Calls drop after 30 minutes

charlie4616 · October 20, 2017, 8:25am

Hi,

We have an issue whereby calls drop after 30 minutes (not all but most), I am now stuck with what to do. When a call is active and it hits 30 minutes, the remote end drops but the handset connected to my FreePBX still thinks the call is active. After a further 30 seconds the internal handset drops the call due to lack of RTP. When carrying out a packet capture I can see that the SIP provider sends an INVITE request at 30 minutes call duration (I am presuming this is a re-invite) and then the next packet is a BYE request again from the provider. No response from the PBX until 30 seconds later where it sends a BYE request with code 44 (channel not available).

Is the issue somewhere on FreePBX where it isn’t responding to the re-invite?

Any help would be appreciated.

Charlie

dickson · October 20, 2017, 9:58am

are these phones internal or external to your network. Sounds like a firewall closing not ports.
Try disabling SRTP if you are using it to see if that changes the behaviour

charlie4616 · October 20, 2017, 10:06am

Hi,

The phones are internal to the network. Does that have any bearing on whether SRTP should be disabled?

lgaetz · October 20, 2017, 12:01pm

Your provider sends an INVITE at the 30 minute mark, and your PBX doesn’t respond? Does the provider resend the INVITE repeatedly or is it just once? Do you know why the reINVITE is being sent? Thirty minutes corresponds to 1800 seconds which is the default session timer setting in chan_sip, so I suspect that the reINVITE is something to do with that. You may fix by tweaking the session timer settings for the trunk peer details.

github.com

asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L655


; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be > rtptimeout)
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
; (default is off - zero)


; -------------------------- SIP Session-Timers (RFC 4028)------------------------------------
; SIP Session-Timers provide an end-to-end keep-alive mechanism for active SIP sessions.
; This mechanism can detect and reclaim SIP channels that do not terminate through normal
; signaling procedures. Session-Timers can be configured globally or at a user/peer level.
; The operation of Session-Timers is driven by the following configuration parameters:
;
; * session-timers    - Session-Timers feature operates in the following three modes:
;                            originate : Request and run session-timers always
;                            accept    : Run session-timers only when requested by other UA
;                            refuse    : Do not run session timers in any case
;                       The default mode of operation is 'accept'.

charlie4616 · October 23, 2017, 8:13am

Your provider sends an INVITE at the 30 minute mark, and your PBX doesn’t respond? Correct

Does the provider resend the INVITE repeatedly or is it just once? Once only

Do you know why the reINVITE is being sent? No

It sounds like it could be session timers to me, would I have to WinSCP to the PBX, find the chan_sip configuration file and edit it, or can this be set via the GUI and where?

Thanks