Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London
By UK standards that's a pretty good salary. Bear in mind that there is no real 1:1 parity in IT salaries. In the US we earn significantly more for our IT efforts than in the UK. To give you an example, when I moved from London to New York I got a 4 fold pay rise in real terms for doing exactly the same job. I was on 28K GBP over there and got paid 120K US$ over here. On 12/22/2010 02:27 PM, Danny Nicholas wrote: Wouldn’t that be 70K USD? Or should we REALLY be worried about the British economy? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Don Kelly *Sent:* Wednesday, December 22, 2010 12:24 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London 45K GBP would probably cover breakfast in South London. It’s about 70 USD. --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *C. Savinovich *Sent:* Wednesday, December 22, 2010 10:23 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote: **Job Description: Asterisk MySQL Support Engineer** Fast Growing Global Telecoms Company requires a very experienced engineer who has a variety of skill levels. The role would suit someone who has worked at switch level and fully understands how calls are to be handled to and from a VoIP platform, using a MySQL data base. Must be able to understand and had experience in dealing with, CLI, PDD, ACD issues arising from suppliers or customers. MySQL, Administration of Database, MySQL knowledge has to be at a very advanced level, stored procedures/triggers, replication and a strong knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for calling stored procedure from MySQL server) Must have experience in using either SIP Express Router or OPEN SER, as we will be deploying Kalamino throughout our Global network. You will need skills in configuration, installation and integration of various Asterisk applications like dial plans, IVR. Call recording, voicemail etc. and experience troubleshooting *One way voice-path, NAT issues, registration, etc. * Analytical thinking and ability to adapt quickly to fast changing requirements. *Required Skills Qualifications:* Candidate must have good knowledge of setting up SIP and IAX Trunks. Must have experience in installing and configuring SIP Express Router or OPEN SER. Installation and trouble shooting of Asterisk Servers using Centos. Installation and configuration PRI / E1s and Analogue cards mainly using Digium Cards. Good knowledge of Asterisk Dial Plans, maintaining and updating current dial plans using extension.conf as well as extensiosn.ael. Being able to write, maintain and update PHP pages linked to the MySQL data base would be useful. Scripting / Bash scripting would be useful. Expert knowledge in Configuring, Maintaining and querying MySQL. Expert level troubleshooting skills in inbound and outbound call flows. *Kind Regards Jess* *08451249555* **Jess Hart*** *__* **Langley James IT Recruitment*** 145-157 St John Street Clayton House Clerkenwell 59 Piccadilly London Manchester EC1V 4PY M1 2AQ 0845 124 9555 0845 225 5189 0207 788 6600 0161 660 7969 E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk Christian Savinovich Telecom Telephony Consulting 646.982.3572 c.savinov...@itntelecom.com -- _ -- 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 -- /\/\ark Phillips -- _ -- 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] Phones slow to ring
I would second that. If you don't set a dial string in your handset then it waits for N seconds before submitting the call. Pressing # will force an immediate dial. Mark On 11/04/2010 07:19 PM, Cary Fitch wrote: Watch the console as you dial. Dial the number and “#”. The ring should be “instant”. Or if not, look at the console and report what you see. Cary Fitch *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *jy *Sent:* Thursday, November 04, 2010 5:32 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- /\/\ark Phillips -- _ -- 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] [asterisk-biz] Asterisk system for church call center
They say confession is good for the soul. Perhaps they are offering a phone in confessional service? Unfortunately the business of the church often flies in the face of the business of the Church. On 03/29/2010 07:48 PM, Alex Balashov wrote: Sounds like the church has strayed from its core competencies and invited the money-changers into the temple. Being the official asterisk-biz harbinger of God's wrath, I suggest an intensely commercial platform, for the meek shall inherit the Earth, not the 700 Club. Fight the power. -- _ -- 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] Asterisk on a Beagleboard?
