Hi to everybody,
I have a problem for received calls form my Grandstream HT-503.
I have a FXO connect to my PABX, and I can make a call from PABX to
VOIP, but I didn't received calls to my VOIP, to my PABX.
See the log:
Using SIP RTP CoS mark 5
-- Executing [27100@ramais:1] MixMonitor("SIP/2000-0000bd8b",
"/media/HDExterno/gravacoes/feitas/APTO/27100/1489675579.120546.wav")
in new stack
-- Executing [27100@ramais:2] Dial("SIP/2000-0000bd8b",
"SIP/136/100,60,tT") in new stack
== Begin MixMonitor Recording SIP/2000-0000bd8b
== Using SIP RTP CoS mark 5
-- Called SIP/136/100
[2017-03-16 11:46:19] WARNING[1554][C-000098b9]: chan_sip.c:23843
handle_response_invite: Received response: "Forbidden" from
'<sip:[email protected]:5089>;tag=as57804b2e'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2000-0000bd8b' status is 'CHANUNAVAIL'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/2000-0000bd8b
And the SIP Debuug:
Called SIP/136/100
<--- SIP read from UDP:192.168.25.169:3329 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
From: <sip:[email protected]:5089>;tag=as62bede9e
To: <sip:[email protected]>
Call-ID: [email protected]:5089
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.25.169:3329 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport=5089
From: <sip:[email protected]:5089>;tag=as62bede9e
To: <sip:[email protected]>;tag=1820807938
Call-ID: [email protected]:5089
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.14.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.25.169:3329:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.25.24:5089;branch=z9hG4bK4433efa4;rport
Max-Forwards: 70
From: <sip:[email protected]:5089>;tag=as62bede9e
To: <sip:[email protected]>;tag=1820807938
Contact: <sip:[email protected]:5089>
Call-ID: [email protected]:5089
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.10.0
Content-Length: 0
---
[2017-03-16 11:34:53] WARNING[1554][C-000098af]: chan_sip.c:23843
handle_response_invite: Received response: "Forbidden" from
'<sip:[email protected]:5089>;tag=as62bede9e'
Scheduling destruction of SIP dialog
'[email protected]:5089' in 6400 ms
(Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/2000-0000bd7a' status is 'CHANUNAVAIL'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/2000-0000bd7a
See my sip.conf
;;
[136]
type=friend
defaultuser=136
secret=XXXXX
qualify=yes
;nat=no
nat=force_rport,comedia
context=ramais
;insecure=invite,port
disallow=all
allow=ulaw,alaw,gsm
host=dynamic
canreinvite=no
regext=136
callgroup=1
pickupgroup=1
I have a LOAD BALANCE too in this Grandstream.
The problem is the NAT/Firewall? Because the FXS is working well.
Thanks in advanced!
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