SOLVED Outgoing calls disconnect after 5 minutes exactly

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CPav

CPav

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Hi all, I'm hoping someone can weigh-in with some SDP/RTP knowledge here.

So I configured some phones for testing over the weekend and delivered that system to a client today. I recall having an issue back when I started testing where a call would not make it past 5 minutes. So over the weekend I tested this and I placed a test call for 7 odd minutes, happy with that. So the test location and the client both use the same ISP and hence the same broadband router, no extra port forwarding or the like.

My PBX sits behind a pfsense firewall, I have adjusted the UDP timeouts to beyond 5 minutes, and I have also tried a regular TPlink router between my PBX and my phone...so no pfsense in between.
I have nat-options-ping enabled on my fusionpbx.
I have tried from a Yealink T19(configure with Manual NAT and the router public facing address), a Cisco 7940 and my Linphone app.
I've tried from different extensions and to different destinations which even utilize different upstream voip providers.

If I connect directly to my PBX LAN there is NO issue, but either side of my pbx, either on the tplink or out on the public internet crossing over any form of NAT outgoing calls fail at 5 minutes exactly. Incoming are fine and go beyond 5 minutes without issue.

So I'm almost certain this is just a NAT issue, I'm pulling my hair out here trying to understand this, I've rebooted my server, tried toggling NDLB and SIP-FORCE-Contact with no difference.

Any advice PLEASE??

Edit- Just to add to this, I'm seeing a normal_clearing_hangup when the line terminates.
 
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CPav

CPav

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Just using a standard
tcpdump -s 3000 -w ~/capture-file.pcap port 5080

Or do you need a port range? I will do this tonight and revert.

Tx
 
DigitalDaz

DigitalDaz

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Code: 
tcpdump -i eth0 -n -s 0 port 5060 -vvv -w yourcapturefile.pcap

you can use port 5060 safely as that's going to be your destination port. I'm sure
 
CPav

CPav

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DigitalDaz said:
Code: 
tcpdump -i eth0 -n -s 0 port 5060 -vvv -w yourcapturefile.pcap

you can use port 5060 safely as that's going to be your destination port. I'm sure
Click to expand...
@DigitalDaz I've never used pastebin before, can I PM you the pcap? If I try and copy the pcap to txt I get many odd characters
 
DigitalDaz

DigitalDaz

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It looks like sessions timers are enabled on one of the boxes, try and disable them
 
CPav

CPav

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DigitalDaz said:
It looks like sessions timers are enabled on one of the boxes, try and disable them
Click to expand...
Thanks @DigitalDaz
It was set to false on my astpp and enabled true on my Fusionpbx, I have since set it to enabled true=false(then rebooted the server) on my fusionpbx but I've just tested it and it's doing the same thing....
enable-timer false True

Made an account on pastebin, please could you cast your eyes over my sip-profiles and let me know if you see anything that may be causing this issue.
FusPBX - https://pastebin.com/ynPkJGp0
astpp - https://pastebin.com/JFnPEZJd

Thanks for your time!
 
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CPav

CPav

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Thanks @DigitalDaz
As a test Ive configured a voip sub account and connected a gateway from Fusionpbx directly to this so as to bypass my astpp. Calls work perfectly. So looks like the issue lies between my astpp and fuspbx. In saying that though calls from accounts directly on the astpp work fine too...its just when the fuspbx uses astpp as its gateway.

When I make a call from fuspbx bypassing astpp as the gateway I recieve a "Processing updated SDP" message every minute, and the call doesn't cut after 5 minutes.

However when I set the the astpp as the gateway, when the call is initiated and up until the disconnect at 5 minutes I don't receive any of those SDP updating messages.

"
018-02-27 12:03:39.904521 [DEBUG] switch_core_media.c:4767 sofia/internal/700@fusion Set 2833 dtmf send payload to 101 recv payload to 101
2018-02-27 12:03:39.904521 [DEBUG] sofia.c:8061 Processing updated SDP
2018-02-27 12:03:39.904521 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/internal/700@fusion.
"
Why/What would cause the "Processing updated SDP" message to not transend the astpp? Looks more like the ASTPP is the issue here but I'm hoping you can help me here since your SIP knowledge far exceeds mine. I can only get around to doing a pcap later, and I will do one from the astpp as well.
 
CPav

CPav

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DigitalDaz said:
Can you do another pcap please and send it me.
Click to expand...
@DigitalDaz
DD, is there a particular way to configure fusionpbx to connect to astpp?
At present I've just created an account on astpp for my tenant on fusionpbx, created a gateway on the fuspbx tenant, I've tried setting it to register, and not register, but both options still fail the calls at 5 minutes. Do I need to set this a different way? I.E a Trunk or Gateway? I've looked at both options but don't see how to add the astpp account for billing purposes.

Any help guidance here much appreciated, tx again.
 
DigitalDaz

DigitalDaz

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You are not being caught here too by fragmenting UDP packets are you? Try setting the transport to TCP.
 
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