Call Transfers - Not going to Voicemail

I am running with Asterisk 13.9.1 on FreePBX. Whenever an incoming call is received, then that call is transferred to another internal extension (B), the caller hears silence, and if extension B does not pick up, the caller is not transferred to voicemail. The extension just rings for about 1 minute or more, and then disconnects the caller without ever going to voicemail. This only seems to happen on calls that are transferred. I captured what happens on the console when a call transfer is initiated.

Connected to Asterisk 13.9.1 currently running on Server1 (pid = 1622)
  == Extension Changed 416[ext-local] new state Hold for Notify User 415 
    -- Started music on hold, class 'none', on channel 'Local/[email protected];2'
[2016-07-22 15:58:35] WARNING[9765][C-00000fae]: format_wav.c:149 check_header: Read failed (type)
[2016-07-22 15:58:35] WARNING[9765][C-00000fae]: file.c:472 fn_wrapper: Unable to open format wav
[2016-07-22 15:58:35] WARNING[9765][C-00000fae]: res_musiconhold.c:361 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh/.nomusic_reserved/silence': No such file or directory
    -- Stopped music on hold on Local/[email protected];2

you need to increase the verbosity of your logging, from the asterisk CLI

core set verbose 3

1 Like

Thanks. So I just tried this again only transferring calls from a soft phone extension. When transferring from the soft phone (Jitsi), it works properly and the call is routed to voicemail. So now it only seems to happen when transferring from a Yelling W52P / W52H phone. I wasn’t aware that a phone could cause this to happen. Any ideas?

There are no mind readers here, no log, no comment possible :wink:

Sorry, it was at a customer location and I couldn’t transfer calls unless someone was in the office. I set the verbosity as you mentioned, but I has already had the verbosity at 6. I increased it to 8 and called in. Then had my call transferred to another extension. After 7 rings, it went to voicemail! So it appears to be fixed now.

How did I fix it? Restart the server :\ (restarting asterisk did not help, but restarting the entire box did)