Call timeout

Hello,

we are using asterisk version 1.4.36.
We have a weird problem. After a non successfull call (like a very long number and such), the phone (Yealink-any) gets “call time out” and then loses it registration.

i know it is not a phone issue because i tried from the same phone but diffrent servers.
and in1 server it does work.

sip.conf:

[general]
language=en
rtpkeepalive=30
rtptimeout=120
srvlookup=yes
dtmfmode=rfc2833
rfc2833compensate=yes
toneduration=300
progressinband=yes
prematuremedia=no
cos_sip=4
cos_audio=5
tos_sip=cs3
tos_audio=ef
allowguest=no

I ran wireshark and noticed that the phone does not ACK to the 487 requst terminated, so the server is sendindgit multiple times.

fllow:

PHONE SERVER
----------------->INVITE
407 PROXY<--------------
------------------->ACK
----------------->INVITE
100 Trying<------------
CANCEL---------------->
487 Requst terminated<–
200 OK<----------------
487 Requst terminated<–
487 Requst terminated<–
487 Requst terminated<–
487 Requst terminated<–
487 Requst terminated<–
487 Requst terminated<–
OPTION sip2002@X.X.X.X<—
200 OK---------------->
487 Requst terminated<–
200 OK---------------->
REGISTER-------------->
and then it reregister.

and this is a regular CANCEL i did -

fllow:

PHONE SERVER
----------------->INVITE
407 PROXY<--------------
------------------->ACK
----------------->INVITE
100 Trying<------------
CANCEL---------------->
487 Requst terminated<–
200 OK<-----------------
ACK-------------------->
I am using Asterisk on Centos.

That version was end of life over three years ago and security fixes only over four years ago.

The phone is failing to send ACK to the 487.

[quote=“david55”]That version was end of life over three years ago and security fixes only over four years ago.

The phone is failing to send ACK to the 487.[/quote]

Hi,
thanks for the replay!
i know we are using very old version…but i cant do anything about that.
and i know it is not a phone problem - yes it is faling to send the ACK but i have 2 lines on
that phone on 2 diffrent servers 1 is getting unregister and 1 is ok.- so its not the phone.

somthing in the server itself.

is there somthing we can do? try to do ? without upgrading the asterisk version?

Thanks!

Normally, the next stage would be to ask for SIP traces, so that they can be checked for compliance with the protocol and obvious signs of NAT misconfiguration, but it takes time to analyse these, and any time spent which results in detecting an Asterisk bug is likely to be wasted.

Basically check:

is the 487 being sent to the correct place.

Is there anything in the way that might block SIP responses.

Is there anything in the way that might mangle SIP responses.

Is the contact address in the response valid for phone?

Etc.

Hello

i was able to solve this problem by adding to /etc/asterisk/sip.conf

this :

pedantic=yes

thanks everyone :smile: