Re: [asterisk-users] UK English sounds packs
On Thu, May 26, 2011 at 02:28:31PM +0100, Steven Howes wrote: On 26 May 2011, at 14:09, Ishfaq Malik wrote: Does anyone know if there are any free UK accented English sounds packs? http://www.tel.net Not perfect, but damned near :) If anything's missing please let me know and I can get stuff corrected (in fact I need to sort the extra sounds stuff). Jay will also do any specific recordings for people (for a very reasonable charge). Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PostgreSQL is asterisk friendly with it?
On Mon, Sep 13, 2010 at 12:31:55PM -0400, Vince Vielhaber wrote: change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully [snippage and probably off topic] Why are you worried re the future of MySQL and it's new owners Oracle. a) MySQL is open source so anyone can take a fork and continue development. b) Oracle own InnoDB already which is the main storage engine for MySQL. c) Oracle dont have any low end DB products for start-ups etc. Developer licenses may be free, but commercial use certainly isn't. d) MySQL is now a business division within Oracle and MySQL (.com) makes money. e) Google have contributed a lot of code for MySQL v6, I'm sure they'd take it on if Oracle in madness decided to drop it. f) I'm sure Oracle will push Oracle on Sun hardware/Solaris for high-end DB platforms and optimise the two so the performance stats look great and so they'll develop a migration platform so high-end MySQL users can easily migrate to Oracle when the need arises. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: UK PPP certification -- what is it?
On Fri, Aug 13, 2010 at 12:46:51PM +0100, Faris Raouf wrote: They mean PhonePayPlus (formerly ICSTIS). www.phonepayplus.org.uk I am not aware of them certifying particular phone systems. Rather, they impose certain requirements and obligations on the service provider depending on the nature of the service being provided and the number range it is provided on. But maybe more stringent regulations and phone system certification does apply to certain types of service which I've never had to deal with - adult stuff, for example - so I'd give them a call if you can't find the info on their website. PhonepayPlus are the 'regulator' for premium phone services in the UK (well they're independent but work with Ofcom the regulator). It depends on what services you're offering, but the rules are pretty stringent to stop fraudulent use of PRS numbers and top stop scames etc. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Femtocell to VoIP?
On Wed, Aug 04, 2010 at 01:13:56PM -0400, Matt wrote: Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming up very dry. OpenBTS sounds like it would work, but is way too expensive to roll out to residential homes. Pretty much all Femtocells use 3G locally and send stuff back over VoIP (in some form or other). In the UK Vodafone sell a 3G femtocell (which has an internal 2G radio too, to ensure it's being used in the UK). ATT sell their own. Try contacting PicoChip or Ubiquisys who both have femtocells. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Femtocell to VoIP?
On Mon, Aug 02, 2010 at 03:36:59PM -0400, Matt wrote: Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? Most people seem to be concentrating on 3G femtocells (there are various companies making designs based on picoChip soft radios). OpenBTS can be used (and there have been some successful quite large installations). Hay Systems were meant to be producing a 2G (GSM/GPRS) femtocell, but they seem to have gone quiet. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk, Skype For SIP
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote: As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. [lots snipped] With Skype's just released SkypeKit it should be possible to build any application with Skype support (costs $20 to register as a dev), they've now got libraries for Linux and now Windows and MacOS X. SkypeKit is basically a headless Skype client. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hosted PBX in the UK
On Wed, Jul 14, 2010 at 10:27:13PM +0100, Wipe_Out wrote: Might be off topic but I thought it would be a good place to ask.. I am investigating switching to a hosted PBX and dumping my old Asterisk box thats been running in my office for the last few years.. The few I have found seem very expensive.. There's several (some being on this list) Gradwell.com cone to mind You could also look at pibix.com who are in early stages Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM Gateway
On Mon, Feb 08, 2010 at 02:52:33PM +0200, Peter wrote: I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Portech GSM gateways tend to work quite well. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi GSM handover
On Sat, Oct 10, 2009 at 10:16:43AM +0200, Patrick wrote: Thank to Frank and Steve for your answers My understanding is that you need to place on operator premise an equipment that checks first the availability of the user on VoIP. If not registered, it's routing the call through the cellular network. Is it correct ? But during the handover (wifi to GSM), how does it works ? Is it the operator that initiates a call on the GSM network. If so, I guess the mobile device need to have some logic to seamlessly switch between the 2 channels, isn't it ? If it's the mobile device that initiates the call to the GSM network, it will also require some logic to do that. So my question is, is the handover something standard in every mobile device supporting GSM and VoIP or do you require an extra piece of software to do the trick ? Is this principal applies to every transport technology, I mean VoIP through WIfi or VoIP over 3G ? GSM calls are handled by an MSC (which is an SS7 switch) that talks to BSCs (basestation controllers) which talk to BTS (basestations), of course MSCs also talk to other MSCs. The GSM operator will have a UMA gateway in the network. A UMA phone will 'listen' for both GSM and WiFi and if it detects that the WiFi is 'known' it connects to that and it will connect through to the UMA gateway and the GSM network will switch the call to WiFi, if the user wanders off the WiFi area it will switch back to normal GSM operation. So the phone has to be UMA aware and the operator has to support it. On a normal GSM phone it is possible to write software that will switch calls between VoIP and GSM but you then generally have to control the endpoint of the call, so the GSM call usually goes through a VoIP access system and the software will switch the call to VoIP if it can, but the end-point is always the VoIP system that then calls the real number dialed. i.e. when the user dials a number it doesn't really go to that number directly, goes through the VoIP company who then can switch the transport in-between them and the handset. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wifi GSM handover
On Sat, Oct 10, 2009 at 03:15:20AM +0200, Patrick wrote: Hello guys, I'm wondering what is required and involved in order to provide a wifi/GSM handover to customers. After googling I haven't found any product/vendor. Do you have an idea ? That's called UMA and you need operator cooperation. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is Enum safe from spammers?
On Tue, Jul 14, 2009 at 06:46:50PM -0500, Karl Fife wrote: [snip] missed the original message - Original Message - From: Gordon Henderson gordon+aster...@drogon.net To: Asterisk Users Mailing List Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 14, 2009 9:14 AM Subject: [asterisk-users] Is Enum safe from spammers? Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... As a Director of UKEC Ltd (the governing body of ENUM in the UK) I'd be interested in knowing more about this. Is anyone using Enum? Currently there is a need to populate the ENUM database. UKEC and Nominet are working together to try and get vendors to support ENUM. Does anyone (other than cynical old me) think that Enum is a spammers best friend? ENUM isn't just about VoIP, it allows end users to set policies on how they want to receive calls. Unfortunately not many telcos yet support ENUM (or public ENUM anyway). The most likely growth area are ITSPs populating the ENUM database with their customer's numbers. Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) If you find out, please do let me know. I can see that Enum is good to provide another way round the PSTN, but at the same time, I'm just not convinced... ENUM is the future of telephony, it's just needs mass adoption. Unfortunately there are likely to be at least 3 ENUM systems in the UK. * Public ENUM as in e164.arpa * Carrier ENUM whereby telcos use ENUM to route calls to other telcos. * Eventually a central porting database for mobiles (and also fixed lines) which uses ENUM to store the port information. It would be good if these all merged into one body. What do others think? Happy to have a chat off-line. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 issue?