Probably worth discussing this over on the AstLinux list as they are all about embedded Asterisk running on machines like this. On 09/22/2009 09:48 AM, Danny Nicholas wrote: I was going to dismiss this, but it does offer an interesting possibility; Since it can boot Debian ARM from an SD card, you could have Asterisk-in-a-can where you would have the Debian build and Asterisk on the SD card and could hook up to a USB hub (for Ethernet connectivity) and process up to 14GB of call-data before having to offload to permanent/traditional media. If you really went nuts, you could possibly even power and use a DAHDI device off of some USB-powered peripheral. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Tuesday, September 22, 2009 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on a Beagleboard? Vincent wrote: Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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 128m of ram 256 m flash for the 'hard drive' is not much in either catagory. And ethernet is a USB addon, not on the board. ___ -- 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 ___ -- 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 ___ -- 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] [UK SPECIFIC] DAHDI and a OpenVox Card
I can really only speak to questions 2; The other 2 pins in your cable are for your second phone line - if you have one that is. You should be able to plug in a 4 wire cable into any socket in the house and get access to Both lines. On Sat, 2009-05-23 at 16:02 +0100, Dunc wrote: Hi everyone, I just found this thread, which is amazing as I'm on my first go with asterisk and so far I've been pulling my hair out for the last week :-) I have 2 questions which were raised while this fault was being debugged. 1) Gordon says:- Is this a place where you get a polarity reversal event on call startup? In the UK we do. (Well on BT lines - I've a funny feeling some Telewest/NTL lines use Bell signaling). On an incoming call we get: Polarity reversal. FSK Caller ID burst Ringing Well I've got an NTL phone line, can anyone tell me what to use for that? 2) Do I still need the same 2pin cable? Because I've been to Maplins too and bought one that I thought was right, but this one is a 4pin too. Can anyone tell me which pins on their 2pin cable are connected at each end? I'll bodge my cable until it works and then get a proper one once I'm sure. Thanks in advance. Dunc ___ -- 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 -- Mark Phillips, G7LTT/NI2O Randolph, NJ ___ -- 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] Alison Keenan (free British English voice)
Hi Folks, I have a few folks whom are interested in another recording session with Alison Keenan but don't have enough work to justify her visit to the studio. If there's anyone whom would like her to do some work but hasn't got around to it yet now might be the time. We need enough work to fill an hour of her time. So far we have about 25 minutes. Le me know off list. Thanks -- Mark Phillips, G7LTT/NI2O Randolph, NJ ___ -- 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] Follow Me app question
Hi all, When one uses the follow-me logic to forward calls to lots of phone devices do subsequent calls get routed to the VM (or whatever the 10x is)? In other words, if I want my office, house and cell phones to ring whenever a call comes in and I answer it on my cell, does the next call that comes in when I'm on my cell get sent to VM or does it ring the follow-me group again? -- Mark Phillips, G7LTT/NI2O Randolph, NJ ___ -- 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] Suing Dell||Dull Computers for CID abuse
Damn!!! Beat me to it ;-} As an Englishman now living in New Jersey (strangely nowhere near an exit) I have to say that the local idiom and accent leaves a significant amount to be desired. Terms like New Joisey, Shuwa ,wadder, badderies, congradulations etc make me wonder if I'm in an English speaking country at all. I've heard better English spoken in Nigeria. Mark On Tue, 2007-07-03 at 17:07 -0400, Andrew Kohlsmith wrote: On Tuesday 03 July 2007 7:20 am, J. Oquendo wrote: (again) Dell. We know based on someone's accent and lack of proper use of grammar, they are not speaking to us from a location in the USA. How can we validate that such instance is illegal. It You obviously have not been around any city centre in North America if you believe that to be true. :-) -A. ___ --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] Round Robin SIP peers?
