Re: [Asterisk-Users] Would DECT cordless phones work with Asterisk and VOIP?
Larry Alkoff wrote: I've just heard about DECT which is used for about 50 million phones in Europe and is just starting to appear in the US. DECT stands for Digitally Enhanced Cordless Telephone and supposedly has much greater range than other cordless telephony. Additionally, you can purchase repeaters that will greatly range. Since reading about poor audio quality and echo issues caused by repeated conversions as the signal traverses the path from the (possibly POTS analogue) station, over tcp/ip and to destination, would DECT (another digital form) agravate this? In my house, a Uniden 5.8 and Panasonic 2.4 cordless system would only work over about 35 feet indoors - not enough for a large house. Does anyone have any hands-on experience with DECT? I am so keen on getting the Kirk telecomms 600 system to hook to asterisk, anyone know where to get one from that will ship to New Zealand? smime.p7s Description: S/MIME Cryptographic Signature ___ --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
[Asterisk-Users] g729 status in New Zealand
Do I need licenses to use the codec in New Zealand? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] g729 status in New Zealand
As far as I was aware a license was only required in contries that had software patents, I know that there arnt here so I am just seeking clarification if thats all there is to it. [EMAIL PROTECTED] wrote: Yes. The G.729.x source code (in C) can be downloaded for free from ITU-T as the spec actually are in C code. But, even if you use this raw, non-tuned version you will need to pay a license fee to ITU-T per channel. The same for G.723.x The version that Digium offer is optimized, meaning that they have done basic improvments on the c-code to be able to run more channels. This is however very basic stuff. Jan Richard Malcolm-Smith wrote: Do I need licenses to use the codec in New Zealand? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] g729 status in New Zealand
trixter aka Bret McDanel wrote: If you know where to look there is another option out there that doesnt use either method, but I have doubts about how legal that one is, so I will not comment on that. Can someone give me some pointers for this? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
[Asterisk-Users] Asterisk on dynamic extrenal IP behind a nat router.
Is there no way to have asterisk determine its IP either via upnp or else resolve a dyndns hostname rather then having an entry in the config file? smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
Re: [Asterisk-Users] DECT gateways
Michiel van Baak wrote: Is there any other solution like this out there that works with asterisk ? If you find something, I would be interested in the outcome. I want something for the house here, at the moment I just have 2 analog dect bases plugged into the same line, but you cant roam between them so if I want to walk from one end of the section to the other I have to park the call, change base station and pick it up again, and then also multi handset doesnt work between bases etc etc. I would ideally like a dect network that plugs into the lan and then the handsets each register as an individual sip extension regardless of what base they are on at the time. I could really use a couple more bases around the place here to get good coverage. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] DECT VoIP Gateway
Is it just me that sees the post above as spam? If we (tinw) even consider buying stuph from spammers, then we are encouraging them in their sociopathic behavior, and as a consequence they will do more spamming. What is the consensus here? It is a product announcement for a new product that is of intrest to users of asterisk, therefore not untargeted spam (like viagra etc mails) - I'm fine with it. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] Ambient MD 3200 (X100P Clone)
Sandy Thomson wrote: Has anyone had any success with this card? Thank you. I am looking for a source for the clones in NZ - getting the real deal here isnt an option (killer shipping) and at the moment I am just having a play with asterisk and have given up on the internet linejacks I rescued from the skip at a previous employer. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] wi-fi phone advice
If it does materialize, im up for 3 or 4 of them at that price. Huddleston, Robert wrote: Well poo - if I can use that word I'm one of those poor family guys who loves to buy hardware on the cheap =) smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] International Caller ID?
David Phelan wrote: Anytime I receive a landline to anything over here in AUS, it comes up as Overseas I asked telecom why, and they said that the standard used doesnt support longer then 3+7 digits, so international numbers may not fit. I would still like to be able to send an NZ number with my calls however. Its amazing what thinking the call is from overseas does for peoples desire to answer it however ;) smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] International Caller ID?
Rod Bacon wrote: We have antiquated caller ID schemes here in Australia. We barely support numbers from other local carriers, let alone OS ones. Certainly no names either. When dialing out thru voipjet, I can put anything I like and it will come thru to my mobiles in New Zealand just fine (on both networks) - However calls to landlines just come up as on the caller ID as they put that for any international call. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] NAT and sip issues
I have an asterisk server behind NAT - no audio on the test external calls I have tried making so far. Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution evident from there, sounds like I have case 9. I would have thought that all I would have to do is port foward and have the external IP on the asterisk server, which I have done I have fowared 5060UDP, 8000UDP, and 35000 to 37000 UDP to the internal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed the xlite machines private IP address on some of the transactions that were logged. The client has a dynamic IP address so cant really be specified anywhere in the xlite configuration, I am also not sure on all the different firewall types. I was under the impression that there was no need to configure any portfowards at the sip softphone end. I will hopefully be using xlite or similar from a location with a very locked down firewall environment. I want to check all works on a normal nat router before trying it behind the nasty nat/firewall at this location. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
Re: [Asterisk-Users] NAT and sip issues
G.Marshall wrote: The rtp audio is going phone to phone, not via asterisk. This is one of the reasons I am trying to set up SER with Asterisk. I thought that canreinvite=no was supposed to force the audio to go via asterisk? smime.p7s Description: S/MIME Cryptographic Signature ___ 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
[Asterisk-Users] Looking for some real basic doccos...
Have asterisk installed, and working with my 2 quicknet phonejacks and 1 linejack cards. I cant seem to get my way thru getting the linejack to answer, and give some choices to the callers. I cant get the phonejacks to work when I change there entry to mode=dialtone in the phone.conf, thats all I get, a dialtone. Initially what I want to do is have all the phones but mine connected to one phonejack card, and mine to the other phonejack card. I want incoming calls initially passed thru to both cards, untill I device a suitable menuing system, but outgoing calls I want to restrict on the card that the housemates phones are connected to (So they cant call long distance, mobiles, or any other calls that cost) I have being looking at all the docco I can find, but its all really really complex looking, even the default configs make no sence to me. smime.p7s Description: S/MIME Cryptographic Signature ___ 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
