Situatation is that operator mangosip.ru got on 1 ip many realms. Problem is
that asterisk automatically changes host to ip in To: field. So operator
send error back. How to force asterisk not to change host to ip?
Settings:
register => A**1:[email protected] <1%[email protected]> - registry OK. But
throw callbackextension not working, trying to register by ip and error.
[mangosip]
secret = pass
defaultuser = A**
trunkname = mangosip
callerid =
hasexten = no
hassip = yes
hasiax = no
host = mangosip.ru
context = incoming
insecure = invite
fromuser = A**
fromdomain = mangosip.ru
type = peer
disallow = all
allow = alaw
nat = no
canreinvite = nonat
dtmfmode = info
Incoming OK
with this settings:
> sip show registry
Host dnsmgr Username Refresh State Reg.Time
mangosip.ru:5060 N A** 285 Registered Wed, 08 Sep 2010
1 SIP registrations.
> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
mangosip/A** 81.88.80.36 5060 Unmonitored
4 sip peers [Monitored: 1 online, 0 offline Unmonitored: 3 online, 0
offline]
And dialplan:
exten => _7495XXXXXXX,1,Dial(SIP/mangosip/${EXTEN})
exten => _7495XXXXXXX,2,HangUp
Getting this error:
Transmitting (no NAT) to 81.88.80.36:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 62.**:5060;branch=z9hG4bK117d12fa
Max-Forwards: 70
From: "user1" <sip:[email protected]>;tag=as7372af08
To: <sip:[email protected]>;tag=06239f873a4c6ea5e6ca1d6186a625d8.17e2
Contact: <sip:a...@62.*:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r285456
Content-Length: 0
...
<--- SIP read from UDP:81.88.80.36:5060 --->
*SIP/2.0 416 Unsupported URI Scheme*
Via: SIP/2.0/UDP 62.*:5060;rport=5060;branch=z9hG4bK1001766a
To: <sip:[email protected]>;tag=410a7c35
From: "user1"<sip:[email protected]>;tag=as7d4fb06c
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Softswitch2
Content-Length: 0
So 416 error because [email protected] host is resolved to ip. How to
change [email protected] to [email protected]?
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