Hi all, I have a cheapskate customer whom wants to leverage some cheap all-you-can-eat VoIP connections rather than pay for a per minute provider. On the inbound side I think I have a solution in that I can activate the call forward on busy option with his provider (some noname white label house) but how do I balance his outgoing minutes? Is there some way that I can set up a round robin where each outgoing call goes out over a different line? If not is there some way that I can create a pool of lines such that when 2 folks make a call they get separate lines? Thanks Mark ___ --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 to Cisco 2600 GW DTMF Not Working
Sounds to me like inband vs rfc2833 issues. I found that one has to use the same codec throughout in order to make DTMF function and then use inband. This in turn forces you down the road of alaw or ulaw codecs. On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote: Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to transmit frame type 1024, while native formats is 4 (read/write = 4/4) Jun 26 17:53:52 WARNING[14248]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024) Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge: Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call: Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (160)? Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received The call continues with this error until I hang up. I have been adjusting the dial-peer dtmf settings in the 2600 and have tried all the dtmf settings in Asterisk. Any guidance will be appreciated. Thanks. JR ___ --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] AstPligg
Great! Another one. With such a catchy name too! On Tue, 2007-06-26 at 01:42 +0200, lenz wrote: Hello list, AstPligg is a new Digg-like website devoted to * and VoIP news. At the moment, it's in beta stage and very basic - no fancy custom templates. It allows posting new stories, comments on stories, RSS feeds and tags. Still, it can be very useful, as the number of * sites and blogs grows every day, and keeping track of what is hot in the * world is increasingly difficult. Yes, I know, it's not much; but at least it's there and can be used immediately. You can find it at http://oinko.net/astpligg I'm looking forward to your comments (and stories) to make it a useful tool for the * community! l. ___ --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] High availability Asterisk
Hi folks, I'm experimenting with Heartbeat and whilst I have it running in an active/standby configuration I cannot get Asterisk to perform properly. I'm able to start the asterisk software (I imported the aterisk start file from /etc/init.d into /etc/ha.d/resource.d) with the heartbeat software but heartbeat continues to start more Asterisk instances. Any ideas? Mark ___ --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] Asterisk as an SCCP client
Hi all, Has anyone tried using Asterisk as an SCCP client? My company has just signed up a 2 year agreement with M5 (fools!!) but are having intellectual issues with things like intra office phone calls and voice mail etc. They suddenly realized after M5 was installed that ALL their calls go out to the Internet and back and they don't like it. M5 uses SCCP. Could an Asterisk box be configured to run as an SCCP client (or many clients) so as to emulate the M5 handsets? At least then we would be in control of our own calls and voice mail. Mark ___ --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] Nokia release
Nokia N95 available via ATT/Cingular for $795 with a 2 year contract. It was advertised in the New Jersey Star Ledger this morning. Mark On Thu, 2007-05-24 at 18:42 +0500, Rizwan Hisham wrote: Hi all, sorry to ask you something not related to asterisk, but i really want to know whether the Nokia N95 cell phone is released in the USA or not? if somebody from USA knows, plz reply. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone tried using a SCCP service provider
Hi all, Has anyone tried using an ITSP that utilizes SCCP as it's prime mover? Would it actually be possible? A customer of mine had M5 installed yesterday and they are already disliking the idea that their provider is in possession of all their VM's and that they have to go out to the Internet and back just to call the next desk. I thought that a reasonable compromise would be to install an Asterisk box and terminate the M5 service upon it. Ideas? Mark ___ --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 Inbound Problem
Without seeing your config files my guess would be that this is something to do with a bad codec negotiation. I'd bet that your IAX phone is using ulaw and your DID provider is using something else like G729. Mark On Mon, 2007-02-19 at 18:07 +0530, Arun Kumar wrote: HI I've configred an Incoming DID in my asterisk and when I call from outside I see call is coming to my Asterisk server and then from asterisk it rings on a particulat exten but when I pickup the call the call get disconnect immediate and on the other end it keep trying (ringing). here is my exten.conf: exten = _80.,1,Answer exten = _80.,2,Dial(IAX2/2001) did starts with 80 and any call comes for my number they are sending to my asterisk IP. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FS: Sangoma 10 port FXO card
Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --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] FS: Sangoma 10 port FXO card
Why not. Users by stuff too! On Fri, 2006-11-24 at 11:40 -0800, Anthony Rodgers wrote: Please don't cross post FS items to *-users - that's what *-biz is for. CP On Nov 24, 2006, at 10:45 AM, Mark Phillips wrote: Hi all, I have a surplus Sangoma 10 port FXO card for sale. This model could be upgraded to 12 ports or even changed to FXS or a combo of FXO/FXS by changing the grand-daughter cards (each card supports 2 lines). You could also downgrade the card by removing any or all of the daughter cards. I'm asking US$450 plus shipping to the lower 48. Paypal or Master/Visa only. Thanks Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FS: Sangoma A200 10 port FXO card
Hi folks, I have a Sangoma A200 10 port FXO card for sale. US$500 secures plus shipping. Thanks Mark ___ --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] IVR for the called part (IVR inside out)
