One way call connectivity (NOT one way audio) issue in FreeSwitch


Please find my below environment:

Operating System: Debian 12
FreeSwitch Version: 1.10
Cloud Environment: Microsoft Azure
Firewall: Opened all the ports in Azure and also deleted iptables from the VM

I have installed FreeSwitch and successfully running. After installation, I put the public IP address of my VM in bind_server_ip, external_rtp_ip, external_sip_ip parameters in vars.xml file. I haven’t changed anything in other configuration files.

I configured the default extensions i.e. 1009 and 1010 in my soft phones and able to call the demo IVR extension i.e. 5000 and it is working.

Problem: I am able to call from 1009 to 1010 and it is connecting and able to hear the audio both sides. But, when I am trying to call from 1010 to 1009, call is not at all connecting and going to the voice mail after sometime.

Below is the FreeSwitch console log for your reference:
[DEBUG] switch_core_state_machine.c:40 sofia/internal/1009@ Standard INIT
[DEBUG] switch_core_state_machine.c:48 (sofia/internal/1009@ State Change CS_INIT → CS_ROUTING
[DEBUG] switch_core_state_machine.c:624 (sofia/internal/1009@ State INIT going to sleep
[DEBUG] switch_core_state_machine.c:581 (sofia/internal/1009@ Running State Change CS_ROUTING (Cur 2 Tot 7)
[DEBUG] sofia.c:7493 Channel sofia/internal/1009@ entering state [calling][0]
[DEBUG] switch_core_state_machine.c:640 (sofia/internal/1009@ State ROUTING
[DEBUG] mod_sofia.c:158 sofia/internal/1009@ SOFIA ROUTING
[DEBUG] switch_ivr_originate.c:67 (sofia/internal/1009@ State Change CS_ROUTING → CS_CONSUME_MEDIA
[DEBUG] switch_core_state_machine.c:640 (sofia/internal/1009@ State ROUTING going to sleep
[DEBUG] switch_core_state_machine.c:581 (sofia/internal/1009@ Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 7)
[DEBUG] switch_core_state_machine.c:659 (sofia/internal/1009@ State CONSUME_MEDIA
[DEBUG] switch_core_state_machine.c:659 (sofia/internal/1009@ State CONSUME_MEDIA going to sleep
AND AFTER some time, connecting to Voice mail.

Any help would be appreciated. Thank you.

Best Regards,


This could be a number of things. The log provided needs to be the complete call attempt. Something to help troubleshoot is SNGREP GitHub - irontec/sngrep: Ncurses SIP Messages flow viewer
So possible issues:

  • Registration time needs to be set at around 120
  • Codec issue
  • SRTP issue with ciphers

Hello LenGraham,

Good morning and hope you are doing well. As you suggested, please find the enclosed sngrep output and complete log file. In the enclosed call log and sngrep log, call is able to connect when I am calling from 1001 to 1002. But, call is not connecting, when I am calling from 1002 to 1001. Please find sngrep output, complete call log, and configuration changes that I made at the below link:

Any help would be appreciated. Thank you.

Best Regards,