Just founnd a weirdy. My end is Asterisk 1.2.32 using an IAX2 link to the US. The IP address of the remote end changed (though in the config file it's registered as a name i.e. asterisk.remote.end), my system didn't recognised the IP change, it must be cached once and then the cached value used for ever. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 issue?
On Tue, Jun 09, 2009 at 02:02:50PM -0500, Danny Nicholas wrote: Did you do an IAX2 show peer on it? Remote end unreachable and old IP address Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] OpenBTS chat with David A. Burgess
On Sat, Mar 21, 2009 at 09:39:47AM +0100, randulo wrote: Hi, The OpenBTS Project is an effort to construct an open-source Unix application that uses the Universal Software Radio Peripheral (USRP) to present a GSM air interface (Um) to standard GSM handset and uses the Asterisk software PBX to connect calls. The combination of the ubiquitous GSM air interface with VoIP backhaul could form the basis of a new type of cellular network that could be deployed and operated at substantially lower cost than existing technologies in greenfields in the developing world. This looks like a great project, sorry I missed the call. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Upgrade to v.1.2.31 ... weird change
On Wed, Jan 14, 2009 at 02:56:44PM -0200, Leonardo Gomes Figueira wrote:
Tilghman Lesher escreveu:
On Monday 12 January 2009 01:26:02 pm Steve Kennedy wrote:
I think it happened when I upgraded an install to 1.2.31
The variable CALLERIDNUM no longer works and CallerID(num) has to be
used.
I don't see why not. There has been no change whatsoever to that body of
code.
I think there is some mistake on his test about CALLERIDNUM. Did a quick
test here on 1.2.31 and it's working fine.
I'll check, but something definately changed. I think I was using
${CALLERIDNUM:1:4} anyway didn't work as planned. Changing to
CallerID(num) and it worked again.
Steve
--
NetTek Ltd UK mob +44 7775 755503
UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk
Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade to v.1.2.31 ... weird change
I think it happened when I upgraded an install to 1.2.31 The variable CALLERIDNUM no longer works and CallerID(num) has to be used. I know the initial one was being depreciated, but I didn't see any mention of it. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM / 3g channel bank
On Wed, Oct 01, 2008 at 12:53:41PM +0100, Julian Lyndon-Smith wrote: More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are so much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30 sims to be connected, and we are based in the UK. Any advice is gratefully received. Are you providing any kind of service to 3rd parties? If so you are NOT allowed to run a GSM gateway. If it's purely your own traffic i.e. say company PBX to mobile traffic then it is allowed. Ofcom ruled on this a while back. Saying that I've used a Portech GSM gateway on SIP and it works well. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] uk tole-free dids?
On Mon, Sep 29, 2008 at 09:17:11AM -0800, Babcock, Michael Alex wrote: right will stay away from them, smile. On Mon, 29 Sep 2008, Babcock, Michael Alex wrote: what are 70 numbers? Prefix 070 (then 8 more digits) These are so-called personal numbers. They're a blot and an anomaly. They are expensive to call and the recipient usually gets revenue from the calls. ie. they are premium rate, revenue generating numbers in disguise. Worth noting that UK premium rate numbers are covered by PhonePayPlus (the regulator for PRS). http://www.phonepayplus.org.uk/consumers/faq/default.asp 070 MAY be covered if they are used for PRS services (not for personal numbering). Ofcom are intending to move PRS out of 070 (and want to make all 07 mobile). There are oddities of course, the Channel Islands adhere to UK numbering plans by agreement with the UK gov (or however it works) but Channel Island mobiles and landline numbers are treated as foreign calls even though they are in UK number space. Great isn't it. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco acquires Jabber
On Sat, Sep 20, 2008 at 12:18:42PM -0400, Dean Collins wrote: No I know they just bought the company and not the protocol basically they bought engineering bums on seats. [1]http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.ht ml Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. You could possible see different Cisco devices communicating with each other (or even using an api to communicate with other manufacturers devices) eg, you might have an XMPP api to 'discover' appliance functionality or to communicate status updates. Jabber.com are in some big US gov departments, these are probably just the bodies Cisco want to get into with their UM systems. Making Cisco's UM based on Jabber and buying the expertise probably is a wise move for them. Also gives them interoperability with other systems ... Then move it down into the SME market as Linksys appliance. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number portability in other parts of the world.
On Thu, Jun 26, 2008 at 12:30:55PM -0400, Alexander Lopez wrote: I think it would be a good idea to start an item in the Wiki about this. Can anyone else chime in for their countries?? Others in the EU, Eastern, Far East? So Far I have: Australia:PSTN to PSTN and Cell to Cell are OK , but Cell to PSTN and PSTN to Cell are NOT OK.Dean Collins Poland: Not Today but possibly in 2009 Daniel UK: Portable if Telco has a porting agreement. Not all Telco have agreements in place. Steve Kennedy France: Porting from France Telcom to another provider not an issue, however if porting between other Telco's, Telco's must have porting agreement between them. Randulo In the UK numbers can be ported between fixed operators and mobile operators, but not (yet) between mobile and fixed (but then the distinction is blurring). Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number portability in other parts of the world.
On Wed, Jun 25, 2008 at 10:49:18AM -0400, Alexander Lopez wrote: Are phone numbers portable in other countries? Depends what country Are the same rules and conditions that exist here in the States mirrored elsewhere? How does a person in Europe go fully VoIP and still keep the main number? In the UK numbers are portable, though the telco wanting the number must have a porting agreement with the telco that has the number. Not all telcos have porting agreements. Do they use call forwarding? Can do. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
On Mon, Jun 23, 2008 at 08:24:42AM -0400, Matt Watson wrote: On June 23, 2008 08:08:53 am OCG Technical Support wrote: I little more digging and I confirmed that cell phone VM and FAX waiting icons are in fact controlled by a proprietary SMS message format. Here's what I found: [snip] could you provide a link to where you got the info from? I'd be interested in seeing if i can get this to do anything useful. You should be able to get most of the info from www.3gpp.org Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling cell phone VM / Fax waiting notification icon for asterisk VM
On Mon, Jun 23, 2008 at 11:03:49AM -0400, Jay R. Ashworth wrote: On Mon, Jun 23, 2008 at 08:24:42AM -0400, Matt Watson wrote: Now the tough part...does anyone want to create an app to send notification to a cell phone to set/clear these bits? could you provide a link to where you got the info from? I'd be interested in seeing if i can get this to do anything useful. Good luck. The strong possibility exists, though, that you'll find the customer-exposed API doesn't have a path to set the bits you need. If you have access to an SMSC you can generally set most of the bits using SMPP/UCP/etc. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Out-Going Calleriid
On Wed, May 07, 2008 at 07:46:59PM +0100, Tim Guy wrote:
Installing a new box onto UK NTL (Virgin Media)
During testing phase the callerid worked, now it doesn't.
Can someone confirm that my syntax is right before I start ripping the
configs to bits
exten = _9.,1,Set(CALLERID(number)=01926xx)
exten = _9.,2,Dial(ZAP/1/${EXTEN:1})
Ive tried all permutations of the CALLERID (ie CALLERID(NAME) and
CALLERID(NUMBER) but it just wont work anymore.
