Issue with some inbound calls

Having an issue on one of our Asterisk systems with calling a device, a Cisco SPA112 ATA.

Call comes in, sometimes it rings the device a couple times, stops then rings the device again. Othertimes, it doesn’t even ring the device. Both occasions the caller hears ringing. The device connected to the ATA is an Avaya phone system. (trying to get this customer over to a SIP trunk or to a hosted solution, but still need to figure this out.)

SIP datagrams below from VoipMonitor. Also, this only happens on one out of every 50 or 60 calls, but still happens daily. One call the caller said that all they heard was ringing on their end, but no ringing on the Avaya system (datagrams are from this call).

The extensions.conf for this is:

[customer]
exten => 5867XXXXX4,1,Dial(SIP/5867XXXXX4,50)
exten => 5867XXXXX4,2,Goto(5867XXXXX4-{DIALSTATUS},1) exten => 5867XXXXX4-BUSY,1,Dial(SIP/5867XXXXX6,50) exten => 5867XXXXX4-BUSY,2,Goto(5867XXXXX6-{DIALSTATUS},1)
exten => 5867XXXXX6-BUSY,1,Dial(SIP/5867XXXXX6,50)
exten => 5867XXXXX6-BUSY,2,Voicemail(5867XXXXX4@default)
exten => 5867XXXXX4-CHANUNAVAIL,1,Dial(SIP/5867XXXXX6,50)
exten => 5867XXXXX4-CHANUNAVAIL,2,Goto(5867XXXXX6-${DIALSTATUS},1)
exten => 5867XXXXX6-CHANUNAVAIL,1,Dial(SIP/5867XXXXX6,50)
exten => 5867XXXXX6-CHANUNAVAIL,2,Voicemail(5867XXXXX4@default)

Relevant sip.conf below:

[5867XXXXX4]
type=friend
secret=(Cleaned Password)
dtmfmode=rfc2833
host=dynamic
nat=no
callerid=Customer Name <5867XXXXX4>
mailbox=5867XXXXX4@default
canreinvite=yes
context=default-586
T38FaxMaxDatagram=38

[5867XXXXX6]
type=friend
secret=(Cleaned Password
dtmfmode=rfc2833
host=dynamic
nat=no
callerid=Customer Name <5867XXXXX4>
mailbox=5867XXXXX4@default
canreinvite=yes
context=default-586
T38FaxMaxDatagram=38

What do the IP addresses represent?

The trace shows an unanswered call cleared by the caller after about 50 seconds. Either only one leg was logged, or their was no B side. The called side requested ringback tone to be generated.

Normally, for chan_sip we would want sip set debug on output for the complete call, and logging at at least verbosity 5.

10.22.190.10 is the Asterisk server, the other is the ATA. There is an SBC involved and then Inteliquent. SBC is just between Asterisk and Inteliquent.

We pull data from Voipmonitor for asterisk, just a sniffer running on the asterisk box. B side datagram is below.

SIP-3

Every singnaling you show us is “normal” behaviour.

The last one is a call that is declined by the destination.

As David is advising you, enable the sip logging, or use ngrep, to caputre the signaling so you can first try and solve the issue yourself. If you are not able to solve it yourself then you can ask for help.

But I would advise that if you are not able to analyse the basic callflows that you have added to your post, you need to brush up on that knowledge.

I know, I may sound like a d**k, but just by looking at all the different questions that have been asked before on this forum you could already have found the information on how to solve your problem.