Solved: VICIdial Disconnects calls after 15min

[mav2287]

For some reason calls that are going longer than 15 minutes seem to get automatically hung up. I made a call directly through asterisk on a soft phone to bypass VICIdial and it did the same thing. I am have not been able to catch it on the asterisk CLI but I will post once I can catch it there. Has anyone seen or heard of this?

EDIT
I did manage to catch it on the CLI and there was nothing. It didn't show any kind of output at all. The only thing I could think of was that it is behind an NAT device. I added NAT=yes to my trunk config and I am hoping to solve this. Not sure if anyone else have any idea of what it could be.

Also this is exactly at 15 minutes almost on the dot.

[ivschakravarthi]

Check the RTP settings in the sip.conf and in your softphone under preferences menu.

[geoff3dmg]

Does the NAT device have any 'features' that may interfere with the RTP traffic. For instance, I had a thompson speedhub firewall at one site that would block incoming calls if the 'DDoS protection' was enabled. Likewise I've had Draytek routers entirely fall over at 6 to 8 concurrent RTP streams when the 'Accept large incoming fragmented UDP or ICMP Packets' firewall feature is enabled. The quickest way to rule any NAT issue out is to temporarily connect the Vicidial server directly to the Internet.

[mav2287]

I think it may have to do with the router itself. I hooked up a magic jack just to see what it would do and got the same result. It's a CISCO so it is a decent NAT but I can't find anything that would affect the traffic. I did however notice that it doesn't have anything for SIP not even in the QOS settings. My thought was to replace it with a NAT that had SIP settings any thoughts or Routers that work well with VICIdial?

[geoff3dmg]

I'm currently using a sonicwall without issue.

[DomeDan]

Enable sip debugging and look at the asterisk CLI again when the call gets hung up

[mav2287]

I enabled the debug mode and logged it. The phone line I called hung up at 15.00 min dead on. The SIP phone took about 30 seconds to figure out it had lost connection the cli gave me this at 30 secs past 15

Code: Select all

<------------>
[Jan 28 09:15:30]     -- Executing [h@default:1] AGI("SIP/201-00000063", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----974-----961") in new stack
[Jan 28 09:15:30]     -- <SIP/201-00000063>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----974-----961 completed, returning 0
[Jan 28 09:15:30]   == Spawn extension (default, 91913593XXXX, 2) exited non-zero on 'SIP/201-00000063'
[Jan 28 09:15:30] Scheduling destruction of SIP dialog 'ZGE2ZGM1MjgzMDA2NThmZDhmZTdkOGU3YWMyYzRlODk' in 6400 ms (Method: ACK)
[Jan 28 09:15:30] set_destination: Parsing <sip:201@XX.XXX.XXX.XXX:50764> for address/port to send to
[Jan 28 09:15:30] set_destination: set destination to XX.XXX.XXX.XXX:50764
[Jan 28 09:15:30] Reliably Transmitting (NAT) to XX.XXX.XXX.XXX:50764:
BYE sip:201@XX.XXX.XXX.XXX:50764 SIP/2.0
Via: SIP/2.0/UDP XX.XXX.XX.XX:5060;branch=z9hG4bK3ee40e24;rport
Max-Forwards: 70
From: <sip:91913593XXXX@XX.XXX.XXX.XXX>;tag=as54ac222d
To: "1234"<sip:201@XX.XXX.XXX.XXX>;tag=f63a787e
Call-ID: ZGE2ZGM1MjgzMDA2NThmZDhmZTdkOGU3YWMyYzRlODk
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.24.0-vici
Proxy-Authorization: Digest username="201", realm="asterisk", algorithm=MD5, uri="sip:XX.XXX.XXX.XXX", nonce="", response="15124879c82dd070681470f56c602abe"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


[DomeDan]

Ok, cant see anything strange there.
did some googling and found this thread http://forums.asterisk.org/viewtopic.php?p=161452
but also more interesting this mail-list thread http://lists.digium.com/pipermail/aster ... 78255.html and someone talking about that the problem was their cisco router
I suggest you try update the firmware of the router

[mav2287]

Tried to update couldn't find anything new to update to. It is also at end of life so I am just going to replace it with a new router. Any suggestions on routers that work well with sip trunking?

[DomeDan]

I'm running PfSense on a HP DL320 G2, its been working good but I'm not NATing to the server.

Take any hardware and install PfSense on it and replace the cisco router, it the problem goes away then its the cisco router thats the problem.
Then you can look for any other router or stay with PfSense if you're happy with it, I've had no problem with it except the previous hardware it was running on (an old consumer pc)

[mav2287]

Got a new router and it solved the problem. Seems it was something to do with the router itself. We have now had hour long calls with no issues.