I was wondering if anyone here can help with strange issue I have been experiencing. It was reported that when you call particular DDI call doesn’t ring on the phone and goes directly to voicemail. Phones are Cisco 6941 but the most interesting part is some of the DDIs actually ring on phones. We have single service provider with couple of DDIs and just for a test, I modified extensions.conf to be as below:
[provider]
exten => s,1,Dial(SIP/$4800,25,Tt)
When I call 1st DDI, the call goes directly to voicemail. When I call another DDI, it rings with no issues. I captured the traffic with tcpdump and there are no differences in invites, etc. The only one is that 1st call shows the phone 180 ringing but the second doesn’t. Also, CLI doesn’t show much difference except one extra line for ringing:
== Using SIP RTP CoS mark 5
-- Executing [s@provider:1] Dial("SIP/provider-00000031", "SIP/4800,25,Tt") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/4800
== Spawn extension (provider, s, 1) exited non-zero on 'SIP/provider-00000031'
[Dec 12 18:16:37] ERROR[22]: cdr_csv.c:316 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
== Using SIP RTP CoS mark 5
-- Executing [s@provider:1] Dial("SIP/provider-00000033", "SIP/4800,25,Tt") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/4800
-- SIP/4800-00000034 is ringing
== Spawn extension (provider, s, 1) exited non-zero on 'SIP/provider-00000033'
[Dec 12 18:17:02] ERROR[22]: cdr_csv.c:316 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
Tried that but doesn’t help much. I can only see 100 trying then cancel from the phone I called from so the cancel initiated by the originator and coming in over the SIP trunk. I mentioned I cannot see ringing in original post but when the voicemail greeting is played and I wait for few more seconds there is also 180 ringing coming from phone. So 100, 180 and then cancel. I can see no clue and nothing in between these messages. I’m starting to think that it maybe caused by some internal logic on Cisco 6941 phone so the only way is to log into the phone with SSH and try to run some debugs there. I have no access now but will keep you posted once I have it and get any further findings.
I agree so I am curious why I cannot see anything in the console indicating taking the call from phone and pushing it out to the voicemail. It’s strange and never seen it before but it looks like phone just follows its own logic and doesn’t communicate back to asterisk.
Even more strange thing now cause using client other than Cisco phone doesn’t change anything. If the invite from service provider looks different for those two numbers dialled I would look there but since both calls come into asterisk same way, over the same trunk I am run out of ideas!
That’s the whole dial plan for testing purposes. I will capture SIP with “sip set debug on” later and provide traces but really can’t see anything that would point me into right direction as all INVITE, TRYING, RINGING are there and for some reason it goes to voicemail which I’m not sure where it’s played from as CLI doesn’t show anything about the voicemail.
Being installed on dedicated server before it was all working. Now guys I did that for in the past moved it to docker and then it started behaving like this. They just built the same version from source and they moved all the configuration files and it’s not working. I have almost no knowledge regarding the docker so not sure if there is something they might have missed, etc.