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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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: [EMAIL PROTECTED] Tel: (305)-477-7303 Fax: (305)-477-0013 http://www.systeamusa.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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" <[EMAIL PROTECTED]> wrote: Are you using NAT? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[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 No virus found in this incoming message. 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_______________________________________________ -- 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