Zapata has the following relevant settings
usecallerid=yes
hidecallerid=no
callwaiting=yes
I presume you're running a PRI or BRI line from Virgin? If you have an
analogue line it's unlikely you have the ability to set your CLID.
If you have an ISDN variant, then you should be able to set it, but
generally (depending on how the switch is set-up) only to numbers
associated with that line. It also depends on the number of digits that
the switch expects i.e. it may just be 1926xx, or it may well be
just the xx part.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: UMA in UK, any use?
On Mon, Apr 21, 2008 at 11:02:13AM +0100, Mike Dent wrote: sorry for off topic post, struggling to find any information on UMA in the UK. I have a Blackberry 8320 phone with wi-fi and UMA capability, its actually an unlocked Orange branded phone. T-Mobile don't support UMA in the UK, is it possible to do anything else with the UMA feature of this phone? Or, is it totally locked to your network provider? Any possible way of hacking it to work as some kind of voip client to work on one's own implementation of UMA, if such a thing even exists? :) UMA is completely tied to the operators. It requires back-end technology to transfer the call from GSM/3G to WiFi or vice versa. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best alternative for getting prompts recorded.
On Wed, Mar 12, 2008 at 03:03:38AM +0530, [EMAIL PROTECTED] wrote: Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source) and would convert Text to Speech. I tried Fetival, but the quality of the sound is not good. Can we improve the sound quality of Festival somehow. What language do you want? There are UK English versions of the 1.2 release available from www.tel.net, I should have the 1.4 version up very soon. They were recorded by Jay Benham (links from the site). In terms of text-to-speech there are several solutions, have a look at www.talklets.com who run a text-to-speech (well accessibility, but TTS as part of it) software as a service solution. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How can I call cheap to UK cell phones
On Tue, Feb 26, 2008 at 01:38:31PM -0600, Erik Anderson wrote: On Tue, Feb 26, 2008 at 1:19 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Greetings, How can I call cheap to UK cell phones. I am located in Toronto, Canada, but need to call UK cell phones both from Toronto and London. I'd guess you could get an account with one of these providers: http://voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#UnitedKingdom The termination rates are set by Ofcom www.ofcom.org.uk around 6-7p per minute dependent on network. BT offer blended rates (i.e. same rates for mobile/landline/etc), but you have to originate from outside UK. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk gateway
On Thu, Jan 31, 2008 at 05:59:03AM +0800, Sam Tam wrote: Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm. err biz again ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP GSM
On Tue, Jan 29, 2008 at 09:12:11AM +0800, Sam Tam wrote: Try cyber-telecom.net May be get a X100P with a CT-G1000 or G2000 a) this should be on the biz list b) why don't you post from your cyber-telecom.net address? c) it must be the end of the sales cycle and trying to get a bit more revenue in? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM SIM Cards and Digium, or GSM SIM Adaptor
On Mon, Jan 14, 2008 at 08:20:11AM -0500, Steve Totaro wrote: On Jan 14, 2008 7:42 AM, bilal ghayyad [EMAIL PROTECTED] wrote: Is there an Digium cards support GSM SIM cards so we can fix an SIM card to be used for calls within mobiles as it is less rate? Or I have to use an FXS to SIM adaptor? If yes, then anyone advise a models and prices? I don't think Digium do one, but Junghams (sp) do. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone to be installed on the Mobile
On Wed, Nov 21, 2007 at 11:35:42AM -0500, Dean Collins wrote: There's an application server that sits between asterisk and the gprs network that can switch calls real time between wifi, your office pabx extensions and the gsm network. I've forgotten the name of it but I remember it costs $US6,000 for 10 licenses. If you want me to find out more I can spend the time to look into it but only after you've said yes to the budget. There are several methods allowing you to do this, but most are operator dependent. UMA (unlicensed mobile access), which is an unfortunate name as in several countries (including all of EU) all spectrum is licensed (though some is license excempt - which isn't unlicensed). This uses a home basestation which connects back to the GSM operator over IP. There's basically a switch in the GSM core which can flip between the call over GSM or WiFi. In the UK BT call this Fusion (in conjunction with Vodafone). Companies like Truphone run a software shim on the phone. When you make a call it actually prefixes the outbound call with a Truphone prefix, so if it's via GSM the call actually goes through them. That way they get termination revenue from the mobile networks, which hopefully covers [1] the cost of the actual onward call. If in range of a WiFi network the phone establishes a connection to an end-point in Truphone's network and switches the call (and if the WiFi degrades, it switches back to GSM). Both the above assume that the calls terminate on a system controlled by the network (or Truphone), so that sessions can be controlled by them. It would be possible to do this with software in the phone and all calls terminating on your kit which then passes the call on to the PSTN, so if WiFi is available it can originate the call from there, but it needs to switch to GSM/etc if you move out of range - so you need to be in control of both end-points of the call. Steve Note [1] Truphone took T-Mobile to the High Court in the UK as T-Mobile refused to route their number block. They won and T-Mobile were forced to route Truphone's numbers. The court also ruled that as they were providing a VoIP service, they should only get sub-pence termination rates - which means they don't cover onward call rates (as they terminate to traditional telcos), so although Truphone won the battle, they pretty much lost the war. -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile phone codecs ...
On Thu, Nov 01, 2007 at 01:09:24PM +0100, Benny Amorsen wrote: AM == Anselm Martin Hoffmeister [EMAIL PROTECTED] writes: AM Maybe the GSM codec is implanted to the GSM chip and that one AM does alaw, ulaw... Also, modern handsets like the E90 rarely use the plain GSM codec. They use newer codecs such as EFR whenever possible. Asterisk probably won't support EFR anytime soon; it is patent encumbered and not particularly suited for LAN or WLAN use. GSM networks are circuit switched and use 13Kb/s slots (i.e. GSM codec is 13Kb/s codec). 3G should just be data, so there's a wider choice of codecs. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Wed, Oct 10, 2007 at 02:10:54PM -0400, SIP wrote: [snip] I think that using 1.5.x as the name for a release candidate for 1.6 is pretty close to as unintuitive as it can possibly be. 1.6.Xrc-Y is a strikingly MORE intuitive naming scheme for 1.6 release candidates. mutt uses the x.y convention where y is odd for a development branch and y is odd for a release branch. So 1.5 would be the development of 1.4 etc. When it's stable a 1.6 would be released which would only have bug/security releases, any new features etc would go into 1.7. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to handle + prefix
On Fri, Aug 31, 2007 at 10:03:07AM -0600, Kai-Uwe Jensen wrote: On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote: Mindfully wanting to use a + instead of knowing the international access code seems like willful ignorance to me. I beg to differ. Consider cell phones as an example. They all provide + keys. And it is considered a best practice to store phone numbers in address books as + country_code area_code number so that you can change locations (ever traveled to Europe?) easily, without having to reprogram all your contacts' numbers. Also all SMS are sent in GSM international format (i.e. all prefixed with +CC). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.2.20.1 and 1.4.5.1 released
On Wed, Aug 22, 2007 at 01:19:14PM -0500, Asterisk Development Team wrote: The Asterisk.org development team has announced the release of Zaptel versions 1.2.20.1 and 1.4.5.1. These releases are to correct an error in the install target in the Makefile of the 1.4.5 and 1.2.20 Zaptel releases, as well as a handful of other issues. See the respective Changelogs for more details. Both releases are available as a tarball as well as a patch against the previous release. They are available for download from downloads.digium.com. Don't seem to be on www.asterisk.org (1.2.19 and 1.4.4) Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
On Tue, Jul 31, 2007 at 03:05:32PM -0400, Jay R. Ashworth wrote: On Tue, Jul 31, 2007 at 01:37:00PM -0500, voiplist wrote: I have done this in the past and I don't recall ever finding any popular music by popular artist. For example, if I wanted to play oh I don't know an original song performed by the original artist such as Nora Jones or The Beatles will I find this sort of thing at a Royalty Free Site? Certainly not. :-) Even if you can find non-original-artist recordings of such music, the *compositions* are registered with BMI and ASCAP, and you'll need blanket licenses to play them. (Well, if you only wanted one or two tracks, you might negotiate specific licenses, but I'm not sure it would be cheaper.) In the UK MCPS-PRS have standard rates for music-on-hold, it all depends on the number of lines. Their website has details. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
On Tue, Jul 31, 2007 at 05:22:20PM -0400, Jon Pounder wrote: Quoting John Millican [EMAIL PROTECTED]: there are plenty of radio stations with internet feeds of their audio, piping that in would not change any coverage area since anyone with internet could listen anywhere already, you're only providing that to the listener through a phone handset instead of a computer speaker, which amounts to just another audio device controlled by an internet connected computer. No it's not, you're rebroadcasting and that would incur a difference license (if legal at all). What if the radio is on in the background when I make a call ? is that rebroadcasting ? kind of gets blurry on the definitions there. That's not as you're listening to it and not trying to rebroadcast. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] improved SMS?
