BYE
We are experiencing a common, yet problematic, SIP dialog flow between our Freeswitch (v1.10.10) and the Zoom SBC, which frequently results in hanging calls. We are seeking insights into why Freeswitch fails to terminate the session properly on the first
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Problem installing FreeSWITCH on Debian β β401 Unauthorizedβ error from SignalWire repo
Hey everyone, Iβm trying to install FreeSWITCH on Debian, but Iβve run into an issue. Previously, I successfully installed it using this guide: FreeSWITCH Installation Guide on Debian 12 apt-get update && apt-get install -y gnupg2 wget lsb-release Hit:1
Freeswitch conference moh
I am running freeswitch in 3pcc. My SIP Server as call controller creates conference in freeswitch using ESL. I have to place one leg on hold (re-invite with a=sendonly). so far good. However, I am unable to play MoH on the leg which is placed on hold.
Domain differentiation
Hello, I want to manage my users by tenant (domain). The problem is that in the register, it understands that the domain is present in the XML curl send, but But if in my XML the domain name= is not the IP, then it refuses, saying: sofia_reg.c:3210 Can't
FreeSWITCH Enterprise v20.25.4 (Oct. 27, 2025) β Release Announcement
Weβre excited to announce the availability of FreeSWITCH Enterprise v20.25.4, the latest version of our commercial-grade communications engine. This release focuses on critical bug fixes, enhanced monitoring capabilities, and new low-level APIs that empower
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Freeswitch get stuck
After a certain period the freeswitch CLI get unresponsive and all INVITES are getting timed out. This behaviour seems not to be related on load as it happened also eraly in the morning where only few calls are on the system. Freeswitch is running on
Codec L16@8000h support
im working with FreeSWITCH Version 1.10.10-release+git~20230813T165739Z~4cb05e7f4a~64bit im trying to get FS to do transcoding between codec L16@8000 on inbound leg to PCMA on outbound leg, I've tried several configurations without any success, I continue
issues in loading mod_audio_fork?
Hello Team, I am trying to intercept call between two users and apply AI/ML processing for the live audio stream. I found out that mod_audio_fork module should be used for that. I installed this module but when I load this module using "load mod_audio_fork"
Learning mod_xml_curl
I'm learning about mod_xml_curl. My application appears to be working but the fs_cli view is showing a lot of 404 errors. My app logs a request with no destination number, cannot fulfil it's request and returns a 404. fs_cli shows that it's looking for
Handling dialplan with mod_xml_curl based on context
I am using mod_xml_curl. I is possible to handle dialplan with this module only for default context, otherwise handle it with static xml files? I want that public context is handled with mod_httapi. Now I return mod_xml_curl dialplan with httapi action.
Presences (BLF) not working with IPv6
Hello everyone, We need some help with presence (BLF) with IPv6. For testing purposes one of our Freeswitch servers is running ipv6 exclusively. We can register extensions and in-/outbound calls are working. But when we Subscribe for an extension, Freeswitch
π£ FreeSWITCH Office Hours β August 2025
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Important Update: SpanDSP Package Issue Resolved
# Important Update: SpanDSP Package Issue Resolved **Date: July 16, 2025** We have identified an issue with the latest SpanDSP package, which has been causing crashes in FreeSWITCH. To address this, we have removed the broken SpanDSP Debian packages from
Is there a module in FreeSWITCH (from GitHub) that allows programming bit transcoding?
Hello everyone I need to find a module that can be programmed using the FreeSWITCH source code available on GitHub. This module should be able to perform bit transcoding. I need to convert audio to 24 bit / 48 kHz. Thanks.
FreeSWITCH error handling of UNENCRYPTED_SRTCP parameter
We are working with an Avaya customer system that uses the UNENCRYPTED_SRTCP parameter from RFC 4568. https://www.rfc-editor.org/rfc/rfc4568.html#section-6.3.2. The customer is not able to remove this parameter from their end and so it is appended to
node module to help writing SIP functional/integration tests
Over the years I have been slowly writing a node module to help me write SIP functional/integration tests: https://github.com/MayamaTakeshi/sip-lab I have been adding new features as I needed for new kinds of test but didn't have time to document it.
FreeSWITCH is shutting down my kubernetes pod after starting six 1-on-1 video conferences
During load testing, my FreeSWITCH instance consistently struggles to handle more than six 1-on-1 video conferences, often leading to performance degradation or service instability. Is this level of performance typical, and what factors might be limiting
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Updating RTP Destination from SDP in 200 OK (Call Answer)
Scenario: FreeSwitch acts as a simple SBC, connecting internal SIP hosts to external services (via the internet). The external service receives the call request and forwards it. Upon answer, it sends a 200 OK back to FreeSWITCH with updated RTP details
FreeSWITCH Enterprise v20.25.2 (June 18th, 2025) β Release Announcement
Weβre excited to announce the latest release of FreeSWITCH Enterprise: v20.25.2 is now available! This release brings significant security updates, stability enhancements, and powerful new features designed for production environments. π§ Enhancements
How to install `sofia-sip` and `spandsp` modules?
The FreeSWITCH docs mention that they were split out from the FreeSWITCH repo and that "Packages for Sofia-sip and SpandDSP are available for all supported platforms from our packaging repos.", but couldn't find more info on this. I searched the Debian
Directory Gateway Register behavior
Hello Everyone, I need to handle users from Internet trying to register to one or more PBXs inside a private network and I'm trying to do it with freeswitch. So I need to send a REGISTER to a private PBX when someone tries to register to my freeswitch.
Why was every(?) `apr_` prefix renamed to `fspr_` in lib/apr? (commit 5c2726f)
I came across commit 5c2726f "[core] rename lib apr symbols to fspr" while diving into the internals of FreeSWITCH and figuring out why the generated Doxygen docs don't point to the correct definitions for the most basic functions (e.g., `switch_mutex_lock`).
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How to configure freeswitch to operate in proxy mode for accurate call billing?
Our freeswitch cdrs billed seconds is always lower than terminating partners billed seconds 20% of the time and at the same time higher than customers cds 15% of the time. Our Investigation reveals that the issue is with freeswitch B2BUA mode of operation.
How to increase NAT Packet quantity on freeswitch
Hi, I am new to FreeSwitch. I have installed FreeSwitch on Google VM. It is working h264 with camera devices. But it was not working with H264 without a camera. The video call is not working, the manufacturer send this message (In our device, for NAT
SMS
How to configure sms with a sip provider to send and receive sms messages with phone numbers. I want to use groundwire sip simple. Will it work with voip.ms. I can send and receive sms between freeswitch extensions.
Conference and Transcoding
Hello all, new to FreeSwitch, I hope this is the correct spot to ask this.. I am trying to use "conference" in my dialplan. Is there a way to use "conference" without transcoding?
Call recording in transcoded call
When FS is recording a call where the bridged legs are transcoded, the recording is not consistently intelligible. Often, one or both of the call legs are at 2x-4x the speed of the other leg during the recording. This especially happens when a call is
Is there a public online documentation for FreeSWITCH C internals?
Came across the Debug Printing section in the docs where all info links point to https://docs.freeswitch.org/ (e.g., https://docs.freeswitch.org/switch__swigable__cpp_8h.html), but it doesn't seem to be publicly available anymore.
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