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<TITLE>Re: [asterisk-users] Voice only works from one way.</TITLE>
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<FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:9.0px'>Yes, both Asterisk and Cisco are behind Nat.<BR>
<BR>
<BR>
On 6/20/08 3:26 PM, "Sam Tam" <samtam888@gmail.com> wrote:<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:9.0px'><BR>
Are you using NAT?<BR>
<BR>
-----Original Message-----<BR>
From: asterisk-users-bounces@lists.digium.com<BR>
[<a href="mailto:asterisk-users-bounces@lists.digium.com]">mailto:asterisk-users-bounces@lists.digium.com]</a> On Behalf Of Sang-Kil (Sam)<BR>
Suh<BR>
Sent: Saturday, June 21, 2008 3:14 AM<BR>
To: asterisk-users@lists.digium.com<BR>
Subject: [asterisk-users] Voice only works from one way.<BR>
<BR>
Hello, everyone.<BR>
<BR>
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)<BR>
with Cisco 2611. Cisco has 2 port FXO card installed on it.<BR>
<BR>
For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx<BR>
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on<BR>
asterisk, which should talk to cisco. After initial connection to Asterisk,<BR>
I have try to call F, and it will ring. Voice from softphone to F carries<BR>
over and I can hear it; however, no voice from F to softphone will carry. I<BR>
have been experimenting with different codec and other cisco/asterisk config<BR>
tips from the web. None had worked so far.<BR>
<BR>
If anyone have experienced such problem and knows how to solve this, I will<BR>
be eternally grateful.<BR>
<BR>
< sip.conf ><BR>
[general]<BR>
port = 5060<BR>
bindaddr = 0.0.0.0<BR>
context = bogon-calls<BR>
disallow = all<BR>
nat=yes<BR>
canreinvite=yes<BR>
allowguest=no<BR>
allow=ulaw<BR>
allow=alaw<BR>
allow=g711<BR>
allow=g729<BR>
allow=gsm<BR>
allow=ilbc<BR>
<BR>
<BR>
[2000]<BR>
type=friend<BR>
context=my-phones<BR>
secret=<BR>
allow=ulaw<BR>
host=dynamic<BR>
<BR>
[2001]<BR>
type=friend<BR>
context=my-phones<BR>
secret=<BR>
allow=ulaw<BR>
host=dynamic<BR>
<BR>
[2002]<BR>
type=friend<BR>
context=my-phones<BR>
secret=<BR>
allow=ulaw<BR>
host=dynamic<BR>
<BR>
[2003]<BR>
type=friend<BR>
context=my-phones<BR>
secret=<BR>
allow=ulaw<BR>
host=dynamic<BR>
<BR>
[xxx.xxx.xxx.yyy]<BR>
context=pstn-incoming<BR>
type=friend<BR>
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway<BR>
dtmfmode=rfc2833<BR>
disallow=all<BR>
allow=ulaw<BR>
insecure=very<BR>
<BR>
[1001]<BR>
context=local-phones<BR>
type=friend<BR>
username=1001<BR>
secret=secret<BR>
host=dynamic<BR>
mailbox=1001<BR>
insecure=very<BR>
<BR>
< extensions.conf ><BR>
[my-phones]<BR>
exten => 2000,1,Dial(SIP/2000)<BR>
exten => 2001,1,Dial(SIP/2001)<BR>
exten => 2002,1,Dial(SIP/2002)<BR>
exten => 2003,1,Dial(SIP/2003)<BR>
exten => 6000,1,MeetMe(600,i,54321)<BR>
;include => lan-phones<BR>
<BR>
[bogon-calls]<BR>
exten => _.,1,Congestion<BR>
<BR>
[pstn-incoming]<BR>
include => lan-phones<BR>
<BR>
[local-phones]<BR>
include => lan-phones<BR>
include => pstn-outbound<BR>
<BR>
[pstn-outbound]<BR>
; Calls starting with 9 have the 9 stripped & are then routed out to the<BR>
PSTN<BR>
exten => _9.,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco<BR>
gateway<BR>
; 9 stripped by Cisco gateway<BR>
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; IP address of Cisco<BR>
gateway<BR>
;exten => _9XXXX,2,Congestion<BR>
exten => _9.,2,Congestion<BR>
<BR>