On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister wrote: Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride: Newbie question(s): From what I can determine it sounds like the SMS messaging isn't as robust as it could be (?). I'm wondering if there's active work on that right now or if it's more of an issue about PSTN carrier that one would be using who would be responsible for passing the messages into the PLMN. Background-- I'm looking into the possibility of setting up an emergency messaging system here at the University that would send out voice, SMS, and emails. Any input relevant to that goal would probably be appreciated. Hi Russ, my personal experience with short messages is that the system sometimes chews on them for minutes, sometimes several hours, even inside one mobile network, from cell phone to cell phone. This surely screws using it as a primary tier emergency system, but as a backup after e-mail and automated phone-out that could be OK. Sending from web-interfaces or via Uwhatever-that-protocol-is-called will not improve the overall performance. SMS was never designed for guaranteed delivery (or guaranteed timed delivery). There are options for messages to time out if they're not delivered in a specified time, or new messages can override old messages that haven't been read yet - but delivery isn't guaranteed. A phone sending an SMS will try and establish a connection (sort of) all the way through to the receiving phone and then deliver the message, if it cant it will be sent to the receiving network's SMSC which will then try and deliver it. If it gets put into a queue then the delivery time will vary drastically depending on the load on the SMSC and other network characteristics. Fixed to SMS always goes through an SMSC, so delivery times vary. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suing Dell||Dull Computers for CID abuse
On Wed, Jul 04, 2007 at 08:06:49AM -0500, Lacy Moore - Aspendora wrote: On 7/3/07, Joe acquisto [EMAIL PROTECTED] wrote: Contrary to the opinions of Anglo-Philes, we, here in the Colonies, speak American, not English. In some places, 'Murican. Merkins speaking Murican ... We get to do that, because, back in the late 1700's . . . we won. We let you win, you were terrorists and England's never been good at fighting terrorists. Now you're having the same problem !!! It is only referred to as English out of a sense of compassion. American English ... Oh, so anyway, who was guy Eng you named the country after? And who was America named after ? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote: Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Has anyone written a configuration tool for Cisco's that generates the correct XML files? Wouldn't that be a useful thing? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
On Fri, Jun 22, 2007 at 08:06:39AM -0400, Dave Bour wrote: So I'll ask the question. What's wrong with top posting. I use a blackberry to read most of my email, and bottom posting means excessive scrolling, often waiting to download additional content resulting in higher usage fees and rsi on my thumb for scrolling 90% of messages including all general email conversations are too posted yet discussion groups want bottom posting. Why? I dont know What's the answer? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] shorting flash time
Is there anyway to change the flash time on a TDM400 phone port (a user has a phone that seems to generate a short flash which isn't being picked up). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] shorting flash time
On Wed, Jun 06, 2007 at 08:46:20AM -0500, Eric ManxPower Wieling wrote: Steve Kennedy wrote: Is there anyway to change the flash time on a TDM400 phone port (a user has a phone that seems to generate a short flash which isn't being picked up). I suspect the phone us going off hook every once in a while to check if there is a stutter dialtone. If there is, it can light it's message waiting light. Don't you love the analog world? No, there's a button which they press which is generating a short break. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] VoiceMail Access
On Mon, May 21, 2007 at 03:10:23PM -0400, Lee Jenkins wrote:
Mike Hammett wrote:
I was looking at the ILECs? web sites to determine how their users
access voicemail.
What method should I use for my users checking their voicemail? Can
Asterisk voicemail be made to accept hitting * during the greeting to
enter the voicemail system? If they call their own number, how do I get
Asterisk to recognize that and take them to the voicemail system?
A common approach is to use the caller id in combination with some digit
sequence. For my systems, I've just used 555 as the VM extension.
exten=555,1,VoicemailMain(${CALLERID(num)})
For access to the VM from outside the system, I've used an AGI script to
query a database to validate the user.
It's also quite easy to set-up if you call your own extension number
from your extension it goes into voicemail for you extension.
You can have another number as above to access voicemail from another
extension.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--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] zap fallback
I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] zap fallback
On Fri, May 18, 2007 at 12:54:19PM -0500, Matthew Fredrickson wrote: On May 18, 2007, at 11:50 AM, Steve Kennedy wrote: I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Asterisk does not do dialtone detection before it starts using a zap channel. That's probably why you are seeing this behavior. Oh well, have to think of another way around this. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] GSM Cards for Asterisk (UK)
On Wed, May 16, 2007 at 09:15:49PM +0100, Matt Brown wrote: [snip] No, this client has a number of engineers all over the UK and they have a large mobile contract with several handsets - their current tariff includes free calls to other mobiles under the contract so what they are trying to achieve is operators in the call centre calling their own engineers get to make the call for free. The plan was to get another 4 or so SIMS from the mobile company and slot them into a PCI based device, having a pre-defined list of mobiles and then these calls are routed via the GSM card and network. I assume this would be acceptable ? In which case I presume that they are unable to find a GSM gateway in the UK ? Yup sounds fine. There are several vendors of external multi-SIM gateways, prices vary considerably. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] GSM Cards for Asterisk (UK)
On Wed, May 16, 2007 at 02:17:11PM +0100, Matt Brown wrote: I am currently building a 1.4.4 Asterisk box for a client and they are interested in GSM functionality. Does anyone have any experience with a GSM card, preferably Quad Span (4 GSM modules or higher) for use in the UK. I have seen the Ensure the client is ONLY using for their own use (i.e. they're not handling ANY 3rd party calls through their system) or their operating in an illegal manner. Ofcom does allow GSM gateways to be used for your own use. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] List of telemarketers??
