so now the call is not being hanged up, but cannot hear anything. Tryied it with ilbc,alaw, ulaw and gsm
I still think it sould be a matter of RTP addressing since I get the following after a sip debug :
-- Executing BackGround("SIP/alyed-5a8d", "/var/lib/asterisk/sounds/test") in new stack
We're at 200.78.243.12 port 13458
Answering with preferred capability 0x400(ILBC)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 90.0.0.10;branch=z9hG4bK5a00000a000000c043bab4f9390f1bef000002ef;received=201.127.53.246;rport=5060
From: "unknown"<sip:[EMAIL PROTECTED]:5060>;tag=2438130825771721203
To: <sip:[EMAIL PROTECTED]:5060>;tag=as7222f729
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 220
v=0
o=root 17028 17028 IN IP4 200.78.243.12
s=session
c=IN IP4 200.78.243.12
t=0 0
m=audio 13458 RTP/AVP 97 101
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 201.127.53.246:5060
-- Playing '/var/lib/asterisk/sounds/test' (language 'en')
Integra2*CLI>
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 90.0.0.10;rport;branch=z9hG4bK5a00000a000000c043bab4f944b4f6f3000002f2
Content-Length: 0
Call-ID: [EMAIL PROTECTED]
CSeq: 2 ACK
From: "unknown"<sip:[EMAIL PROTECTED]:5060>;tag=2438130825771721203
Max-Forwards: 70
To: <sip:[EMAIL PROTECTED]:5060>;tag=as7222f729
User-Agent: SJphone/1.60.299a/L (SJ Labs)
9 headers, 0 lines
think it is addressing the rtp to my internal IP, but don't know who can I address it in the right way. I'm using the default STUN config in the SJphone :
STUN server address --> stun.softjoys.com :3478, refresh time out -->1200000 conclusive response timeout-->0 retrunsmissions number --> 13
and nat= yes in the sip.conf
But still no sound in my endpoint
Alyed
Alyed Tzompa Sosa
Simitel
VoIP developer
+52 (55) 24 52 43 90 Ext. (107)
[EMAIL PROTECTED]
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From: "Tony Jago" <[EMAIL PROTECTED]>
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Subject: Re: [Asterisk-Users] SIP through freeBSD NAT
Date: Tue, 3 Jan 2006 06:29:27 +1000
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"Unable to agree on media streams" means that the two devices can't pick a common codec. This shouldn't have anything to do with the firewall.
----- Original Message -----From: Alyed TzompaSent: Tuesday, January 03, 2006 5:32 AMSubject: [Asterisk-Users] SIP through freeBSD NATHi everyone
My problem is the following:
I'm trying to make a call from a sip phone (SJphone) behind a Restricted Cone NAT towards and Asterisk behind another NAT
(a freeBSD 3.3 using pf). By now I'm only trying to play a record set in the remote Asterisk.
My soft phone registers without problems to the Asterisk but once the record starts to play I get a hangup. SJphone outputs
"End reason: Unable to agree on media streams".
I'm forwarding SIP and IAX ports from the remote NAT towards the Asterisk box (i've tryied it with IAX with no problems) using
the following config in the remote NAT:
/etc/pf.conf
.....
# outgoing UDP port 5060 connections use source port 5060 on firewall
nat on $ext_if inet proto udp from any port = 5060 to any -> ($ext_if) port 5060
# Redirect all trafic from NAT:asterisk_port to 192.168.1.5:asterisk_port
rdr on $ext_if inet proto { tcp, udp } from any to any port 4569 -> 192.168.1.5 port 4569
rdr on $ext_if inet proto { tcp, udp } from any to $ext_if port 5060 -> 192.168.1.5 port 5060
rdr on $ext_if inet proto { tcp , udp} from any to any port 10000:20000 -> 192.168.1.5 port 10000:20000
# Let the Internet see our services
pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 4569 keep state
pass in log-all quick on $ext_if inet proto { tcp, udp } from any to any port 5060 keep state
.....
------------------------------------------------------------------
I think the problem might relay in this "pass in log-all" since once I commented the last line and the SJphone was unable to
register, but I haven't found a way to set up a range using this "pass" command (it complains saying that the " : " is valid only
with the "rdr " command) but I haven't found info explaining why I should (or shouldn't) use this "pass" command.
My Asterisk config is:
sip.conf
[general]
port=5060
externip = www.theip.net
localnet = 192.168.1.0
localmask = 255.255.255.0
allow=all
[user]
....
nat=yes
canreinvite=no
host=dynamic
--------------------------------------------
extensions.conf
exten => 400,1,Background(/var/lib/asterisk/sounds/myrecord)
exten => 400,2,Hangup
exten => 400,102,Hangup
---------------------------------------------
Thanx a lot!
ww6
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