Pjsip Turnk Configuration issue

Asterisk Asterisk SIP
1

Dear
Experts, I am straggling with one issue of my asterisk pjsip configuration, I have done before same it was working but with this its not working, I don’t understands where I am wrong. let’s come to the point

I have sip trunk (IP Phone) credential, when I used zoiper, portasip, 3cx, microsip dialer to login & make a call its working perfect for both incoming & outgoing but when I configure same trunk with my asterisk I am facing issue with incoming calls & also outgoing take multiple attempt like 1-5 times to get response mostly getting congestion, I have try with both pjsip & sip both have same issue please help me out from this situation I share my configuration hope someone can help me who already face similar issue, thanks for your kind attention.

sip.conf


[general]
useragent=MicroSIP/3.21.3
realm = MicroSIP
context=default
allowoverlap=no
udpbindaddr=0.0.0.0
tcpbindaddr=0.0.0.0
bindaddr=0.0.0.0
bindport=5060
transport=udp
srvlookup=yes
disallow=all
allow=alaw
trustrpid=yes 
sendrpid=yes
limitonpeers=yes
keepalive=60

[09xxxxxxxxxx]
type=friend
username=09xxxxxxxxxx
fromuser=09xxxxxxxxxx
secret=09xxxxxxxxxx368
host=152.52.54.88
port=5060
call-limit=1
canreinvite=no
insecure=port,invite
qualify=yes

pjsip.conf

[global]
type=global
user_agent=MicroSIP/3.21.3
default_from_user=MicroSIP
default_realm=MicroSIP

[transport-udp]
type=transport
protocol=udp    udp,tcp,tls,ws,wss,flow
bind=0.0.0.0

[09xxxxxxxxxx]
type = registration
transport = transport-udp
outbound_auth = 09xxxxxxxxxx_auth
client_uri = sip:09xxxxxxxxxx@152.52.54.88
server_uri = sip:152.52.54.88
retry_interval = 60
expiration = 3600

[09xxxxxxxxxx_auth]
type = auth
password = 09xxxxxxxxxx368
username = 09xxxxxxxxxx

[09xxxxxxxxxx_aor]
type = aor
contact = sip:09xxxxxxxxxx@152.52.54.88

[09xxxxxxxxxx]
type = identify
endpoint = 09xxxxxxxxxx
match = 152.52.54.88

[09xxxxxxxxxx]
type = endpoint
context = from-client
disallow = all
allow = alaw
callerid = 09xxxxxxxxxx
from_user = 09xxxxxxxxxx
from_domain = 152.52.54.88
auth = 09xxxxxxxxxx_auth
outbound_auth = 09xxxxxxxxxx_auth
aors = 09xxxxxxxxxx_aor
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
timers = no
2

Share the error you are facing from the CLI.

3

Thanks for your reply, no exact error I am getting in my CLI for incoming calls for outgoing calls I am getting congestion 1-5 attempts then working then again 4-5 times congestion again working something like that, btw my config is correct?

4

Why are you claiming to be a phone, and why haven’t you told us who the ITSP is. What is really wrong here probably depends on the provider, and without either logging in sufficient detail, to see what is going wrong, or the identity of the provide, so other people can look at their set up instructions, it is going to be very difficult to help.

You need to use the command “pjsip set logger on”, and enable the full log.

This is not a translation of the sip.conf, and the name tends to imply trusted callers, but this seems to be a provider, who should never be trusted.

With a valid dialplan, the user part is always going to get overridden, so why include it?

Your sip.conf shows signs of cut and paste coding, as it has insecure=invite without an explanation of why it is needed (it usually isn’t), it uses type=friend when type=host is almost always better, uses username in a context where it has no effect, and it has the obsolete names for directmedia and default _user…

There seems to be a missing “;” here.

5

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