On Sun, May 13, 2007 at 01:54:08AM +0100, Chris Bagnall wrote: 3. a list of bogus entries..so when you look at it, you know it's a fake phone number...one that recently came in that got me thinking this was 407 111 . I don't know much about the legal position over the other side of the pond, but I'm pretty sure that in the UK caller ID spoofing is illegal. There's nothing to stop you withholding your CLI of course, but to deliberately fake someone else's CLI (whether it exists or otherwise) pushes you over the line. Is the same not the case in the US? I don't know if it's illegal (it would fall under the Comms Act if it was), but I know it's discouraged. There are legal reasons you might spoof CLI. Most telcos will have agreements that end-users can't do nasty things with CLI (withholding doesn't actually block anything, just flags the CLI should be withheld, so telcos, law enforcement etc still get it). Quite a few SS7 providers will allow customers to do what they like with CLI, just have an agreement they wont do anything they shouldn't. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] force outgoinc callerid
On Thu, May 10, 2007 at 06:36:53PM +0200, nik600 wrote:
i have a Te205P connected to a PRI E1, can i force the outgoing
callerid to change for each context?
for example:
[outgoing_context_one]
;force callerid to 12345
exten = _XXX,1,Set(CALLERID(number)=12345)
exten = _XXX,1,Dial(Zap/${EXTEN})
[outgoing_context_two]
;force callerid to 2
exten = _XXX,1,Set(CALLERID(number)=2)
exten = _XXX,1,Dial(Zap/${EXTEN})
Can i do that?
thanks to all
Assuming your telco allows it.
Steve
--
NetTek Ltd UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
___
--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
Re: [asterisk-users] IAX Trunk
On Thu, May 03, 2007 at 01:22:07PM -0300, Ronaldo wrote: Can you suggest me any documentation about using IAX trunking? Thank you. There are examples in the iax.conf files I think, but basically just put something like [iax-toremote] type=friend username=whatever secret=somesecret auth=plaintext host=somewhere.com peercontext=some-context qualify=yes trunk=yes then you dial with Dial(iax2/iax-toremote/number) Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] IAX Trunk
On Thu, May 03, 2007 at 03:23:16PM -0300, Ronaldo wrote: OK Steve, Just one more question. Using this configuration can I make more than one call at the same time? The whole point of trunking is to support multiple calls down the same IAX trunk (well actually down the same packets). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] IVR dictionary dial-plan
Does anyone know of an (E)AGI or program to develop a IVR dial-plan which will take a list of words and then do something when a unique branch has been found. i.e. Say there's 3 words demon deacon bishop On a phone they'd be represented as 33666 332266 247467 So if the user enters 2 we know they want bishop if they enter 336 they want demon and 332 they want deacon. Could run the dictionary through a script which could generate the dial-plan or do it via some script interactively. Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Asterisk brute force watcher (was FYI)
On Thu, Apr 26, 2007 at 06:46:41AM -0400, J. Oquendo wrote: Steve Totaro wrote: I suspect that this will happen more and more. I also suspect that many people who have weak SIP credentials like user=100 secret=100 will be the victim of toll fraud and worse, call to 900 and other very high termination rates. How does $25 per minute sound? Ashtray is an Asterisk brute force watcher. Checks logs from cron and emails admin of potential brute forcers http://www.infiltrated.net/scripts/ashtray Can have it set in .bash_profile so whenever you log on, you'd see anomalies. With FC5 had to change to $8 and $11 Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] FYI
Just been getting lots of failed SIP registrations to a system here. All coming from Taiwan. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] UK zaptel and zapata.conf for TDM400P
Has anyone got a sensible zaptel.conf and zapata.conf for 2 TDM400P's working with UK set-up. They're set-up with 7 analogue phones and 1 PSTN port. Currently zaptel.conf has fxoks=1-7 fxsks=8 loadzone=uk defaultzone=uk It's really zapata.conf that would be useful. Currently using the zaptel/asterisk that comes with Ubuntu (latest) which needed a bit of tweaking (1.2.16), but could compile latest 1.4 release. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] sending an SMS via Asterisk?
On Thu, Apr 19, 2007 at 08:36:12AM -0400, Steve Totaro wrote: Just a thought, try kannal, use system in your dialplan and call lynx with a properly formatted URL for Kannal. Or indeed Kannel (www.kannel.org) Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] UK PRI and outgoing CLI FYI
Just a FYI to the list. It seems that although BT only present 6 digits (as standard) for CLI they expect the full number minus the leading 0 to set CLI. So if a number is 01234 987654 They will present 987654 and you need to present to them 1234 987654 Hmmm Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] UK PRI and outgoing CLI FYI
On Thu, Mar 29, 2007 at 10:40:50PM +0100, Julian Lyndon-Smith wrote: We only present the 6 digits ... and they give us 6 digits. For our outbound calls, for the the numbers 01702 1234[00-99] we have to present 1234[00-99]. BT isdn pri line. Weird, seems they're inconsistant or there's some oddity at the driver level? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] UK BT PRI
Has anyone got a working zaptel.conf and zapata.conf for a Digium Wildcard TE110P T1/E1 Card. It's connected to a BT ISDN PRI (EuroISDN) with 24 channels. Inbound works fine, but outbound isn't setting CLI (it seems the line supports 6 digit CLI). Inbound CLI works fine. In the dial-plan using Set(CALLERID(num)=123456) then Dial(Zap/g1/01234567||frT) Where 123456 is in the range of BT allocated numbers. Using Asterisk 1.4.1 and Zaptel 1.4.0 Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Queues
A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Queues
On Sat, Mar 17, 2007 at 11:44:52AM -0700, Steve Edwards wrote: Yes. to which bit? auto-agent (as per resource) or voicemail to an agent? Steve On Sat, 17 Mar 2007, Steve Kennedy wrote: A quick question on queues in Asterisk, if you specify a specific resource as a queue member (i.e. member = SIP/40 say) is it automatically a member of the queue without having to specifically log on via AgentLogin stuff? I under stand if you specify something like member = Agent/100 you then have to go through the login process (or AgentLoginCallback). If an Agent logs in, can a voice mailbox be assigned to an agent? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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 -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Re: Sending SMS
