Brian McCarey
2005-Aug-07 09:55 UTC
[Asterisk-Users] Configuring Asterisk@home for Sipgate.
Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in Adobe pdf format at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050807/a8826a15/attachment.htm
Zachary Whitley
2005-Aug-07 10:19 UTC
[Asterisk-Users] Configuring Asterisk@home for Sipgate.
> I've posted my config files in Adobe pdf format at > http://www.brianmccarey.com/voip/sip > http://www.brianmccarey.com/voip/extensions > http://www.brianmccarey.com/voip/trunkI think you're either going to get complaints about the pdf files or people are simply going to ignore your question. Is there any reason you chose to post pdf's instead of just posting the ASCII files? And you're really going to hear it when people follow your link and find the file isn't there.
Brian McCarey
2005-Aug-07 13:46 UTC
[Asterisk-Users] Configuring Asterisk@home for Sipgate.
I've re-uploaded the config files in NON pdf Any help welcomed. Regards ----- Original Message ----- From: Brian McCarey To: asterisk-users@lists.digium.com Sent: Sunday, August 07, 2005 5:55 PM Subject: [Asterisk-Users] Configuring Asterisk@home for Sipgate. Hi all, I'm new to the forum. Oh no....newbie question coming, I hear you all cry! I'm playing around with Asterisk@home and have installed software and fiddled around with sip and extensions files. I have manage to make out going calls through Sipgate using X-Lite but cannot for some reason receive incoming calls. Incoming calls do not even show up on the switchboard panel. I've posted my config files in at http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've spent at least a couple of weeks trying to sort it out and am now seeking your good advice. Asterisk pc is attached to a small network which connects to the internet via a 3COM firewall broadband router. The Asterisk has an IP on the network off DHCP and it's IP is cleared through the firewall by DMZ setting. I'm signed up with sipgate.co.uk Any advice of sorting out incomming calls would be gratefully received. Thanks Brian. ------------------------------------------------------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users 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/20050807/6c3d2295/attachment.htm
Brian McCarey
2005-Aug-07 23:35 UTC
[Asterisk-Users] Configuring Asterisk@home for Sipgate.
Hi Arik, X-Lite is just set up as an extension for the Asterisk@home. I've used X-Lite because it appears to be a bet better at sorting out NAT firewalls for it's self. Once I get that working I can then plugin a Sipura 2001 box. I need the Asterisk to manage two Sipgate and one voipuser lines. If you have Asterisk working with Sipgate then it may be just compairing my config files with yours. My problem is that I can make out going calls through Asterisk but not receive incomming ones. http://www.brianmccarey.com/voip/sip http://www.brianmccarey.com/voip/extensions http://www.brianmccarey.com/voip/trunk I've set up a Sip trunk. Should it have been a IAX? Regards Brian ----- Original Message ----- From: "Arik Funke" <arik.funke@gmx.de> To: <brian_mccarey@hotmail.com> Sent: Monday, August 08, 2005 1:49 AM Subject: Re: [Asterisk-Users] Configuring Asterisk@home for Sipgate.> Hi Brian, > > I have Sipgate running on Asterisk (not Asterisk@Home though). I am not > sure what your problem is... > > You say: "I have manage to make out going calls through Sipgate using > X-Lite". > > What does this have to do with Asterisk? What does Asterisk do or not do? > I think I might be able to help you out if you give me a bit more info. > > Best regards, > Arik >
