Miles Scruggs wrote: > Hmm all your questions are covered in this email, but I'll summarize it > again in this reply: > > Server: 1.2.7.1 direct connection to the Internet > config settings: > [pap2] > type=friend > secret=something > qualify=yes > nat=yes > host=dynamic > canreinvite=no > context=private > callgroup=6 > pickupgroup=6 > callerid=name <1234567890> > disallow=all > allow=ulaw > allow=alaw > allow=gsm > dtmfmode=rfc2833 > > Clients behind single NAT with a Linksys WRT54GS default settings > Clients are 2 Eyebeam clients & 1 linksys PAP2-NA > > the audio has never worked consistently on the PAP2 only intermittently > with better results in calling the asterisk box directly but only rarely > when calling outside lines. > > I have now set the phones to register every 60 seconds with no change in > results. > > There was no change in the 'sip show peers' as no settings were changed, > all you had requested was the output. > > finally the "yup everything is there" was in direct response to your > statements in the previous email which asked me to confirm several things. > > sip debug doesn't reveal anything more. > > I hope this summery helps > > Thanks > > Miles > > > Steve Totaro wrote: >> N means NAT. No N no NAT. >> >> Can you call now with audio in both directions? Can you set the >> phones to register every two minutes (expiration)? Is the output from >> sip show peers still the same before and after the audio working? >> Does sip debug give any info? What type of router? >> More info is good! "yup everything is there" is a little hard to work >> with. >> >> Is this a double NAT or is your asterisk box on a routable IP? If it >> is double NAT, forget it. >> Thanks, >> Steve >> >> Miles Scruggs wrote: >>> yup everything is there: >>> >>> Name/username Host Dyn Nat ACL Port >>> Status pap2-2/pap2-2 123.123.123.123 D N >>> 5062 OK (93 ms) >>> pap2-1/pap2-1 123.123.123.123 D N 5061 OK (39 ms) >>> >>> I'm really confused why it has N for NAT when the sip settings listed >>> in previous post have NAT set. >>> >>> Thanks >>> >>> Miles >>> >>> Steve Totaro wrote: >>>> Make sure you have qualify=yes for each phone. Type "sip show >>>> peers" in the asterisk CLI and post the output when and when you are >>>> not able to make calls. Make sure that the new port settings are >>>> reflected in asterisk. >>>> >>>> Miles Scruggs wrote: >>>>> Well I just set the port to 5061, and no other devices on this end >>>>> have that port. I still have the same problems though. The >>>>> strange thing is that I have better luck calling the asterisk box >>>>> itself rather than an outside line, but even that is intermittent. >>>>> Actually what I have found is that after my SIP device restarts I >>>>> can call the asterisk box (but only once the second time it will >>>>> not send audio), but I can't call an outside line, well it calls, >>>>> answers, and bridges but no audio happens to pass. I'm really >>>>> confused. >>>>> >>>>> Miles >>>>> >>>>> Steve Totaro wrote: >>>>>> SIP uses port 5060 by default. Chances are your SIP phones are >>>>>> set to use port 5060 by default. Some phones have a tick box that >>>>>> says "Use Random Port" or you can specify a port. Start with port >>>>>> 5060 and move up so phone one would be 5060 phone two 5061 and so >>>>>> on. The problem is most likely that your Linksys is mapping port >>>>>> 5060 to the phone that has last sent data which explains why it >>>>>> works sometimes but not others. If your asterisk server is setup >>>>>> not to bind to a particular port for sip (sip.conf) then just try >>>>>> configuring the phones with unique ports and give it a try. >>>>>> >>>>>> It is still a good idea to use qualify=yes in your asterisk >>>>>> (sip.conf) for each extension since it keeps port mappings open >>>>>> and active on your linksys. Otherwise your Linksys port mapping >>>>>> may expire and an incoming call will be seen as unsolicited >>>>>> traffic and block it. >>>>>> >>>>>> Thanks, >>>>>> Steve Totaro >>>>>> >>>>>> Miles Scruggs wrote: >>>>>>> The asterisk host is connected directly to the internet, the >>>>>>> phones I am having issues with are behind NAT, but I'm only >>>>>>> having issues with some of them. Most specifically the phones on >>>>>>> my linksys PAP2 adapter. NAT at the remote location is provided >>>>>>> via a standard out of the box config of a Linksys WRT54GS >>>>>>> router. Here are the settings for the PAP2: >>>>>>> >>>>>>> [pap2] >>>>>>> type=friend >>>>>>> secret=something >>>>>>> qualify=yes >>>>>>> nat=yes >>>>>>> host=dynamic >>>>>>> canreinvite=no >>>>>>> context=private >>>>>>> callgroup=6 >>>>>>> pickupgroup=6 >>>>>>> callerid=name <1234567890> >>>>>>> disallow=all >>>>>>> allow=ulaw >>>>>>> allow=alaw >>>>>>> allow=gsm >>>>>>> dtmfmode=rfc2833 >>>>>>> >>>>>>> This is a situation where I do have multiple SIP devices behind >>>>>>> NAT, tell me more about using different port numbers for >>>>>>> different devices, and what other things should I look out for? >>>>>>> >>>>>>> Thanks >>>>>>> >>>>>>> Miles >>>>>>> >>>>>>> >>>>>>> Steve Totaro wrote: >>>>>>>> You need to describe your NAT setup more. >>>>>>>> One thing to try is to set qualify to yes or a short number. >>>>>>>> Essentially a keepalive for any routers in the middle. If you >>>>>>>> have multiple phones behind a remote NAT, make sure they are >>>>>>>> using different ports. >>>>>>>> >>>>>>>> Miles Scruggs wrote: >>>>>>>>> Using sip connections some peers are not able to transmit or >>>>>>>>> recieve audio. All peers are setup the same aside from the NAT >>>>>>>>> settings. The call will go through, called device will ring, >>>>>>>>> but when it answers there is no audio connection. From the >>>>>>>>> callee, they will not here the rings, only silence when they >>>>>>>>> dial the phone. >>>>>>>>> >>>>>>>>> The kicker is that sometimes it will work, and other times it >>>>>>>>> will not. >>>>>>>>> >>>>>>>>> Miles >>>>>>>>> _______________________________________________ >>>>>>>>> --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 >> > _______________________________________________ > --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
you blocking the RTP ports? (rtp.conf)
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