On Sat, Mar 03, 2007 at 12:01:58PM +, Gordon Henderson wrote: You're missing nothing; The telcos have us by the short curlys. For them, it's money for old rope. They probably (in the UK at least) make many times more money through TXT messages than voice. The base rate here is about 12p a message. 12p for 160 bytes, or a single data packet over their network - which would be over ?700 per MB. There are now bolt ons or additional packages depending on the network you're with - eg. with my contract I get up to 500 free TXTs a month. I know some people who send dozens a day here. (Especially young people - I think most 10 year olds now have mobile phones!). It's scandalous, but no-one challenged it when they first anounced it because we all thought it was fantastic! The best thing they ever did was for the 4 networks (in the UK) to agree to pass TXT messages between each other. That was some 6 or 7 years ago, maybe more, and that's when it really took off big time in the UK. The networks in the UK are regulated in terms of voice termination (Ofcom didn't like their high termination rates), so it's between 6p and 8p'ish per minute to get on to a mobile network. Currently mobile termination isn't regulated and to get a connection is a commercial agreement. The networks do have agreements with each other (mainly to stop foreign operators injecting cheap SMSs into the networks), there's a document AA19 which is the SMS interconnect agreement - however it's a GSMA thing and to get it, you need to join the GSMA. Most operators in the UK will no longer allow direct connection to their SMSC (unless you're expecting to generate very high millions of messages per whatever period) so you have to go through an aggregator. They tend to have agreements with multiple operators and will have agreeed a commercial rate with them (somewhere between 2.5p and 3p per message). They then will mark their rate up and offer that rate to customers. Ofcom have stated they are going to look at SMS termination rates and the operators are resisting. In the UK retail rates for both SMS and voice may well be below wholesale rates Which is a reason people use GSM gateways, which are still illegal for 3rd party use (i.e. an organisation can use a GSM gateway for their own traffic, but not carry anybody else's) - which means telcos cant use them. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] UK SIP Gateway
On Thu, Mar 01, 2007 at 12:17:49PM -, Chris Stenton wrote: I have used www.voiptalk.org for a number of years with their IAX2 connectivity and they seem very reliable with no echo issues. They will also change the CID to your number if you fax them proof of ownership. There's several VoIP players in the UK 1899 Gradwell.com to name a few. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Sending SMS from Asterisk
On Wed, Feb 14, 2007 at 03:42:23AM +0100, Patrick wrote: On Tue, 2007-02-13 at 18:31 -0700, Stephen Bosch wrote: Singer Wang wrote: by your .ca address I assume your in Canada.. both Telus and Rogers have a email-to-SMS gateway... Well, those are notoriously unreliable. I've had messages take hours to arrive when sent by the email-to-SMS gateway. I was kinda hoping for something more direct. Rogers prioritizes internal SMS messages over e-mailed ones. What I'd like is some kind of SMSC -- or something that accomplishes the same thing. Maybe http://www.kannel.org/ provides some useful info. Kannel is a pretty mature solution, it will drive a local GSM terminal or connect through to SMSC's using standard protocols (SMPP, CIMD, UCP/EMI etc) or even http/SOAP. Terminals such as Siemens TC/MC35, Wavecomm, Falcom etc seem to work well and Kannel tends to have driver modules for them, also many phones can also work. Make sure SIM buffering isn't used or you'll wear out the SIM (they have limited writes). Most operators wont allow direct connectivity unless you delivering 10's of millions of SMSs per month and you'll have to go through an aggregator. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] GSM Gateway promotion from ?69GBP
On Wed, Feb 14, 2007 at 09:41:32AM +0100, Dave Cotton wrote: On Wed, 2007-02-14 at 15:33 +0800, Sam Tam wrote: Hello All This month we would like to offer our GSM Gateway range for less to clear up some spaces. etc Perhaps, you could explain what is NON COMMERCIAL about your post. I would not buy anything from a spammer. Because Sam likes to do this about once per month. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Sending SMS from Asterisk
On Wed, Feb 14, 2007 at 10:29:20AM -0700, Stephen Bosch wrote: [snippage] If I understand correctly, this means I'll need an extra SIM just to send messages -- is that right? I build a Kannel server so that it can talk to a terminal that is on the network and can send messages. (It's an awful lot of extra hardware just for messaging capacity that will only be used by a few users, though.) What if I don't want to get my own terminal? Then you need to talk to someone who offers connectivity into the operators. Most operators wont allow direct connectivity unless you delivering 10's of millions of SMSs per month and you'll have to go through an aggregator. Can you show me an example of an aggregator? I don't know in the US? There are some ... they'll have an API and you then utilise that API to inject messages. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] SMS via VoIP and web
On Wed, Feb 14, 2007 at 07:17:32AM +0800, Ronald Wiplinger wrote: Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? Some telcos support sending SMS down phone lines, it's reasonably common in Europe and there's an ETSI spec for it. However it's probably easier to use something like Kannel which has an http interface and then either connect that to an SMSC or locally through a GSM terminal (phone). SMSC connections and pricing will vary depending on what country you're in. As a small customer (in the UK at least) it's unlikely you'd get an connection to an operator's SMSC and you'd have to go through an aggregator. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] error when compiling zaptel-1.4
On Thu, Feb 08, 2007 at 05:58:08PM +, younss azzayani wrote: when i compile zaptel make linux26 make install i got these errors: make[1]: Leaving directory `/usr/src/zaptel-1.4/wct4xxp' make -C datamods clean make[1]: Entering directory `/usr/src/zaptel-1.4/datamods' make -C /lib/modules/2.4.27-3-386/build SUBDIRS=/usr/src/zaptel-1.4/datamods clean make[2]: Entering directory `/usr/src/kernel-headers-2.4.27-3-386' make: *** arch/i386/boot: No such file or directory. Stop. make: Entering an unknown directorymake: Leaving an unknown directorymake[2]: *** [archclean] Error 2 make[2]: Leaving directory `/usr/src/kernel-headers-2.4.27-3-386' make[1]: *** [clean] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4/datamods' make: *** [clean] Error 2 any idea You've got kernel 2.4 headers so either you haven't got the right headers for the kernel or you really have a 2.4 kernel. In which case just do make instead of make linux26 Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] H.264 *Not Patented*
On Sat, Jan 27, 2007 at 10:20:40AM -0500, Matthew Rubenstein wrote: How does H.264 compare with GSM and G.729 in CPU demand (MIPS:Kbps) and audio quality at low bitrates? GSM is $free, but G.729 is higher quality (tho patented with at least $10 per running codec instance royalties). Will H.264 become the favorite high-quality Asterisk codec, or will it perhaps force G.729 to become free, or negligibly cheaper? G.729 is $10 from Digium. If you want to go license several thousand codecs (or probably more like 10's of thousands) I think the Sipro license is more like a couple of bucks. Unfortunately you have to license a large number in one go, so the initial set-up is very high. Digium have done a deal (I presume) whereby they've taken the intial hit and are just sub-licensing at a cost which make it whorth while for them. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Re: Is there a low cost cell phone base station for asterisk ?
On Wed, Jan 10, 2007 at 06:33:05PM -0500, M.Hockings wrote: That is more what I was thinking of but it is still a cell provider type of hardware. In my mind I was thinking of something very low powered and turning off the roaming, etc on the phone so they only work with the one base. Think single cell base-station transceiver that can talk to a cell phone and turn it into a sip conversation to Asterisk. Here in Canada, and back years ago, when I worked with radio I think the law was something like less than 100mw of input power didn't require a license. However, with the advent of cell phones that could very well not be the case in those bands. But one never knows... PicoCell have a reference design for a pico GSM basestation, but any country allowing cell phones will require licensing (even for low power). You'd have to pick frequencies not used by any network and that may be problematic. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Is there a low cost cell phone base station for asterisk ?
