freeswitch

freeswitch

absolute_codec_string not working in dialplan

if of-hook agents in freeswitch can confirm incoming call from queue?

Call through 3 freeswitches

How can I rewrite SDP with switch_r_sdp in freeswitch?

freeswitch not accepting calls, tls invalid

How can I change the channel_variable like destination_number at run time (dynamically) in Freeswitch?

Lawful intercept in freeswitch

Freeswitch's `mod_dingaling` Google Voice authentication failure

Play say function playback file with G729 Format

How to put the session.id of caller SIP.JS (wss-binding) to CDR log file or in dialplan (Freeswitch Windows)?

How to detect voicemail on freeswitch?

Freeswitch converts 407 proxy authentication to 480 Temporary unavailable

Compiling Freeswitch source from master branch with mod_opus on debian jessie failed

originate conference from python esl

FreeSwitch Start Gives this Error: ==>freeswitch.service failed. See 'systemctl status freeswitch.service' and 'journalctl -xn' for details

Originate command with originate_retries in Freeswitch

Freeswitch API command problems

Freeswitch: mod_xml_curl and call groups

Connect internal numbers on freeswitch

Why onference_auto_outcall_flags doesn't work

Block annoying numbers on freeswitch

Freeswitch - mod_spandsp: Cannot send non-existant fax file

Freeswitch terminates with signal SIGSEGV, Segmentation fault

Recording Call streams separately in FreeSWITCH

Freeswitch : getting error while loading module mod_h323

Not able to install mod_perl in freeswitch 1.6.5

Freeswitch verto performance is low

FreeSWITCH filter registrations

bigbluebutton 0.9 drops meeting with people through voicebridge after last web user quit

freeswitch crash for libspeexdsp.so.1.5.0 error

Import variable from bleg to aleg‏

Connecting freeswitch remotely using node esl fails

freeswitch - group dialing, registration issue

Freeswitch bgapi originate command with ignore_early_media=true

Advanced fail2ban parameters in filter/action

Freeswitch performance improved after changing default password

Fusionpbx: How can I change Attended Call Transfer feature code from *4 to *2

How to modify the handling of a SIP message in Freeswitch?

Freeswitch update profile using mod_xml_curl

Access gateway/profile params/variables in Freeswitch

DTMF during playback in Freeswitch via ESL

Freeswitch allow any user to register

FreeSwitch Mod_Event_socket Events handling

SIP 408 Request Timeout for Freeswitch

Freeswitch prevent hangup

Transferring Calls between FreeSWITCH and Session Manager

Bridge an outbound call to a conference with freeswitch

Freeswitch pocketsphinx won't recognize me

Call Recording in Freeswitch

Bridge a call from sip phone to real phone using freeswitch

Freeswitch Originate and session heartbeat

Freeswitch ESL library

freeswitch enterprise originate - how to determine if all failed

Installing OPUS in freeswitch

How to Test the IVR or Dialplan of Freeswitch using Sipp

What's the difference between event's Unique-ID and Channel-Call-UUID?

MOD_OREKA configuration on Freeswitch

Freeswitch Conference Parameters max-members value limit

Bridge PSTN calls using Freeswitch

Access the name of agent who received the call in a Queue

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