Sip status showing UNREACHABLE

Asterisk Asterisk SIP
1

i am getting this massage while try to register the sip. But i am able to make outgoing calls.
i am using Asterisk 13.26.0 and server is in cloud.
[Jul 16 18:13:21] NOTICE[18554]: chan_sip.c:30196 sip_poke_noanswer: Peer ‘5744’ is now UNREACHABLE! Last qualify: 0

[5744]
type=friend
callerid=5744
username=5744
secret=XXXX
host=dynamic
;name=other
canreinvite=yes
nat=force_rport,comedia
;nat=yes
disallow=all
allow=alaw
allow=ulaw
qualify=yes
context=zingg
;call-limit=1

Please suggest me. . .

2

qualify=no, if the peer’s OPTIONS handling is broken, or fix the network problem that is casing OPTIONS to be lost.

Although not currently implicated, please review the following options as they have values are aeither obsolete or unusual for a local device: type, canreinvite and nat.

3

i removed canreinvite and nat
also did qualify=no but still same problem.
my server and sip-phone is in different network.

4

Have you reloaded the configuration, as qualify=no should disable the code that is producing the diagnostic. (Note if the message is genuine, actually trying to make a calll is likely to fail with “no reply to critical request”.)

5

Yes, I reload and also restart asterisk.
Now it is showing unmonitored.
I also get this when i restarted asterisk : saved useragent

6

Is anyone can suggest me, What to do in this case?
The same server with same setting is working at other location with soft phone. but here i am using panasonic-kx phone.

7

An unmonitored state is the state that setting qualify=no was intended to achieve. If the system is sitll broken, you need to find out why the peer isn’t responding to normal requests. That will typically be because the request isn’t reaching it, or the reply is going to the wrong place or getting lost in transit.

You should statrt by getting a protocol trace https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information#CollectingDebugInformation-Enablechanneltechorfeaturespecificdebug Look at ti to see that requests are going to the right place and that the Contact header contains an address that the peer can use.

8

<------------->

Retransmitting #1 (NAT) to 50.241.23.122:5060:
OPTIONS sip:5747@45.32.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP 45.32.XX.XX:5060;branch=z9hG4bK1ad20237;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@45.32.XX.XX;tag=as7d92e38d
To: sip:5747@45.32.XX.XX:5060
Contact: sip:asterisk@45.32.XX.XX:5060
Call-ID: 1e2c37c2139c7078228cd1b408d1c6e2@45.32.XX.XX:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.26.0
Date: Thu, 18 Jul 2019 04:42:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #1 (NAT) to 50.241.23.122:5070:
OPTIONS sip:5746@45.32.XX.XX:5070 SIP/2.0
Via: SIP/2.0/UDP 45.32.XX.XX:5060;branch=z9hG4bK3772b6b2;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@45.32.XX.XX;tag=as5a873d5a
To: sip:5746@45.32.XX.XX:5070
Contact: sip:asterisk@45.32.XX.XX:5060
Call-ID: 0acb26284b5debc444b642973a520ec7@45.32.203.100:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.26.0
Date: Thu, 18 Jul 2019 04:42:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

and my SIP is like:
[5747]
type=friend
callerid=5747
username=5747
secret=XXXXXXXX
host=dynamic
canreinvite=yes
nat=force_rport,comedia
disallow=all
allow=alaw
allow=ulaw
qualify=5000
context=Chinnggg

Some times its work automatically and sometimes not.

9

If this is intermittent, you have network overload problems.

Note that 5000 is a very large value for qualify. If you need such a large value, your network is so overloaded that no VoIP is likely to work satisfactorily, even without packet loss.

10

i changed this for few minutes. after that passed
qualify=yes

the same setting is working for second location.

11

Still this is not resolved.
Please guide me further . . . .

12

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