When I am trying to call someone, facing this error. i am using Twilio Sip trunk.how to fix this error?
Sending to 192.168.0.100:5060 (NAT)
Sending to 192.168.0.100:5060 (NAT)
Using INVITE request as basis request - 7IOAg3AH4ZPcXOAM2ee8dQ…
Found peer ‘7001’ for ‘7001’ from 192.168.0.100:5060
Got SDP version 1 and unique parts [- 1638992831 IN IP4 192.168.0.100]
Found RTP audio format 105
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format opus for ID 105
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g729|opus)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.100:10008
Looking for 4692512437 in from-phones (domain 192.168.0.101)
sip_route_dump: route/path hop: sip:7001@192.168.0.100:5060
<— Transmitting (NAT) to 192.168.0.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK-524287-1—d80fb07a67d8aa5d;received=192.168.0.100;rport=5060
From: <sip:7001@192.168.0. 101>;tag=5861e577
To: <sip:4692512437@192.168.0. 101>
Call-ID: 7IOAg3AH4ZPcXOAM2ee8dQ…
CSeq: 1 INVITE
Server: Asterisk PBX 19.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:4692512437@192.168.1. 101:5060>
Content-Length: 0
what should I do now?