Hello Brian, attached are the relevant part of the config files I use. My Setup: I use internally only ISDN phones and only use SIPgate to dial out and receive calls. I do not use SIP phones. [sipout] demonstrates how to dial out on sipgate. Best regards, Arik === sip.conf ==[general] port = 5060 bindport = 5060 bindaddr = 0.0.0.0 qualify=no disable=all allow=alaw allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes canreinvite=yes language=de register => "sipg. no":"sipg. no"@sipgate.de/"sipg. no" [sipgate-out] type=friend insecure=very ; otherwise I get authentication errors nat=yes username="sipg. no" fromuser="sipg. no" fromdomain=sipgate.de secret="sipg. passwd." host=sipgate.de qualify=yes context=sipgate-in === extensions.conf ==[sipgate-in] exten => "your sipg. no",1,Dial(${INTERN}/29) [sipout] exten => _X.,1,SetCallerId,"your sipg. no" exten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trg) exten => _X.,3,Hangup asterisk-users-request@lists.digium.com wrote:> Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > > You can reach the person managing the list at > asterisk-users-owner@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. request for clarification on Asterisk T.38 bounty (Adam Megacz) > 2. Re: BudgeTone 100 Woes (Tony Mountifield) > 3. RE: SPA 841 form SIPURA (Paul Dugas) > 4. Re: z-machine + asterisk = fun! (Lists) > 5. Unable to connect to FWD (Balaji NJL) > 6. NT1 devices with analog ports on HFC based ISDN BRI cards in > NT mode and asterisk (chan_mISDN) (Nenad Radosavljevic) > 7. Re: BudgeTone 100 Woes (Jim Duda) > 8. VoicePulse Connect down Sunday evening? (Trent Tuggle) > 9. Re: VoicePulse DTMF Problems Anyone? (Mark Edwards) > 10. http://www.voip-info.org/ front page taken out by spammer > (Paul Belanger) > 11. Re: Re: BudgeTone 100 Woes (Doug Lytle) > 12. Re: request for clarification on Asterisk T.38 bounty > (Steve Underwood) > 13. Re: request for clarification on Asterisk T.38 bounty > (Kevin P. Fleming) > 14. Re: Does anyone run Asterisk on FC4? with Digium's TDM40B > cards (Kumara Jayaweera) > 15. RE: list of T.38 providers on wiki: please contribute > (Dean Collins) > 16. Re: Unable to connect to FWD (Balaji NJL) > 17. Re: Unable to connect to FWD (Tony Hoyle) > 18. Re: function declaration isn't a prototype (chris) > 19. Re: function declaration isn't a prototype (Dave Cotton) > 20. Re: Configuring Asterisk@home for Sipgate. (Brian McCarey) > 21. mysql sock location (Wei Kun) > 22. Re: BudgeTone 100 Woes (Tony Mountifield) > 23. ISDN BRI Funkyness (gw@adcomcorp.com) > 24. How to config voicemail with mysql? (Wei Kun) > 25. X100P with Caller-ID in Australia, anyone? (Jon Whitear) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sun, 07 Aug 2005 14:34:45 -0700 > From: Adam Megacz <adam@megacz.com> > Subject: [Asterisk-Users] request for clarification on Asterisk T.38 > bounty > To: asterisk-users@lists.digium.com > Message-ID: <x164uh8m8a.fsf@nowhere.com> > Content-Type: text/plain; charset=us-ascii > > > The bounty stands at $5,500. I'm seriously considering taking a shot > at it if I can find a decent T.38 provider to test with (I'm still > hoping for reliable PAYG T.38). > > It looks like a lot of very smart people have done a lot of very hard > work (t38modem, spandsp) that would go towards getting this working. > At this point it appears to be mostly a matter of integration > (libspandsp+asterisk), encapsulating T.38 inside IAX2 (not too hard), > and testing (tedious and time-consuming). Basically the easier but > less-fun part of the "big-picture" task. > > My main question is this: how is the bounty divided? Does the person > who does this "grunt work" get the whole $5,500, or does part of it go > to the authors of t38modem/spandsp (which would surely be a large part > of any solution)? > > I guess on one hand it would be unjust *not* to divide the bounty with > them, but on the other hand, if the bounty is to be divided, I think > the uncertainty about exactly how that would happen might be a factor > in why the bounty has gone unclaimed for so long. > > - a >