This seems to be an easy-to-solve problem but it may be again my lask of knowledge in linux: My linux fedora core 3 asterisk box has a public IP and a private IP (two NIC) I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for both interfaces. I was able con connect my sip soft phone from a NAT connection inside my network pointing to the public IP. When attempting to do the same from outside my network (from my dsl connection from home), I get to hear the asterisk auto attendant but not able to send any sound from my laptop. This is my sip.conf file: [general] context=ramosoft allowguest=no realm=ramosoft.com bindaddr=0.0.0.0 bindport=5060 srvlookup=yes pedantic=yes tos=184 tos=lowdelay maxexpirey=3600 defaultexpirey=120 disallow=all allow=ulaw allow=ilbc allow=gsm musicclass=default language=es relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=RamoSoftPBX regcontext=ramosoft localnet=10.10.10.0/255.255.255.0 rtcachefriends=yes [authentication] [311] type=friend regexten=311 username=311 secret=311 callerid="Elpidio Ramos" <311> host=dynamic nat=yes canreinvite=no Is there anything I am missing here to get two way voice? Thank you in advance all -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060904/8f18652c/attachment.htm
On 9/4/06, Elpidio Ramos <elpidio@ramosoft.com> wrote:> When attempting to do the same from outside my network (from my dsl > connection from home), I get to hear the asterisk auto attendant but not > able to send any sound from my laptop.ARE YOU SURE IT ISN'T A DTMF PROBLEM!!!!!!!!!!!!!!!!!!!!!!!!!!????????????????????
On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote:> This seems to be an easy-to-solve problem but it may be again my lask > of knowledge in linux: > > My linux fedora core 3 asterisk box has a public IP and a private IP > (two NIC) > > I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for > both interfaces. > > I was able con connect my sip soft phone from a NAT connection inside > my network pointing to the public IP. >You do not have either the externip or externhost directives in your sip.conf. If you are connecting from the outside you need to tell Asterisk the IP address or hostname of the outside connection.>-- Carlos Chavez Prats Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060904/ee54d626/attachment.pgp
Elpidio Ramos wrote:> This seems to be an easy-to-solve problem but it may be again my lask of > knowledge in linux: > > My linux fedora core 3 asterisk box has a public IP and a private IP > (two NIC) > > I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for > both interfaces. > > I was able con connect my sip soft phone from a NAT connection inside my > network pointing to the public IP. > > When attempting to do the same from outside my network (from my dsl > connection from home), I get to hear the asterisk auto attendant but not > able to send any sound from my laptop. > > This is my sip.conf file: > > [general] > context=ramosoft > allowguest=no > realm=ramosoft.com > bindaddr=0.0.0.0 > bindport=5060 > srvlookup=yes > pedantic=yes > tos=184 > tos=lowdelay > maxexpirey=3600 > defaultexpirey=120 > disallow=all > allow=ulaw > allow=ilbc > allow=gsm > musicclass=default > language=es > relaxdtmf=yes > rtptimeout=60 > rtpholdtimeout=300 > useragent=RamoSoftPBX > regcontext=ramosoft > localnet=10.10.10.0/255.255.255.0 > rtcachefriends=yes > > [authentication] > > [311] > type=friend > regexten=311 > username=311 > secret=311 > callerid="Elpidio Ramos" <311> > host=dynamic > nat=yes > canreinvite=no > Is there anything I am missing here to get two way voice? > > Thank you in advance allIf you have two working nic's, then when the soft phone is on the inside of the network, it should register with the IP address of the inside nic. When the soft phone is on the outside (eg Internet), then it should be registering with the IP address of the outside nic. Any other combination is going to give you problems and particularly if you are using a firewall. The problems will be associated with basic layer-3 stuff and nating.
But the soft phones have dynamic ip addresses. I have read this is why we use host=dynamic and nat=yes. Carlos Chavez <cursor@telecomabmex.com> wrote: On Mon, 2006-09-04 at 09:49 -0700, Elpidio Ramos wrote:> This seems to be an easy-to-solve problem but it may be again my lask > of knowledge in linux: > > My linux fedora core 3 asterisk box has a public IP and a private IP > (two NIC) > > I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for > both interfaces. > > I was able con connect my sip soft phone from a NAT connection inside > my network pointing to the public IP. >You do not have either the externip or externhost directives in your sip.conf. If you are connecting from the outside you need to tell Asterisk the IP address or hostname of the outside connection.>-- Carlos Chavez Prats Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 Elpidio Ramos President RM International Services SA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax +52 (55) 1755-6601 Cell USA: +1 (801) 494-1415 Office +1 (240) 250-8264 Fax +1 (801) 938-4740 Direct -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060904/41da0671/attachment.htm
Try setting in sip.conf: nat=route This tells asterisk to send all responses back to where the inquiry came from rather then from the info contained in the sip packet. Good luck, Mark Coccimiglio IS Director Payroll Services Hawaii, Inc. http://www.psh-inc.com Elpidio Ramos wrote:> This seems to be an easy-to-solve problem but it may be again my lask > of knowledge in linux: > > My linux fedora core 3 asterisk box has a public IP and a private IP > (two NIC) > > I got the ports open in fedora core 3 (5060 and 10000 thru 30000) for > both interfaces. > > I was able con connect my sip soft phone from a NAT connection inside > my network pointing to the public IP. > > When attempting to do the same from outside my network (from my dsl > connection from home), I get to hear the asterisk auto attendant but > not able to send any sound from my laptop. > > This is my sip.conf file: > > [general] > context=ramosoft > allowguest=no > realm=ramosoft.com > bindaddr=0.0.0.0 > bindport=5060 > srvlookup=yes > pedantic=yes > tos=184 > tos=lowdelay > maxexpirey=3600 > defaultexpirey=120 > disallow=all > allow=ulaw > allow=ilbc > allow=gsm > musicclass=default > language=es > relaxdtmf=yes > rtptimeout=60 > rtpholdtimeout=300 > useragent=RamoSoftPBX > regcontext=ramosoft > localnet=10.10.10.0/255.255.255.0 > rtcachefriends=yes > > [authentication] > > [311] > type=friend > regexten=311 > username=311 > secret=311 > callerid="Elpidio Ramos" <311> > host=dynamic > nat=yes > canreinvite=no > Is there anything I am missing here to get two way voice? > > Thank you in advance all > >------------------------------------------------------------------------ > >_______________________________________________ >--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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061228/3995ffef/attachment.htm