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: asterisk-users@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:[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. > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-user > 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
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users