[asterisk-users] Voice only works from one way.

[Sam Tam]

Are you using NAT?

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Sang-Kil (Sam)
Suh
Sent: Saturday, June 21, 2008 3:14 AM
To: asterisk-users at lists.digium.com
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/${EXTEN:1}@xxx.xxx.xxx.yyy) ; IP address of Cisco
gateway
; 9 stripped by Cisco gateway
;exten => _9XXXX,1,Dial,SIP/${EXTEN}@xxx.xxx.xxx.yyy ; 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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