[lan-phones]<BR>
exten => 1001,1,Dial(SIP/1001,20)<BR>
exten => 1001,2,Voicemail(u1001)<BR>
exten => 1001,3,Answer(SIP/1001)<BR>
exten => 1001,102,Voicemail(b1001)<BR>
exten => 1001,103,Hangup<BR>
<BR>
< Cisco 2611 config ><BR>
<BR>
Building configuration...<BR>
<BR>
Current configuration : 2030 bytes<BR>
!<BR>
version 12.2<BR>
service config<BR>
service timestamps debug datetime msec<BR>
service timestamps log datetime msec<BR>
no service password-encryption<BR>
!<BR>
hostname fxroute<BR>
!<BR>
logging queue-limit 100<BR>
enable secret<BR>
enable password<BR>
!<BR>
clock timezone GMT 0<BR>
ip subnet-zero<BR>
no ip routing<BR>
!<BR>
!<BR>
!<BR>
ip audit notify log<BR>
ip audit po max-events 100<BR>
!<BR>
!<BR>
!<BR>
!<BR>
!<BR>
voice rtp send-recv<BR>
!<BR>
voice service voip<BR>
sip<BR>
!<BR>
voice class codec 1<BR>
codec preference 1 g711ulaw<BR>
codec preference 2 g711alaw<BR>
codec preference 3 gsmefr<BR>
codec preference 4 gsmfr<BR>
!<BR>
!<BR>
!<BR>
!<BR>
!<BR>
!<BR>
!<BR>
no voice hpi capture buffer<BR>
no voice hpi capture destination<BR>
!<BR>
!<BR>
mta receive maximum-recipients 0<BR>
!<BR>
!<BR>
!<BR>
!<BR>
interface Ethernet0/0<BR>
ip address xxx.xxx.xxx.yyy 255.255.255.0<BR>
no ip route-cache<BR>
no ip mroute-cache<BR>
full-duplex<BR>
no cdp enable<BR>
!<BR>
interface Ethernet0/1<BR>
no ip address<BR>
no ip route-cache<BR>
no ip mroute-cache<BR>
shutdown<BR>
half-duplex<BR>
no cdp enable<BR>
!<BR>
ip http server<BR>
no ip http secure-server<BR>
ip classless<BR>
!<BR>
!<BR>
!<BR>
!<BR>
call rsvp-sync<BR>
!<BR>
voice-port 1/0/0<BR>
input gain 10<BR>
output attenuation 10<BR>
no comfort-noise<BR>
connection plar opx 1001<BR>
station-id number 100<BR>
caller-id enable<BR>
!<BR>
voice-port 1/0/1<BR>
input gain 10<BR>
output attenuation 10<BR>
no comfort-noise<BR>
caller-id enable<BR>
!<BR>
voice-port 1/1/0<BR>
!<BR>
voice-port 1/1/1<BR>
!<BR>
!<BR>
mgcp profile default<BR>
!<BR>
dial-peer cor custom<BR>
!<BR>
!<BR>
!<BR>
dial-peer voice 100 pots<BR>
destination-pattern .T<BR>
progress_ind setup enable 3<BR>
progress_ind progress enable 8<BR>
port 1/0/0<BR>
!<BR>
dial-peer voice 2 voip<BR>
destination-pattern 1...<BR>
progress_ind setup enable 3<BR>
progress_ind progress enable 8<BR>
voice-class codec 1<BR>
session protocol sipv2<BR>
session target ipv4:xxx.xxx.xxx.xxx:5060<BR>
session transport udp<BR>
dtmf-relay h245-alphanumeric<BR>
clid strip<BR>
no vad<BR>
!<BR>
dial-peer voice 1 pots<BR>
!<BR>
sip-ua<BR>
retry invite 3<BR>
retry response 3<BR>
retry bye 3<BR>
retry cancel 3<BR>
timers trying 1000<BR>
sip-server ipv4:xxx.xxx.xxx.xxx<BR>
!<BR>
!<BR>
!<BR>
telephony-service<BR>
transfer-pattern ....<BR>
transfer-system full-blind<BR>
!<BR>
!<BR>
line con 0<BR>
exec-timeout 0 0<BR>
line aux 0<BR>
line vty 0 4<BR>
password<BR>
login<BR>
!<BR>
!<BR>
end<BR>
<BR>
Thank you<BR>
<BR>
Sang-Kil (Sam) Suh<BR>
System administrator<BR>
<BR>
--<BR>
Ticoon Technology Inc.<BR>
<BR>
<BR>
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</SPAN></FONT></BLOCKQUOTE><FONT FACE="Verdana, Helvetica, Arial"><SPAN STYLE='font-size:9.0px'><BR>
<BR>
Thank you<BR>
<BR>
Sang-Kil (Sam) Suh<BR>
System administrator<BR>
<BR>
-- <BR>
Ticoon Technology Inc.<BR>
56 The Esplanade, Suite 404<BR>
Toronto, Ontario<BR>
M5E 1A7<BR>
<BR>
Tel: (416) 513-9524 (ext. 299)<BR>
Cell: (416) 902-2890<BR>
Fax: (416) 513-9525<BR>
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