I have a desk phone (Digium A25) which works great and Zoiper on my cell phone which has it’s moments but works. I can make outbound calls all day and can use Zoiper to call my desk phone great. But my desk phone cannot call Zoiper, it ends immediately with a busy signal saying congested on the asterisk logs and what’s more is the incoming call screen flashes momentarily in Zoiper before closing and Zoiper reports a missed call.
Using Digital Ocean to power my asterisk server
-- Executing [110@internal:1] dial("PJSIP/desk-phone-00000005", "PJSIP/june-mobile,,hKTWX")
-- Called PJSIP/june-mobile
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'PJSIP/desk-phone-00000005' status is 'CHANUNAVAIL'
is june-mobile peer registered ?
-
CHANUNAVAIL : Channel unavailable. On SIP, peer may not be registered.
I can make outbound calls with it so I presume it’s registered, if it’s not registered wouldn’t I not be able to make outbound calls with it?
Furthermore I can make internal calls to the same devices just fine that can’t do the reverse and call june-mobile
Register means device is rechable by Asterisk, so when an INVITE is sent to the device Asterisk can locate it
How would I know if it’s registered or not? My phone says it’s connected just fine to the server and logged into it’s SIP account.
run this command and post the output
pjsip show contacts
Contact: desk-phone/sip:desk-phone@### ###7dce NonQual nan
Contact: desk-phone/sip:desk-phone@### ###7dce NonQual nan
Contact: june-mobile/sip:june-mobile@### ###c86 NonQual nan
Contact: june-mobile/sip:june-mobile@### ###c86 NonQual nan
Contact: ###Trunk Inbound### ###ae1 NonQual nan
Contact: ###Trunk Outbound### ###443 NonQual nan
Enable qualify on the aor section
qualify_frequency=5
I did that and restarted asterisk, can still make internal and external calls with zoiper but still can’t receive any calls
That command wont fix the issue it just tell us if the peer is rechable
run again pjsip show contacts
Contact: <Aor/ContactUri..............................> <Hash....> <Status> <RTT(ms)..>
==========================================================================================
Contact: desk-phone/sip:desk-phone@### ###dce NonQual nan
Contact: desk-phone/sip:desk-phone@### ###dce NonQual nan
Contact: june-mobile/sip:june-mobile@### ###b56 Avail 97.344
Contact: june-mobile/sip:june-mobile@### ###b56 Avail 97.344
Contact: ### Trunk Inbound ### ###ae1 NonQual nan
Contact: ### Trunk Outbound ### ###443 NonQual nan
If you turn up the logging sufficiently, Asterisk will tell you why it is abandoning the call. The fact that something flashes up suggests it is registered.
I’m running asterisk currently with
sudo asterisk -cvvvvvvvvvvvvvvvvvvvvvvv
That is correct I just didnt read the part it flashes up
You may need to enable pjsip protocol logging and you may need to enable the debug log (typically uncomment full in logger.conf), and to increase the debug level.
5060 is my servers SIP port, the default SIP port
I just got off work so I’ll go through the masses of debugging tomorrow but I hate this has gotten so complicated over a simple internal call but I get it happens.