Peercall drop issue with asterisk sever

Hi guys,
Im using Asterisk 1.8.11-cert4 , am a little bit new to asterisk, i had configured asterisk server on cloud and using SIP line from in my small callcenter, there is only about 15 agents, my issue is on some time all of my peers are showing like bellow given error message, and the call will be disconnected on that time
NOTICE[1161]: chan_sip.c:26661 sip_poke_noanswer: Peer ‘2004’ is now UNREACHABLE! Last qualify: 1116
and i had configure asterisk with the option in sip.conf as qualify= yes
kindly advice me if i need to be more specfic on this issue
pls help me

This is the wrong forum for support questions.

You basically have network quality problems. Your choices are:

  • fix the network;
  • turn off qualify;
  • set an achievable round trip time for qualify; or
  • live with the current situation.

Addressing the network quality problem might amount to prioritising SIP (and RTP) traffice at both your end and the ISP end.

Hi David,
Before implementing this system i was using vocalcloud for phone purpose, on that time i didnt had any issues like this
and please advice me on how to give a better vallue for qualify parameter in sip.conf instead of turning off of using default values


Find the 99 percentile round trip time in milliseconds, and set qualify to that. Note that the default is 2 seconds, which already represents an almost unusable connection.

Also note that the problem may be lost packets, which can only really be fixed by reparing the network.

Hi David thanks for your reply,

But actually am confused, about the last post

;--------------------------- SIP timers ----------------------------------------------------
; These timers are used primarily in INVITE transactions.
; The default for Timer T1 is 500 ms or the measured run-trip time between
; Asterisk and the device if you have qualify=yes for the device.
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
; Defaults to 100 ms
;timert1=500 ; Default T1 timer
; Defaults to 500 ms or the measured round-trip
; time to a peer (qualify=yes).
;timerb=32000 ; Call setup timer. If a provisional response is not received
; in this amount of time, the call will autocongest
; Defaults to 64*timert1

this is the round trip section in my sip.conf

You have to measure them empirically. However, the default is sufficiently high that I think your only solution is fix the network problem, not to try and hide the symptom. Also, if you are losing a lot of packets, increasing the timeout won’t help.

Hi David , thanks for your support…
and how can i mark this thread as “solved” ?

i have one more doubt , iam running asterisk on a server which is having bellow hardware can you please advice me for the maximum number of extensions that i can use with this

Intel® Xeon® CPU E5645 @ 2.40GHz
MemTotal: 1711932 kB(1.7 GB)

Also want to know that the average usage of bandwidth with one extension…

Tanks &Regards