Vicibox11 - IAX Issue

[sandan]

We recently moved to Vicibox 11, apart from initial module load issues (which was fixed later) everything seems to be working well and also noticed AMDs also got better.
Our setup is; multiple standalone dialers with an Asterisk (FreePBX) server that handles the endpoint registration so there is a flexibility for users to jump into any dialer without changing endpoint SIP settings, however carrier settings are defined in the dialer; Asterisk connection is just for the SIP endpoints and nothing else. The dialers are connected using the IAX trunk with the Asterisk. It was all ok, till the time we moved to Vicibox 11 from Vicibox 10. No audio, not even the connection call "you are the only person.." or ring. Had to switch to SIP to interconnect Asterisk with Vicibox dialers - this works. Very strange, so was checking if anyone noticed similar issues. Had look at Asterisk 16 IAX issues and all, couldn't find anything; mostly refers to NAT but this even doesn't reach the router/firewall; it's within the LAN itself. The IAX that hadles audio sound board etc within the dialer works perfectly; not an issue there.

Account Entry:

Code: Select all

[Asterisk]
host=192.168.1.X
type=friend
username=1XXXX
fromuser=1XXXX
secret=XXXXXXXXXXXXXXXXXXXXXX
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=g729
insecure=no
qualify=yes
dtmfmode=rfc2833
context=from-internal

Dialplan Entry:

Code: Select all

exten => _1[01]XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _1[01]XX,2,Dial(IAX2/Asterisk/${EXTEN},${CAMPDTO},tTkK)
exten => _1[01]XX,3,Hangup()

[carpenox]

Are the iax tunnels running? What about ip_relay?

[sandan]

Are the iax tunnels running? What about ip_relay?

Yes, they are. Nicely registered and peered up, 3 dafault ones and 1 with this Asterisk server. ip_relay working too.
The same configuration works with other dialer running Vicibox 10, if that helps.

[callforamerica]

Having same issue fresh vicibix 11 install. Trying sip now for outbound trunk to test. Iax drops intermittent and never ring back.

[ambiorixg12]

What do the Asterisk logs show related to this connection?

The dialers are connected using the IAX trunk with the Asterisk.

[callforamerica]

Okay, I have successfully changed my outbound termination from an IAX trunk to a sip trunk and now everything is working as expected including three-way calls and park customer on hold. While making a third party call previously using the IAX trunks like I've always used, I was seeing drop calls. The system was saying that the call was still ringing when it was connected and then a call logger would kick in and disconnect the call 45 seconds in and I also was not able to do three-way calls while using the IX trunk. So if anybody else is having this problem, I don't know what the issue is but 100% changing to a sip trunk did fix it for me as well. Confirm