I don't think that that there's any way around this. At some point you require human intervention. Perhaps the only way to do it would be to set up some sort of timer. After x seconds if you don't get a key press Asterisk moves the call to it's own VM? On Wed, 2006-10-04 at 07:00 -0200, Daniel Cyt wrote: Hello, I'm trying to get it to work but I can't find the right way. I would be glad if the list could point me the right directions. What I want: My Asterisk dialing out to a number (my mobile phone) and playing an IVR to the called part saying press one to accept this call. If the called part (my mobile) press 1 the call goes thru, otherwise it goes straight to asterisk voicemail. Reason (my scenario): I'm going to setup a follow me from my extension to my mobile phone and I don't want people to find out they are actually rining on my mobile. I don't have the option to disable voicemail feature on the mobile company. The problem happens if I don't pick the call or I'm, for instance inside a tunnel, where my mobile lose signal. Asterisk will think my mobile voicemail is somebody answering, and whoever called me will her the mobile voicemail. I've been searching for a while before emaiil the list but I could not find anything like it. Thank you very much _ MSN Messenger: instale grtis e converse com seus amigos. http://messenger.msn.com.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail maintenance questions
You don't need to restart Asterisk. Just do a reload app_voicemail.so On Wed, 2006-10-04 at 06:45 -0500, Jordan Novak wrote: How is the best way to add,clear mailboxes and change passwords for voicemail. I am guessing you need to remove the conf entries for the mailbox restart asterisk and then add them back in and restart asterisk. Is there a better way? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 failover when out of licenses
What do yo mean by fails? If you don't if one party doesn't have the preferred CODEC Asterisk will fall back to the next preferred CODEC and so on until a match is found. Can't help you on the licensing thing though. I guess no one wants to touch it since Digium's stance seems to be that you should have a license for each seat rather than a pool. Mark On Thu, 2006-09-07 at 11:04 -0400, Tod Detre (CampusEAI Consortium) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Is there a way to have asterisk failover to another codec when you're out of g729 licenses? I did some google searching and all I could find was this post from early 2005. http://lists.digium.com/pipermail/asterisk-dev/2005-February/009405.html Has three been any work done on this? In fact, I would actually prefer if it didn't failover just on availability of licenses. If it would just try another codec on the list if the first one fails. ___ --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] Capacity for transcode G711 to G729
What tools are you using for this? I'm sure you are aware of SIPp but wondered if you had anything else? Mark On Thu, 2006-09-07 at 21:41 +0200, Matt Riddell (IT) wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 RR wrote: Hi matt, sorry this might be a stupid question but is a bit pertinent to me, I'd asked something similar in one of my last email regarding SMP. Do you know if (*) is capable of making use of HT support i.e is multi-threaded and improves performance for operations like transcoding? Is that a valid question or is this only dependant on the I don't think you will get double or anything, in fact many people have suggested that HT be turned off when people experience problems. OS/Kernel, the CPU itself and the chipset on the motherboard? If I boot into an SMP kernel with Asterisk compiled with the SMP kernel source, would it just make use of multi-threading as the load increases on cpu-intensive operations? The best use I have seen is the newly converted IAX2 which can use multithreading in version 1.4, the beta of which should be released later this week. The best idea would be to compile Asterisk, run some tests (show translation recalc 60) with HT turned on, restart the box, bring it up with HT turned off and try again. You should also run a few calls and check the CPU. Also, when you said the normal is 120 simultaneous transcoding operations, what is normal? I have a P4 w/HT 3.4Ghz, 2GB RAM machine. Would that be above or below normal? Thanks much \R I would think that is above normal but not by much, I'm not sure what normal was, nor can I find the Digium document where this was stated. It wasn't that long ago. I'm doing some more tests on a 3000 line setup (external DS3s via Asterisk and SER clusters) at the moment which we are splitting to be half G.729 and half ulaw, and I will try to post some results. - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://wap.sineapps.com (Daily Asterisk News for your cellphone) http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFAHYFS6d5vy0jeVcRAluMAJ0du5Itu3Va1yAXu0+2gxMrC3JjLACePaTL fdZacwEIEm4Z63ht6E/KrAY= =DbHV -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Abstraction for a newbie
Sounds to me like you don't have a proper connection with Stanaphone. The only time you'll get these problems is when they cannot contact you to forward the call to your system. Double check you firewall settings. They need to be able to reach your system on port 5060UDP (assuming SIP) as well as ports 1-2UDP (Asterisk default media ports). They'll contact yo when a call comes in. You'll accept the call and at the same time tell them which port to send the incoming audio to. They'll also tell you where to send your outgoing audio. Hope that helps. Mark On Fri, 2006-08-11 at 15:45 -0700, Dominic Son wrote: Hi. Can someone explain to a right brained person what is going on with In/out bound trunks, how it connects to my Trixbox.. 1. i get issued a free NY phone number from a voip service like stanaphone . 2. i then call this number, it connects to the stanaphone voicemail 3. i turn off the voicemail because i want it to connect to my Askterisk, I've set up all the trunks in the PBX setup, ( sip.stanaphone, etc) 4. now i call my NY number, and it says 'this phone is not in service, please check the number and dial again' my Q: how does this work, more specifically, if i turned off the VM, how does stanaphone then know to look for my asterisk server to use the trixbox? -- Anything else, let me know. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] for you guys setting up customer offices...
