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-users