On Tue, Jan 09, 2007 at 05:11:55PM -0500, M.Hockings wrote: I don't really know the name of what I want to look for but maybe someone could tell me if it would be available. I have a number of old analogue cell phones laying about here and I was thinking it would be useful if I could set up a short range base station for them that would cover maybe an acre or so. What I would like to be able to do is use it to connect into Asterisk and this way have a useful wireless extension-phone range. Where are you. Generally you cant do that sort of thing as you don't have a license to operate in those frequencies. In the UK you definately don't (each cellphone has a license attached to it, it's just the operator pays the license fee). You cant get a license to operate a base station. Even if you could, running a basestation tends to need a hell of a lot of infrastructure behind it: - Basestation or BTS BSC (basesite controller) - generally can control up to about 100 BTSs. MSC (mobile switch centre) - like a telephony switch, connects BSCs and PSTN. HLR (home location register) - database of registered phones. Might need a VLR if allowing roaming. SMSC (short message service centre) handles SMS. Lots of glue ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: RE: [asterisk-users] WIFI SIP- The Best phone
On Tue, Jan 09, 2007 at 02:40:07PM -0800, mitcheloc wrote: Wait for the iPhone...seriously. I assume you mean Apple iPhone not Linksys iPhone ? It looks lovely, shame it's not available in UK until Q4. (also not FCC approved yet, but I assume that was deliberate as most phone leaks tend to come from filed FCC submissions). Steve p.s. also look at Truphone, they do WiFi/GSM/etc switching in client. -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Some queries on g729 license.
On Mon, Jan 08, 2007 at 10:51:03AM -0500, Al Bochter wrote: What about the free open source G729 There's no such thing ... g.729 (as per the ITU specification) is patent encumbered. Anyone USING the codec has to pay a license to the patent holders. Digium have negotiated a bulk-buying agreement and can sub-license (or relicense - however they've worded their agreement) the codec to end users. The same is true for several other codecs like AMR etc. even though there are open source implementations of them. MP3 is also patent encumbered, but since so many people were using it they changed the licensing so that freeware players could continue giving away the implementation. Any commercial software (or hardware) has to pay license fees (for encoding or decoding). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Some queries on g729 license.
On Mon, Jan 08, 2007 at 02:53:39PM -0500, Al Bochter wrote: So tell me what this FREE open source G729 is I am told that you can use these Codecs with your Asterisk ! [1]http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ You can do it Freely !! No, Ready Technology have packaged the codecs based on Intel's IPP code. The codecs link against Intel's IPP libraries. The code here is a diff and other material to compile the codecs once you've downloaded the IPP libraries. It will then produce a binary. To download Intel's libraries you need to agree to their licensing terms. To utilise the codecs you still need to pay a royalty fee to Sipro (as is clearly stated on the site). There are some pre-built binaries held on servers were the patents don't apply, however utilising those binaries on a system in a country where they do apply means you have to pay royalties. If you look it's the patches which are distributed under GPL, not the actual code itself. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Slightly updated UK English voice prompts
I believe there were some new prompts added for 1.4 for Directory Info. These have now been added to http://www.tel.net Have a good 2007. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] VoIP GSM Gateways
On Fri, Dec 01, 2006 at 06:24:46AM +0800, Sam Tam wrote: We do have @cough VoIP GSM Gateway for sell as well @ cough Try to search on ebay for gsm voip gateway and you will see some in there As far as I am concern it is cheaper than 2n. And if you are looking for multi ports then it will come off as RJ11 ports rather than voip and they are ?100 per port with a max of 16 ports in 1 chassis. Wrong list .. again ... Monthly ad ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Zaptel drivers for Solaris?
On Tue, Nov 28, 2006 at 08:30:55AM -0500, Frank Tarczynski wrote: I'm looking to build the zaptel drivers on a Solaris 10 X86 box. I've found the driver source code on https://svn.sunlabs.com/svn/solaris-asterisk but this source is posted along with Asterisk 1.2.7.1 Does anyone know of a fresher version? Is this code considered somewhat ready for prime time use? I thought it was for Solaris/Sparc anyway. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Asterisk and UK ISDN 30
On Fri, Nov 24, 2006 at 11:51:41PM -, Neil Tancock wrote: Anyone know if Asterisk will work with ISDN 30 and what sort of device I'll need to connect it? It will work with UK ISDN, but ensure it's EuroISDN and NOT UK ISDN (it's set in the telco switch and can generally be changed). UK ISDN is v85 and EuroISDN v110. ISDN2e (as in basic rate) is the Euro variety. Modern PRI lines should be Euro, but some telcos still provision the older UK variant. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V
On Thu, Nov 23, 2006 at 12:40:03AM -0800, Brad Templeton wrote: [snip] The USA uses 120v for house current. That's enough to hurt you and can kill you if you touch it wrong, though I've touched it a few times. A lot of the world uses 220. This causes enough of a spark that they require all receptacles to have a switch on them so you don't plug things in live. On the other hand, 220 can deliver twice the power in the same current. Kettles in the 220 world are _really_ fast. Your dryer and oven run on 220 even in the 110 world, only way to get enough power. Same with electric car chargers. The higher the voltage, the more chance your skin will find a conductive path across the body that's dangerous. You only need 9uA across the heart and it will stop - for good. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Hairping calls and Originating CLI
On Thu, Nov 23, 2006 at 08:55:41PM +, Tim Panton wrote: I've asked gradwell about my second point (still waiting...), but your thoughts are the same as mine. In theory it should be ok, because I have to authenticate the IAX connection with a username/password, which in turn they own and can look up if needed.. But I think theres something in UK law that says you can't be allowed to spoof the originating CLI. I don't know about a law, but the downstream interconnecting points probably make them sign contracts to that effect. Of course if you can prove to Gradwell (or whoever) that the number is yours, then it isn't spoofing - even if the call didn't really originate on that line. You can set your CLI to whatever number is within your number range. Several providers allow you to set it to whatever you like, but they generally have an agreement (that you sign up to) that says you'll only set it to numbers you own (or are within a number range allocated to you). Just because you can set your number to something, doesn't mean you're allowed to. This became very apparent when telcos used trombing to get cheap UK termination but you had to set your origination number to your real number, and then the trombing operator would be charged the UK termination rate, not the blended rate (which is an ITU regulation). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote: On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? 48V is also a sort of standard for telco devices if I remember it correctly... Power is nothing to do with voltage (well it is, but not alone), you need the current too i.e. V * A. Pylon electricity lines run at very high voltage (several hundred thousand volts) or the current going down the lines would heat the cables and you'd lose a lot of power. 48V is just a telco standard, and most telco equipment (that runs in racks) is 48V. Probably because 110 (or 220/240 here in EU) is enough to electrocute an engineer, and 5V/12V would require too many Amps so wiring would have to be huge to carry the current. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] A question on ISDN cards... (in the UK)
On Wed, Nov 15, 2006 at 11:15:16AM +, Gordon Henderson wrote: (I'm in the UK if that makes a difference) There seems to be a plethora of different ISDN cards available in both the BRI and PRI range - all with varying prices too - from ?25 to nearly ?1000 from some popular reseller sites... Does anyone have (or know of) a good comparison site, or have views on one card type over another? I'm assuming that the more expensive cards have additional features like better echo cancellation and audio processing abilities (or less CPU overhead?) Would anyone like to recommend a good and reasonable quality ISDN card for use in the UK, as after a lot of good results with TDM400P cards with several systems installed now, I need to look at a few ISDN BRI (old business highway about to move to ISDN2) and possibly a single-line PRI (ISDN-30) system. Remember that ISDN may not be ISDN (well it is), but you specifically need ISDN2e for BRI and make sure a PRI is configured as EuroISDN (ISDN v110, the UK default is v85). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] VOIP Bandwidth questions
On Thu, Nov 02, 2006 at 02:47:42PM -0500, Erick Perez wrote: This one will surely heat up. Usually the telcos have to calculate the subscribers vs telco capacity. I use simple figures, so extrapolate this to millions of customers, millions of lines, peak amount of calls at any given time of the day and of course houndreds,thousands of millions of dollars in equipment. Do a google on Erlangs ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] porting numbers in UK telewest/bt/adept
On Sun, Oct 29, 2006 at 03:09:42PM +, Conrad Wood wrote: On 29 Oct 2006, at 11:02, Matthew Thompson wrote: On 26 Oct 2006, at 11:59, Conrad Wood wrote: A client used to use BT isdn30 and ported the numbers to telewest several years ago. Now, the client moved to adept telecom. I *think* adept resells BT products. We got new numbers from adept (bt?) and the old pbx on the telewest lines forwards the calls to the new numbers. What is the old PBX and how are Telewest presenting? We had Telewest lines once and they were the same RJ-45 ISDN 30 as BT. Would it not be possible to use a dual port card and use Adept for the outgoing and Telewest for the incoming service? Ah - I forgot to mention that there are 2 offices involved. The client moved to new premises and the telewest lines are in the old office, Adept in the new office. Otherwise I would do as you suggest, yes. Though most OLO's have the ability and can port numbers away from BT, BT will not port numbers out of area i.e. if they are geographic numbers in BT terms they are tied to an exchange. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] TNT Max Password reset
On Wed, Oct 04, 2006 at 06:01:25PM -0400, Jay R. Ashworth wrote: On Wed, Oct 04, 2006 at 02:18:49PM -0600, Natambu Obleton wrote: Anyone have happen know how to reset the password on a TNT Max? Thanks. Does your asking here suggest that the the MAX's can do, say, voice gateway service? Protocols? Codecs? Ascent TNT's with the right software and hardware can do SIP, E1 termination/origination, and all sorts of codecs. Similar functionality to Cisco AS5200'ish. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] GSM Gateway Promotion from ?69GBP
On Mon, Sep 25, 2006 at 06:09:02PM -0400, Alex Robar wrote: This is a non-commercial discussion list, hence the name Asterisk Users Mailing List - Non-Commercial Discussion. Post this to the -biz group. He does this every month or so Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] DSL router with integrated SIP proxy?