SPA941's and 7960's On Thu, 2006-07-06 at 12:44 -0700, Shaun wrote: What brand/model phones are you using. ___ --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] can't dial Scotland ...
Perhaps the BT crew are all on a drunken rampage along Sochiehall
Street?
On Mon, 2006-07-03 at 15:14 +0100, Colin MacMillan wrote:
Hello,
For some reason I can't call Scotland from London ...
The details:
Asterisk v. 1.2.9.1
ISDN2 Interface - Junghanns card with BRIstuff 0.3.0-PRE-1q
Extensions.conf (context SIP-PHONES)
exten=_X.,1,Dial(Zap/g1/${EXTEN},60)
When I call this number - 01417778979 (this is a building company and
the number should work fine) - a woman's voice from BT announces -
'call cannot be completed as dialed, please check the number and try
again'.
I have only had this problem with calls to Glasgow, no other telephone
number is having a problem, local, national, or international.
Any help appreciated
Colin
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RE: [Asterisk-Users] Cell gateway for T-Mobile US??
T-Mobile do GSM, GPRS and EDGE and not GSM only as stated below. Devices connected to their network typicly use GSM but may use GPRS if a data plan is subscribed. Edge is available o those that have both an Edge device and a data plan. Not that I'm a T-Mo reseller or anything ;-} On Mon, 2006-06-12 at 09:24 -0400, Brian C. Fertig wrote: Typically yes, as long as you can get power for them compatible with ours. Tmobile is GSM. Well only GSM. They don't do anything else. You can check the WIKI I have found a few smaller ones that will probably work but don't remember what they are except that I found them there. _.._ Brian Fertig - dCAP, MSCE, CCNA, DCSE, RHCE Data/Telecom Engineer IT Administrator Planet Telecom, Inc Tampa, FL Office o: +1.813.864.3161x107 f: +1.813.881.9762 d: +1.813.864.3164 SIP URI: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BerkHolz, Steven Sent: Monday, June 12, 2006 9:03 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cell gateway for T-Mobile US?? Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the asterisk side. Thanks. Steven Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems TESCO Group Companies Fax. 248-836-5101 www.TESCOGroup.com Board member of www.glimasoutheast.org ___ --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 This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is Echo?
Actually this is an Elastic Impact. Throwing an object at another object as suggested below could cause the kinetic energy possessed by the ball to be diverted thus causing the ball to travel in a different direction after impact. This type of impact is commonly seen when insufficient kinetic energy is presented to a much larger object thus causing the larger object to dissipate the energy (usually as heat or sound) or in the case of most hard surfaces to return the incoming energy along the same path it arrived. On Tue, 2006-06-13 at 13:44 -0400, C F wrote: Echo is when you throw a basket ball on the floor and it bounces back, the effect of the ball coming back to you is called Echo. If you go into an empty big room and yell out I hate clinton you should hear the walls agreeing with you and thats called echo. ___ --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] Anyone know anything about VoiceWing?
