Hi all,
I have face a problem when I try to configure SIP trunk for my office. I was use Huawei Softx3000 as the point (SIP Gateway). Its seem I can’t reach the server. Even I was open 5060 port to the firewall for both receive/transmit.
Below is the results:
asterisk -rx "sip show peers"
Name/username Host Dyn Nat ACL Port Status
SS-GXXX 10.X.XXX.X N 5060 UNREACHABLE
8888 (Unspecified) D N A 0 UNKNOWN
8000 (Unspecified) D N A 0 UNKNOWN
I have put the SIP Debug also for the Softx3000 IP.
Here is the result:
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
anura-01*CLI>
<— SIP read from 10.X.XXX.X:5060 —>
SIP/2.0 504 Server Time-out
Via: SIP/2.0/UDP 10.0.X.X:5060;branch=z9hG4bK3c747b42;received=10.X.XXX.X;rport=55976
Call-ID: [email protected]
From: "Unknown"sip:[email protected];tag=as366b8d66
To: sip:10.X.XXX.X;tag=e6edfc95
CSeq: 102 OPTIONS
Warning: 399 10.X.XXX.X "SSF00120L00667 No Response From Transaction User"
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Reliably Transmitting (NAT) to 10.X.XXX.X:5060:
OPTIONS sip:10.X.XXX.X SIP/2.0
Via: SIP/2.0/UDP 10.0.XX.X:5060;branch=z9hG4bK69f2c1f5;rport
From: “Unknown” sip:[email protected];tag=as1fa1ea17
To: sip:10.X.XXX.X
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 06 Oct 2010 07:22:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
I want to configure one SIP Trunk with multiple DID.
Below is my current configuration:
[B]sip_additional.conf[/B]
[GXXX]
disallow=all
type=user
nat=yes
insecure=invite,port
host=10.X.XXX.X
fromdomain=10.X.XXX.X
dtmfmode=rfc2833
context=from-trunk
qualify=yes
allow=g726
allow=g729
allow=ulaw
allow=alaw
allow=ilbc
[SS-GXXX]
disallow=all
type=peer
host=10.X.XXX.X
nat=yes
insecure=invite,port
allow=g726
allow=g729
allow=ilbc
allow=ulaw
allow=alaw
fromdomain=10.X.XXX.X
dtmfmode=rfc2833
qualify=yes
[B]sip_general_custom.conf[/B]
disallow=all
allow=ulaw
allow=alaw
jbenable=no
rtptimeout=30
maxexpiry=3600
allowguest=yes
defaultexpiry=600
minexpiry=60
srvlookup=no
registerattempts=0
registertimeout=20
notifyhold=yes
g726nonstandard=no
t38pt_udptl=yes
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
checkmwi=10
notifyringing=yes
nat=never
externip=10.X.XXX.X
[B]sip_custom.conf[/B]
pedantic=yes
silencesuppression=yes
Anybody have face this problem? How can I solve this?