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

Sat Jun 21 07:17:48 CDT 2008

Have you tried keeping asterisk in on the call with a /n connection in  
the dial-plan?

Is there any firewall that is blocking udp ports to any of your clients?

Fred Posner
fred at teamforrest.com

On Jun 21, 2008, at 12:36 AM, Sam Tam wrote:

> 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com 
> ] 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: fgarcia at systeamusa.com
> Tel: (305)-477-7303 Fax: (305)-477-0013
> http://www.systeamusa.com
>
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com 
> ] 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" <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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>
> 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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