[asterisk-users] Softphone behind NAT issues
We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching the traffic and noticed that there doesn't appear to be any rtp traffic going back to asterisk (this is where we think the problem might be). The firewalls on both sides have ports 5060, 1-2 and 3478 (STUN) open. Out conf files are: -- [sip.conf] *[general] context=incoming; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls allow = all [1000] nat=yes type=friend secret=Polycom context=internal host=dynamic canreinvite=no [EMAIL PROTECTED] callerid=TESTUSER1 1000 * - [extensions.conf] exten = 1000,1,Macro(stdexten,[EMAIL PROTECTED],SIP/1000) [rtp.conf] [general] rtpstart=12000 rtpend=12005 dtmftimeout=3000 What are we missing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone behind NAT issues
In [general] section: externip=your_extern_ip_address localnet=your_local_net/bits i.e. 192.168.0.0/24 Try this... On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching the traffic and noticed that there doesn't appear to be any rtp traffic going back to asterisk (this is where we think the problem might be). The firewalls on both sides have ports 5060, 1-2 and 3478 (STUN) open. Out conf files are: -- [sip.conf] [general] context=incoming; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls allow = all [1000] nat=yes type=friend secret=Polycom context=internal host=dynamic canreinvite=no [EMAIL PROTECTED] callerid=TESTUSER1 1000 - [extensions.conf] exten = 1000,1,Macro(stdexten,[EMAIL PROTECTED],SIP/1000) [rtp.conf] [general] rtpstart=12000 rtpend=12005 dtmftimeout=3000 What are we missing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone behind NAT issues
On 12 Jun 2007, at 17:53, Rob Schall wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. Alternatively you could use an IAX softphone. They generally don't have a problem with NAT or firewalls. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone behind NAT issues
No luck. Still no outbound sound. Leonardo Kamache (Gmail) wrote: In [general] section: externip=your_extern_ip_address localnet=your_local_net/bits i.e. 192.168.0.0/24 Try this... On 6/12/07, Rob Schall [EMAIL PROTECTED] wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching the traffic and noticed that there doesn't appear to be any rtp traffic going back to asterisk (this is where we think the problem might be). The firewalls on both sides have ports 5060, 1-2 and 3478 (STUN) open. Out conf files are: -- [sip.conf] [general] context=incoming; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls allow = all [1000] nat=yes type=friend secret=Polycom context=internal host=dynamic canreinvite=no [EMAIL PROTECTED] callerid=TESTUSER1 1000 - [extensions.conf] exten = 1000,1,Macro(stdexten,[EMAIL PROTECTED],SIP/1000) [rtp.conf] [general] rtpstart=12000 rtpend=12005 dtmftimeout=3000 What are we missing? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone behind NAT issues
This is probably what we'll have to do. We wanted to try to use all SIP though. As I read through the documentation, it seems possible though. Not sure where i'm off. Rob Tim Panton wrote: On 12 Jun 2007, at 17:53, Rob Schall wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. Alternatively you could use an IAX softphone. They generally don't have a problem with NAT or firewalls. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone behind NAT issues
Here's an update. We now see RTP traffic being generated by the laptop using the softphone. However its destination address is the asterisk servers internal IP address and not the outside address it needs to be pointing to. We set the externip setting, but that doesn't seem to change xlite's behavor of where to send its packets back to. Any thoughts? Rob Rob Schall wrote: This is probably what we'll have to do. We wanted to try to use all SIP though. As I read through the documentation, it seems possible though. Not sure where i'm off. Rob Tim Panton wrote: On 12 Jun 2007, at 17:53, Rob Schall wrote: We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. Alternatively you could use an IAX softphone. They generally don't have a problem with NAT or firewalls. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
