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<title>Re: [asterisk-users] Voice only works from one way.</title>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I was never able to get it to work that way. When I had Asterisk
in NAT I was able to make calls, but most of the times they were one way voice.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I was able to get two-way voice when I configured the remote phone
using STUN and Symetrical RTP. However, the calls dropped every 19-20 seconds. I
read several threads online, but nobody explained the requirements in details.
Everything works fine if you have a public IP address or DMZ on Asterisk.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Good luck and please let me know if you get it up and running.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Fidel Garcia<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>System Engineer<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>sysTeam.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:black'>7205 NW 19th Street, Suite 302<br>
Miami, Florida 33126<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Email: fgarcia@systeamusa.com <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Tel: (305)-477-7303 Fax: (305)-477-0013 <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>http://www.systeamusa.com<o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Sang-Kil
(Sam) Suh<br>
<b>Sent:</b> Friday, June 20, 2008 3:48 PM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] Voice only works from one way.<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:7.0pt;font-family:"Verdana","sans-serif"'>Yes,
both Asterisk and Cisco are behind Nat.<br>
<br>
<br>
On 6/20/08 3:26 PM, "Sam Tam" <samtam888@gmail.com> wrote:</span><o:p></o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:7.0pt;
font-family:"Verdana","sans-serif"'><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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<p class=MsoNormal><span style='font-size:7.0pt;font-family:"Verdana","sans-serif"'><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</span><o:p></o:p></p>
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