[asterisk-users] xxxx SPAM Low xxxx Re: Voice only works from one way.

Fri Jun 20 14:36:51 CDT 2008

Yes. Both Asterisk and Cisco is behind it.

On 6/20/08 3:26 PM, "Sam Tam" <samtam888 at gmail.com> wrote:

> 
> 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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> 
> _______________________________________________
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> 
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> Register Now: http://www.astricon.net
> 
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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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