Well to be honest, our experience with asterisk never works with
under NAT. if you got DMZ then it will otherwise don’t hold your
breath for it.
If you want to use it as a production server
Your option is 1. Get a Real IP
2. there is no 2 really just get an ReaL Public IP
Sam
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Fidel Garcia
Sent: Saturday, June 21, 2008 6:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Voice only works from one way.
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.
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.
Good luck and please let me know if you get it up and running.
Fidel Garcia
System Engineer
sysTeam.
7205 NW 19th Street, Suite 302
Miami, Florida 33126
Email: [EMAIL PROTECTED]
Tel: (305)-477-7303 Fax: (305)-477-0013
http://www.systeamusa.com
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] On Behalf Of Sang-Kil (Sam) Suh
Sent: Friday, June 20, 2008 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voice only works from one way.
Yes, both Asterisk and Cisco are behind Nat.
On 6/20/08 3:26 PM, "Sam Tam" <[EMAIL PROTECTED]> wrote:
Are you using NAT?
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sang-
Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: [email protected]
Subject: [asterisk-users] Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on
Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-
xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension
1001 on
asterisk, which should talk to cisco. After initial connection to
Asterisk,
I have try to call F, and it will ring. Voice from softphone to F
carries
over and I can hear it; however, no voice from F to softphone will
carry. I
have been experimenting with different codec and other cisco/
asterisk config
tips from the web. None had worked so far.
If anyone have experienced such problem and knows how to solve this,
I will
be eternally grateful.
< sip.conf >
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
disallow = all
nat=yes
canreinvite=yes
allowguest=no
allow=ulaw
allow=alaw
allow=g711
allow=g729
allow=gsm
allow=ilbc
[2000]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2001]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2002]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[2003]
type=friend
context=my-phones
secret=
allow=ulaw
host=dynamic
[xxx.xxx.xxx.yyy]
context=pstn-incoming
type=friend
host=xxx.xxx.xxx.yyy ; IP address of Cisco gateway
dtmfmode=rfc2833
disallow=all
allow=ulaw
insecure=very
[1001]
context=local-phones
type=friend
username=1001
secret=secret
host=dynamic
mailbox=1001
insecure=very
< extensions.conf >
[my-phones]
exten => 2000,1,Dial(SIP/2000)
exten => 2001,1,Dial(SIP/2001)
exten => 2002,1,Dial(SIP/2002)
exten => 2003,1,Dial(SIP/2003)
exten => 6000,1,MeetMe(600,i,54321)
;include => lan-phones
[bogon-calls]
exten => _.,1,Congestion
[pstn-incoming]
include => lan-phones
[local-phones]
include => lan-phones
include => pstn-outbound
[pstn-outbound]
; Calls starting with 9 have the 9 stripped & are then routed out to
the
PSTN
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ; IP address of
Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/[EMAIL PROTECTED] ; IP address of
Cisco
gateway
;exten => _9XXXX,2,Congestion
exten => _9.,2,Congestion
[lan-phones]
exten => 1001,1,Dial(SIP/1001,20)
exten => 1001,2,Voicemail(u1001)
exten => 1001,3,Answer(SIP/1001)
exten => 1001,102,Voicemail(b1001)
exten => 1001,103,Hangup
< Cisco 2611 config >
Building configuration...
Current configuration : 2030 bytes
!
version 12.2
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname fxroute
!
logging queue-limit 100
enable secret
enable password
!
clock timezone GMT 0
ip subnet-zero
no ip routing
!
!
!
ip audit notify log
ip audit po max-events 100
!
!
!
!
!
voice rtp send-recv
!
voice service voip
sip
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 gsmefr
codec preference 4 gsmfr
!
!
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
mta receive maximum-recipients 0
!
!
!
!
interface Ethernet0/0
ip address xxx.xxx.xxx.yyy 255.255.255.0
no ip route-cache
no ip mroute-cache
full-duplex
no cdp enable
!
interface Ethernet0/1
no ip address
no ip route-cache
no ip mroute-cache
shutdown
half-duplex
no cdp enable
!
ip http server
no ip http secure-server
ip classless
!
!
!
!
call rsvp-sync
!
voice-port 1/0/0
input gain 10
output attenuation 10
no comfort-noise
connection plar opx 1001
station-id number 100
caller-id enable
!
voice-port 1/0/1
input gain 10
output attenuation 10
no comfort-noise
caller-id enable
!
voice-port 1/1/0
!
voice-port 1/1/1
!
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 100 pots
destination-pattern .T
progress_ind setup enable 3
progress_ind progress enable 8
port 1/0/0
!
dial-peer voice 2 voip
destination-pattern 1...
progress_ind setup enable 3
progress_ind progress enable 8
voice-class codec 1
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx:5060
session transport udp
dtmf-relay h245-alphanumeric
clid strip
no vad
!
dial-peer voice 1 pots
!
sip-ua
retry invite 3
retry response 3
retry bye 3
retry cancel 3
timers trying 1000
sip-server ipv4:xxx.xxx.xxx.xxx
!
!
!
telephony-service
transfer-pattern ....
transfer-system full-blind
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
password
login
!
!
end
Thank you
Sang-Kil (Sam) Suh
System administrator
--
Ticoon Technology Inc.
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Thank you
Sang-Kil (Sam) Suh
System administrator
--
Ticoon Technology Inc.
56 The Esplanade, Suite 404
Toronto, Ontario
M5E 1A7
Tel: (416) 513-9524 (ext. 299)
Cell: (416) 902-2890
Fax: (416) 513-9525
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