I’m having issues with dropped calls where the calls could be new, waiting in the queue to calls that have been going on for several minutes. So far it looks like in each case I get an ACK request, followed by a BYE and then a 481 Call let/transaction does not exist. This appears to be the sequence of events, and it is happening intermittently, but often happens in groups of several drop calls right together.
The BYE shows “HangupCause: Requested channel not available” and the “HangupCauseCode: 44” is given.
System is 10.13.66-11 with Asterisk 13.8.2 loaded from Distro. I’m using Grandstream GXP-2160 phones SIP trunks.
Has anyone else encounter this same or similar issue?
This cause is returned when the circuit or channel indicated by the requesting entity cannot be provided by the other side of the interface.
Thank you so much for taking time to respond to this. The problem occurs sporadically, and most calls day in and day out work fine. Can you offer any guidance on how best to troubleshoot this issue, how to track it further, or find a resolution?
macjacfish - Did you ever identify the root cause of this issue. It appears that I having the same issue using Cisco 7942 handsets with an 8.5 SIP load on them.
Tested and duplicated by calling from an outside number (i.e. cell) to a DID/EXT and then placing that outside call on hold. After 5 minutes the call is dropped each time.
Thanks in advance for any assistance!
If exactly 5 minutes, it sounds as a timer hitting the timeout limit.
If using chan_sip, go to Asterisk SIP Settings and change RTP Hold Timeout as desired (the default of 300 seconds is 5 minutes).
If pjsip, I believe that you have to edit the config manually; see http://lists.digium.com/pipermail/asterisk-commits/2015-July/073583.html .
This turned out be to an internet provider issue for us…the node was oversubscribed and when peak traffic would happen calls would drop. Two things come to mind on this, one is some firewalls need keep-alives to stay open…and I’ve taken to putting 25 seconds in the rtp keep alives under asterisk sip settings and that has helped occasionally. The other has been that some routers implementation of the SIP ALG Layer have not been good…thinking Netgear here, and disabling that has also helped clean up issues…the symptom for the ALG issue can bee seen in a hosted environment when the extension qualifies times are way high…100ms and greater. This problem also will manifest itself when you go to transfer a call and it never makes it to the destination. Hope that helps!
Thanks Stewart1. The issue was resolved by increasing the RTP Hold Timeout to 1800 seconds. However I am going to try to do a bit more digging because the dropped calls do not happen with any of the Yealink handsets but rather only the Cisco 7942’s.
When I look at the packet capture between a Cisco 7942 and the PBX there is an OPTIONS packet being sent every minutes from FreePBX server to handset and the handset is returning a 200 OK but after the 5th OPTIONS and 200 the PBX is hanging up the call by sending a BYE. I do not see the same BYE with the Yealinks.
Thanks again for the assist!