I have issue on my configuration I made 3 sip extension when We call each other ,we can not hear every one of us except when we made the caller on hold after that on call.
where is the missing on my config ?
You haven’t provided your configuration, or even told us which SIP channel driver, or Asterisk version.
However, even if you did, I would probably want protocol debugging including the SDP exchange.
Note that Asterisk doesn’t use “extension” to refer to local telephone instruments.
which configuration you want to share it with you ?
any suggestions for this issue ?
It’s been less than an hour since @david551 responded to you. You shouldn’t expect realtime responses with immediate help in a community forum like this. As for what he requested, the configuration of the SIP channel driver you are using (chan_sip or chan_pjsip), the SIP logging (sip set debug on or pjsip set logger on), and the Asterisk version.
Retransmitting #8 (no NAT) to 10.16.36.46:61055:
NOTIFY sip:email@example.com:61055;rinstance=99dff5eaae5f6502 SIP/2.0
Via: SIP/2.0/UDP 172.17.9.26:5060;branch=z9hG4bK642e3bd9;rport
From: “asterisk” sip:firstname.lastname@example.org;tag=as6158e1e3
CSeq: 141 NOTIFY
User-Agent: Asterisk PBX 13.18.0-rc1
Voice-Message: 0/0 (0/0)
<— SIP read from UDP:10.16.36.46:61055 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.9.26:5060;branch=z9hG4bK642e3bd9;rport=5060
CSeq: 140 NOTIFY
User-Agent: 3CXPhone 6.0.26523.0
The SIP logging for a failing call. That for a successful NOTIFY is of little use.
I have addding the below command It’s work fine now thank you for your helping:
iptables -A INPUT -s 10.0.0.0/24 -p tcp --dport 22 -j ACCEPT
Doesn’t make any sense at all, but ok…
Agreed. The rule quoted seems to be about ssh, and nothing to do with Asterisk.