Thanks Dave. My problem is pretty wier. I know my IAX trunk works but my IAX softphone does not work from the internet. It does not even register with my *server. It only works internally.
Richard ----- Original Message ----- From: "David Cook" <[EMAIL PROTECTED]> To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]> Cc: <[email protected]> Sent: Thursday, May 31, 2007 1:38 PM Subject: Re: [on-asterisk] Allow inbound SIP traffic thru d-link router > IAX will require port fwd 4569/UDP from the outside to your Asterisk > box. Control & data use the same port so no nat issues. > -- > David Cook > > > Quoting "Richard (Rogers @ work)" <[EMAIL PROTECTED]>: > > > Thanks Simon. Yes, my network was what you suggested. > > I will make the additional config on my *server tonight and try it. > > > > On the same token, can you suggest any additional config required for > > IAX2? > > I am also having problem with IAX2 traffic too. > > > > Thanks a lot! > > Richard > > ----- Original Message ----- > > From: "Simon P. Ditner" <[EMAIL PROTECTED]> > > To: "Richard (Rogers @ work)" <[EMAIL PROTECTED]> > > Cc: <[email protected]> > > Sent: Thursday, May 31, 2007 11:54 AM > > Subject: Re: [on-asterisk] Allow inbound SIP traffic thru d-link > > router > > > > > > > Is this what your network looks like? > > > > > > [Asterisk] -- [Dlink Router] -- (Internet) -- [Phone] ? > > > > > > On the Asterisk server, you will need the following in your > > sip.conf: > > > > > > externip = <public IP address of your connection> > > > > > > or: > > > > > > externhost=<DNS name, i.e. home.dyndns.org> > > > externrefresh=10 > > > > > > You might also need to put the following in your phone's profile: > > > > > > [phone1] > > > nat=yes > > > > > > In rtp.conf, you should see a range of UDP ports that should likely > > be > > > forwarded to your asterisk box in your D-Link router as well. i.e.: > > > > > > rtpstart=10000 > > > rtpend=20000 > > > > > > Cheers, > > > spd > > > > > > On Thu, 31 May 2007, Richard (Rogers @ work) wrote: > > > > > > > Hi, > > > > > > > > I know there was a session on SIP last night but I was unable to > > attend. > > I > > > > am hoping some of you could help me out based on what you > > learned. > > > > I am having problem getting my router to work with my Sip phone > > which is > > > > connecting from the internet to my asterisk server. > > > > > > > > I am not able to get the voice channel thru my router. And my > > SIP phone > > > > sometimes register with the server successfully but sometime not. > > > > I opened up both TCP and UDP port 5060 to point to my * server. > > > > > > > > Is there anything else I have to add? I am not sure how I would > > add a > > range > > > > of ports to my * server. > > > > > > > > Thanks, > > > > Richard > > > > > > > > > > > > > > --------------------------------------------------------------------- > > > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > > > > > > > > --------------------------------------------------------------------- > > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > > > > > > --------------------------------------------------------------------- > > To unsubscribe, e-mail: [EMAIL PROTECTED] > > For additional commands, e-mail: [EMAIL PROTECTED] > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [EMAIL PROTECTED] > For additional commands, e-mail: [EMAIL PROTECTED] >
