asterisk*CLI> sip set debug peer 710
на 710 набираю *8
в выводе INVITE *8 я не вижу
Код: Выделить всё
SIP Debugging Enabled for IP: 10.7.0.10
Reliably Transmitting (NAT) to 10.7.0.10:5060:
OPTIONS sip:710@10.7.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 10.7.0.1:5060;branch=z9hG4bK2972a339;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.7.0.1>;tag=as562cf897
To: <sip:710@10.7.0.10:5060>
Contact: <sip:asterisk@10.7.0.1:5060>
Call-ID: 5cd343923fc73365186ab9e63c1167d8@10.7.0.1:5060
CSeq: 102 OPTIONS
User-Agent: CallWay PBX
Date: Tue, 02 Apr 2013 13:39:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.7.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.7.0.1:5060;branch=z9hG4bK2972a339;rport=5060
From: "asterisk" <sip:asterisk@10.7.0.1>;tag=as562cf897
To: <sip:710@10.7.0.10:5060>;tag=1646884712
Call-ID: 5cd343923fc73365186ab9e63c1167d8@10.7.0.1:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1405 1.0.3.30
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5cd343923fc73365186ab9e63c1167d8@10.7.0.1:5060' Method: OPTIONS