Wrong FROM in OPTIONS

Asterisk Asterisk Support
1

so i am trying to send OPTION to another server which have only option to be registered with DOMAIN name instead of IP but my asterisk is sending regularly IP only.. here are OPTIONS logs my ip is 52.114.132.46

Reliably Transmitting (no NAT) to 52.114.132.46:5061:
OPTIONS sip:sip.microsoft SIP/2.0
Via: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK39781a8f
Max-Forwards: 70
From: “asterisk” sip:asterisk1@52.114.132.46;tag=as06b0a877
To: sip:sip.microsoft
Call-ID: 5d9099d343d03c2714985ead3a712ced@52.114.132.46:5061
CSeq: 102 OPTIONS
User-Agent: MagnusBilling
Date: Thu, 24 Apr 2025 10:29:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from TLS:52.114.132.46:5061 —>
SIP/2.0 403 Forbidden
FROM: "asterisk"sip:asterisk1@52.114.132.46;tag=as06b0a877
TO: sip:sip.microsoft
CSEQ: 102 OPTIONS
CALL-ID: 5d9099d343d03c2714985ead3a712ced@52.114.132.46:5061
VIA: SIP/2.0/TLS 52.114.132.46:5061;branch=z9hG4bK39781a8f
REASON: Q.850;cause=63;text=“3d0c007f-e41a-4031-8f04-dcf3b7d4f8a2;SBC certificate is not issued correctly. Provided trunk FQDN ‘52.114.132.46’ is not included in certificate’s CN or SAN list. Certificate allows following FQDNs only: sip.voippro.online.”
CONTENT-LENGTH: 0
ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
SERVER: Microsoft.PSTNHub.SIPProxy v.2025.3.31.2 i.USEA.11

HERE IS MY SIP.CONF

[general]
domain=sip.voippro.online
sipdebug=yes
bindaddr=0.0.0.0 ; Changed to listen on all interfaces, use domain for identification
bindport=5060
context = billing
dtmfmode=RFC2833
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=alaw
allow=gsm
rtcachefriends=yes
srvlookup=yes
useragent=MagnusBilling
allowsubscribe = no
alwaysauthreject=yes
rtptimeout=60
rtpholdtimeout=300
rtsavesysname=yes
ignoreregexpire=yes
externhost=sip.voippro.online
localnet=10.0.0.0/24
nat=force_rport,comedia
externip=52.114.132.46
language=en
rtupdate=yes
allowguest=no
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/letsencrypt/live/sip.voippro.online/fullchain.pem
tlsprivatekey=/etc/letsencrypt/live/sip.voippro.online/privkey.pem
tlscafile=/etc/ssl/certs/ca-certificates.crt
tlsclientmethod=tlsv1_2
tlscipher=ALL
tlsverify=no

[azure]
type=peer
host=sip.microsoft
port=5061
transport=tls
tlsenable=yes
tlsbindaddr=0.0.0.0:5061
tlscertfile=/etc/letsencrypt/live/sip.voippro.online/fullchain.pem
tlsprivatekey=/etc/letsencrypt/live/sip.voippro.online/privkey.pem
tlscafile=/etc/ssl/certs/ca-certificates.crt
clientmethod=tlsv1_2
tlscipher=TLS_AES_256_GCM_SHA384:TLS_AES_128_GCM_SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384>
tlsverify=no
tcpenable=yes
disallow=all
allow=g729,gsm,opus,alaw,ulaw
directmedia=no
context=from-trunk
dtmfmode=RFC2833
insecure=port,invite
nat=no
qualify=yes
domain=sip.voippro.online
fromdomain=sip.voippro.online
fromuser=asterisk1
sendrpid=no
localnet=10.0.0.0/24

2

That’s an OPTIONS, not an INVITE.

3

sorry i am newbie to it. even if its options how to change ip to domain?

4

OPTIONS is used by Asterisk to determine if the remote side is available/reachable or not.

You’re also using chan_sip, and my knowledge of that is mostly gone so I can’t comment on that aspect. Someone else may be able to.

5

ok i modified my post changing it to OPTIONS. anyways thanks.

6

domain doesn’t do what you think it does. You probably want fromdomain.

chan_sip is not included in the latest official source code versions, and very few people here are familiar with it.