[asterisk-users] one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip[1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com nat=no[1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com nat=no I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service( sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
On 09/16/2010 07:59 PM, Thomas Johnson wrote: the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? I don't think so. It's just that you didn't specify which client is which. My next guess is that it is a codec problem. I've had similar problems - and upon checking Asterisk logs - I discovered that the client and Asterisk weren't agreeing correctly on codecs. Can you double-check your X-lite configuration - and maybe try to ulaw or alaw as the only codec at both ends? Sebastian On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
I have tried doing that with just ulaw and alaw, respectively, and nothing changed Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. On Thu, Sep 16, 2010 at 2:50 PM, Sebastian s...@open-t.co.uk wrote: On 09/16/2010 07:59 PM, Thomas Johnson wrote: the client that is behind nat is [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip do I have to enable nat on all of them? I don't think so. It's just that you didn't specify which client is which. My next guess is that it is a codec problem. I've had similar problems - and upon checking Asterisk logs - I discovered that the client and Asterisk weren't agreeing correctly on codecs. Can you double-check your X-lite configuration - and maybe try to ulaw or alaw as the only codec at both ends? Sebastian On Thu, Sep 16, 2010 at 1:36 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: On 09/16/2010 06:58 PM, Thomas Johnson wrote: I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip [1001];Work type=peer dtmfmode=rfc2833 context=sip insecure=very host=sip.domain.com http://sip.domain.com http://sip.domain.com nat=no [1000];IPKall type=peer dtmfmode=rfc2833 context=sip insecure=very host=voiper.ipkall.com http://voiper.ipkall.com http://voiper.ipkall.com nat=no You seem to be using nat=no shouldn't that be nat=yes? I pasted the log here - http://pastie.org/1163238 I have tried connecting both of the clients to another sip service(sip2sip.info http://sip2sip.info http://sip2sip.info) and did not have the same problems. Any suggestions would be great. Thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote: Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. http://www.aocomputing.net/?p=3 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
I already have that covered [tomfmason] type=friend secret=secret callerid=Thomas Johnson host=dynamic nat=yes canreinvite=no disallow=all allow=gsm allow=ulaw allow=alaw qualify=yes context=sip The server is not behind NAT only the client above is On Thu, Sep 16, 2010 at 4:59 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson tomfma...@gmail.com wrote: Also, if I disable the firewall in my router I lose incoming audio and outgoing audio works. http://www.aocomputing.net/?p=3 -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote: The server is not behind NAT only the client above is Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio for xlite clients behind NAT
If you are using linux firewall, try this, it was very usefull to me: iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FORWARD -p UDP --dport 5060 -j ACCEPT Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Thu, 16 Sep 2010 18:45:38 -0400 From: paul.belan...@polybeacon.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] one way audio for xlite clients behind NAT On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson tomfma...@gmail.com wrote: The server is not behind NAT only the client above is Sounds like a phone (not asterisk) issue then, make sure you have setup your NAT and port forwarding properly on the client side. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
