IAX runs on port 4569. If you have that forwarded to your asterisk box, then it should "just work".
If it doesn't, you might turn on IAX debugging on the asterisk console (I think it's console> iax2 debug), and start up tcpdump and see if the IAX packets are reaching your asterisk box: tcpdump -i eth0 port 4569 ---- Not to confuse the issue, but I thought I should throw this out there, as it's often overlooked: If your devices don't have an audio codec in common with the server, the call will be hung up as soon as you answer. What's often happened in this case is that someone has set up the phone to only allow G.729. Or someone is being silly, and has banned all the useful codecs in sip.conf, with something like: disallow=all allow=gsm In this case, if the phone doesn't support GSM, then the call will be hung up on as soon as it's answered. Oops! Cheers, spd On Thu, 31 May 2007, Richard (Rogers @ work) wrote: > 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] > >
