SIP phone ---- Asterisk —iax2— Asterisk ----- SIP provider
We have Asterisk/FreePBX systems at our office and hosting location. We recently changed the trunk between our local system and our hosted system to utilise IAX2 (trunk=yes) instead of SIP.
When a SIP phone dials a busy number the call terminates without playing the customary advisory tones although everything works when we change trunk back to SIP again. All other call functionality is working perfectly.
I found an Asterisk JIRA bug logged which matches our scenario (https://issues.asterisk.org/jira/browse/ASTERISK-15925?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel) but I can’t make out if the bug was closed because it was an actual confirmed bug in Asterisk or because the developers feel that FreePBX should first answer the channel, play the advisory tone for 20 seconds and then only hangup.
PS: The last comment on the Asterisk JIRA report states: ’ This issue is related to SWP-1275 [ SWP-1275 ]’ but I don’t know where to find information relating to this…
Any assistance would be greatly appreciated.