Freeswitch
I want to create predictive dialer using php with freeswitch, how to do that ?
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
$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??
+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
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
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 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
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
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
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.
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.
ESL Python
I am a Beginner . I need a help on these issue MY project make ai caller agent so i am setting up the sip call flow and using ESL to connect with the ai agent on runtime import ESL not working undefined symbol _ZN8ESLeventC1EPS_ also ESL.so and _ESL.py
Account Deletion Request
I would like to delete my SignalWire account and all associated data.