Am 01.06.2014 um 19:34 schrieb Adrian Hodgson <[email protected]>:
> UK based but guess that does not matter. > I have managed to get my hardphone to work with sipgate, incomming and > outgoing calls OK. > > But when I try to switch on the second account or line within the phone to > talk to astlinux then Sipgate becomes incomming calls only. It could be > because I am using port 5060 for sip on both accounts not sure? > > The end game is to have Astlinux register with sipgate then route to internal > phones, but, I get confused with all the examples out there, some refer to > trunks for incomming calls only! > > I have a simple residential sipgate account, so I would assume that I make > the entry into sip.conf, I think I could use type=peer to match on the > registration to sipgate, is that correct or should it be friend as calls will > go both ways, this is one area I get confused. > > I want to do the usual 9 to dial an outside line and direct these over > sipgate, but first steps are to get an incomming call to go to an internal > phone. > > Anyway following an example from sipgate as below: > > In sip.conf I have the following insertion > with bogus id and password > > ;-------------------------------------------- > > register=>1234567:[email protected]/1234567 > [sipgate] > type=peer > secret=abcdefg > insecure=invite > username=1234567 > defaultuser=1234567 > fromuser=1234567 > context=sipgate_in > fromdomain=sipgate.co.uk > qualify=yes > disallow=all > allow=alaw > dtmfmode=rfc2833 > > ;--------------------------------------------------- > > in extensions.conf I have: > > > ;---------------------------------------------------------------------- > [sipgate_in] > exten => 1234567,1,Playback(hello-world) > ;exten => 1234567,1,Dial(SIP/000b822d88d8) > same => n,Hangup() > > ;---------------------------------------------------------------------- Here are the settings from Sipgate Germany, which work fine for some of my customers: http://www.sipgate.de/faq/index.php?do=displayArticle&article=540&id=257 > The hope was that I would get "Hello World" but all that happens is like a > direct hangup. > > The sipgate webpage says that there is a successful registration and just > shows calls to my account as not answered. When I check the status page of > astlinux there is no messages to show it is even hitting asterisk. > > Could I have router problems? > > I have not set any port forwards at this point in my router, or placed > astlinux in any DMZ. You need to forward 5060 UDP and the configured RTP ports (by default 10000-20000 in rtp.conf) to your AstLinux box. > I have also changed my hardphones so that each one operates on a different > local SIP port and RTP port, such as 5160, 5260, 5360 etc and the RTP's 5104, > 5204, 5304 etc. Calls between hardphones still works OK. > > Can anyone assist or point me in a direction? > > Cheers > > Adrian Michael http://www.mksolutions.info ------------------------------------------------------------------------------ Time is money. Stop wasting it! Get your web API in 5 minutes. www.restlet.com/download http://p.sf.net/sfu/restlet _______________________________________________ Astlinux-users mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [email protected].