On Thu, Sep 21, 2006 at 10:11:43PM +0100, Brian Candler wrote: Does anyone here know of an ADSL router with integrated SIP proxy? Netscreen 5GT ADSL, it has what's called an ALG (application layer gateway) and it does indeed support SIP. Full featured firewall etc too. Steve p.s Hi Brian :) -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [Asterisk-Users] UK Male English Voices
On Fri, Sep 22, 2006 at 02:56:39PM +0100, Will Tatam wrote: Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ [snip] The website appears to be down Yup, did an upgrade on Fri and something went wrong - will be fixed tomorrow. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] ChanIsAvail
I managed to work around my Dialplan. The ChanIsAvail application is great, except it only returns the 1st available channel. Could there be a ChansAreAvail which returns all the channels available instead of just the first. I'm sure it could be implemented as a macro or I guess a rewrite of the code. Anyone want a go? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Available channels
I'm trying to dial multiple SIP channels and check availability before I dial them. i.e. say I have an internal group that I define (extension 50) which actually dials SIP extensions 51 and 53 I'd use Dial(SIP/51SIP/53), but if a phone isn't registered (i.e. someone's unplugged 53) it does weird stuff (say coming in from PSTN). I'm using ChanIsAvail(SIP51SIP53) which works great, but only returns the 1st working channel, when what I need is something to return ALL working channels so it can dial them all (some extensions have 3 or 4 phones associated with them). They are all internal SIP extensions. I guess I could use Cut and check each available SIP extension passed into the macro I'm using, but that how do I cut a variable length string and parse each SIP/XX string? Any help appreciated. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Grandstream SX2000 attended tranfer
On Wed, Sep 20, 2006 at 04:12:34PM +0100, Faris Raouf wrote: magnus wrote: Hi all, could anyone share how to perform attended transfers with Asterisk and Grandstream SX2000's - we are able to perform blind transfers with no problem, but attended transfers fail - is it necessary to set two line identities on the phones to be able to do this? Appreciate all input, thanks - Magnus Funny you should ask -- I was going to ask the exact same question about the GXP-2000 (is that the model you mean or is there a new similar phone?). At any rate they both seem to have the same problem: In order to do an attended transfer on the Grandstreams we have to have two accounts defined on the phone (both on separate usernames/numbers in our case - maybe you can do it with one?), one on Line 1 and one on Line 2. [snip] Indeed, found it out (with Magnus) my accident. Defined both lines and it works. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] amr codec
On Sun, Sep 17, 2006 at 03:04:35PM -0700, Net Nut wrote: Well this would not be for comercial use.. I just want it for my own cell phone to talk on my own asterisk system. is that ok? Voiceage are quite agressive in terms of licensing. However as an individual it's probably not worth their efforts to do anything as the results wouldn't be worth it. If you run a business and the business has assets, then it's a different matter. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] CSR introduces UniVox reference platform
I'm not anything to do with them, but sounds a nice design. CSR have introduced a VoWiFi reference design that costs around $20. The interesting thing is that it supports both SIP and IAX2. Maybe Digium should make a WiFi handset ... Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] amr codec
On Fri, Sep 15, 2006 at 07:28:29PM -0700, Net Nut wrote: I have been searching, but I have not found the answer.. How might I add the amr codec to my asterisk server? I believe I found the amr source from http://www.3gpp.org/ftp/Specs/latest/Rel-6/26_series/26073-600.zip I compiled it but did not end up with any .so files like I thought I would need to put it into asterisk. Any pointers on how to get an amr codec into asterisk would be most helpful.. AMR is patent encumbered, just because the source is available doesn't mean you can use it without a license. Voiceage (at least) run licensing for AMR. It's about $1 per license (simulataneous encode or decode) with a minimum of something like $50K. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic
On Wed, Sep 13, 2006 at 12:33:01PM -0400, Steven Totaro wrote: Use SVN and not the tarball. Digium updated to 1.2.9.1 earlier this week. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Building Zaptel 1.2.9 with Octasic
On Wed, Sep 13, 2006 at 01:34:30PM -0400, Mark Hulber wrote: Yes, it worked. I didn't get the announcement of 1.2.9.1. Seems it wasn't announced, nor Asterisk 1.2.12.1 Nor their new Asterisk Appliance that seems to run off Flash (with a GUI that configures it all). ALso the new 4 port BRI card is on the site. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] updated zaptel tarball
When are Digium going to upload a corrected 1.2.9 zaptel tarball that compiles? I know it's correct in svn, but the public ftp servers still hold the incorrect version. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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] Cisco MWI
I have Asterisk 1.2.11 running and a Cisco 7960 (SIP v7.3). I cant seem to get the message waiting indicator working. I did try changing the MIME type as suggest, but then the phone kept continuously ringing. Any pointers? Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
Re: [asterisk-users] Cisco MWI
On Wed, Sep 06, 2006 at 03:36:53PM -0400, Doug Lytle wrote: Steve Kennedy wrote: Phone itself. [S-5200] This is incorrect. It should be: [5200] mailbox=5200 That bit seems to work, phones registers ok and can receive and make calls. You're missing the @context on your mailbox line. i.e. my phones are in the from-sip context, so: [EMAIL PROTECTED] But my mailbox (5200) is in default. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --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