Hi folks, Verizon hand just installed FIOS to the side of my house. Anyone know anything about their VoiceWing offering? Is is a SIP offering? Their technical staff can only tell me that it's a VoIP service. Mark ___ --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] Re: [asterisk-biz] UK Male English Voices
Yet another set? I get about 50 downloads a week for mine. Mark On Tue, 2006-06-06 at 22:27 +0100, Steve Kennedy wrote: I'd like to announce that the UK Male English Voices are now up on http://www.tel.net/ There's a complete set of base sounds and additional sounds (it should be complete compared to current Asterisk and Asterisk-Sounds-1.2.1). There's also a set with the word 'pound' replaced by 'hash' for both the base and additional sounds (only the actual replacements not a complete set). There's sets of gsm and pcm files. I'd like to thanks Jay Benham [EMAIL PROTECTED] who did all the hard work of recording them, and Jim Credland [EMAIL PROTECTED] for doing all the converting/sound work. Regards Steve ___ --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] Avaya 4624 Ip phone
Hi Gabriel, This phone does not have a SIP image available for it. It does use a modified version of H323 but you should be able to use it with Asterisk if you have something like Open323 installed. You'll need to install a TFTP server onto your network which the phone looks for to find it's configs. Mark On Tue, 2006-06-06 at 15:15 -0400, Gabriel Rosca wrote: Hi guys, I installed asterisk and it’s working really well. For now I`m using soft phones IAX and SIP but now I want to use the regular IP phone, what I have now is Avaya 4624 and I didn’t find a firmware for SIP for this particular phone I believe now is working with H.323, can please someone advice me if exist firmware for this phone to register SIP or IAX2 with my asterisk box, and to show me an example of config file for this phone. Thank you, Gabriel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Avaya 4624 Ip phone
Ig nore my last post. I had not seen this posting On Tue, 2006-06-06 at 22:33 +0200, Henk wrote: Have a look at the attached link. http://support.avaya.com/japple/css/japple?temp.documentID=283920temp.productID=107755temp.bucketID=108025PAGE=Document Henk __ From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gabriel Rosca Sent: dinsdag 6 juni 2006 21:15 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Avaya 4624 Ip phone Hi guys, I installed asterisk and it’s working really well. For now I`m using soft phones IAX and SIP but now I want to use the regular IP phone, what I have now is Avaya 4624 and I didn’t find a firmware for SIP for this particular phone I believe now is working with H.323, can please someone advice me if exist firmware for this phone to register SIP or IAX2 with my asterisk box, and to show me an example of config file for this phone. Thank you, Gabriel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone got a used T1 card I can have?
Hi folks, Anyone got a gently used working T1 card I can have? Can pay by CC, check, cheque or Paypal. Thanks Mark ___ --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] US telco lingo
Not quite. It refers to the contiguous 48 States that make up the US mainland. Alaska and Hawaii, whilst States, are separated by either another country or large amounts of ocean. Places like Key West whilst technically are over 50 miles from the mainland are considered part of the Lower 48/US48 Mark (a Brit whom lives in the US) On Wed, 2006-05-24 at 12:45 -0500, Don Pobanz wrote: Eric Bishop wrote: Could someone explain to a non-US dummy the following phrases I have What is US48? I assume by US48 they mean RJ48 which is a 8 conductor modular jack with signal from the phone company on 12 and signal to the phone company on 45. Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More Alison Keenan British English files
Hi folks, I've posted uLaw, aLaw, G729 and G723 variants of the Alison Keenan British English files. http://www.enicomms.com/cutglassivr/ Thanks -- Mark Phillips [EMAIL PROTECTED] ___ --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 on Proxy
Hi Paul, Asterisk often uses a proxy for its calls. What kind of proxy do you have? Also, If you have the server setup for nat=yes in the [general] area then ALL calls will get nat'd regardless of their locality. The best place to put this stement is in the relevant part of the sip.conf file that deals with the devices you want to have nat'd. If you only want to nat devices from your office LAN but not devices or service providers out on the Internet then you need to do a bit more configuration. I've pasted my config file below for your perusal. I have phone handsets on the LAN but my phone provider is on the Internet. I don't nat internally but do externally. ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 192.168.201.15 ; Address to bind to localnet = 192.168.201.0/24 ; Internal NETWORK address ;externhost = g7ltt.dyndns.org ; Address for NAT'd SIP messages ;externrefresh = 10 externip = 68.196.143.250 nat = no srvlookup = yes ; Enable DNS lookups context = from-sip-external dtmfmode = inband disallow = all allow = ulaw allow = g726 allow = gsm allow = ilbc tos = lowdelay canreinvite = no pedantic = no videosupport = yes callerid = 9738281625 qualify = yes realm=g7ltt.dyndns.org ; put external SIP provider registration here register = user:@sip.broadvoice.com:password:[EMAIL PROTECTED] [2201] ; Mark's MDA type=friend host=dynamic context=from-sip-internal username=2201 secret=blah dtmfmode=rfc2833 mailbox=2201 disallow=all allow=gsm [2202] ; WiFi cordless type=friend host=dynamic context=from-sip-internal username=2202 secret=blah dtmfmode=rfc2833 mailbox=2202 callgroup=1 pickupgroup=1 [2203] ; type=friend host=dynamic context=from-sip-internal username=2203 secret=blah dtmfmode=rfc2833 mailbox=2203 callgroup=1 pickupgroup=1 [sip.broadvoice.com] ; main outgoing provider user=phone username=9738281625 type=peer secret=password nat=yes insecure=very host=sip.broadvoice.com port=5060 fromuser=9738281625 fromdomain=sip.broadvoice.com dtmfmode=inband context=enicommunications canreinvite=no authname=9738281625 qualify=1000 disallow=all allow=ulaw allow=g726 allow=ilbc You'll notice that nat=no is set in my [general] area. That means that unless I say otherwise all devices are considered local and so no nat required. In the [sip.broadvoice.com] area I turn on the nat. I think that in your case you do the reverse of this. I'm on the end of a cable modem and so I *should* use the externhost settings as my number could change dynamicly but as I've found that it never does save myself the DNS lookup. Hope this helps. Mark On Mon, 2006-05-22 at 02:55 -0700, Paul David wrote: Good Day All I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings. But on external network with PROXY setting ASTERISK DID NOT WORK. My question are 1 Can ASTERISK work in a PROXY setting . 2 If it can work how can i implement it . Expecting your reply Thanks Paul __ Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
As I understand it, the device uses either GSM or VoIP to access the carrier? Which cell phone carrier supports GSM and VoIP in the EU? They've been punting this thing around the shows in the US for a couple of years now but none of the carriers support it. With GSM having such blanket coverage I don't see many carriers going this way. I can understand this working in Asia where coverage is only in the majorly populated areas and even then only outside. On Mon, 2006-05-22 at 16:53 +0800, Sam Tam wrote: Well I think we all need to look at something like this first. We will be one of the first people in Europe who will be selling this. If anyone is interested do drop me an email. Picture of the phone can be found here. http://cyber-telecom.net/wifi-gsm.jpg GSM / VoIP Over WiFi Dual-Mode Phone CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi dual-mode smart phone, in March 2006. With a tri-band GSM/GPRS (Class 10) radio and an IEEE 802.11b WLAN chipset, It enables end customers to enjoy broadband multimedia services at WLAN covered homes, offices, hot spots/zones as well as reliable GSM/GPRS service anytime anywhere. It shows an outstanding performance in power management, mobility management, security, mobile VoIP, and voice quality, no matter what kind of access points it connects, as the result of CYBER-TELECOM Wireless's advanced technologies solving the critical problems of VoIP Over WiFi. the phone has passed most of regulation certification programs and has done interoperability testing with over 40 VoIP service providers, system integrators, and infrastructure equipment vendors worldwide. the phone is an ideal device for fixed mobile convergence. Hardware Specification Intel PXA271 processor with embedded Linux 2.4 inch TFT touch screen, QVGA, 260k Colors Built-in speaker/microphone, 2.4mm stereo and headset 1.3M pixel CMOS camera USB slave Mini SD 1100 mAh Li-ion battery GSM Specification Frequency bands: 900/1800/1900 MHz GPRS Class 10 SMS, MMS, WAP applications FTA/CTA certification FCC/CE certification WLAN Specification IEEE 802.11b RF channels: US: 11, ETSI: 13, Japan: 14 High-gain internal antenna WEP 64/128 bits, WPA, 802.1x EAP PSK/LEAP/PEAP/TTLS/SIM Power saving modes Fast roaming between access points VoIP Specification SIP: IETF RFC 3261 Codec: G.711, G.729a/b, G.723 Acoustic echo cancellation Dynamic jitter buffer Voice activity detection Stun-based NAT traversal Input Methods Handwriting Recognition English Chinese Numeric characters Soft Keypads Qwerty Standard phone dialpad Symbol Power Management Features Standby time 100 Hours (GSM on, WLAN on) 200 Hours (GSM on, WLAN off) Talk time VoIP Over WiFi: 3.3 Hours GSM: 7.8 Hours MP3 play time 5.8 Hours (GSM on, WLAN on) 6.2 Hours (GSM on, WLAN off) Fixed Mobile Convergence Features Simultaneously activated GSM and WLAN air interfaces Handling simultaneously GSM and VoIP Over WiFi incoming calls SIP-based seamless handover between GSM/VoIP Over WiFi Automatic/manual switch for out-going calls between GSM and VoIP Over WiFi Automatic/manual switch for data applications using GPRS or WLAN Unified phone book for both GSM and VoIP Over WiFi. Unified GUI for applications (phone, E-mail, browser, QQ) Call Features Call hold Call waiting Call mute Call forward Call transfer 3-way conference Voice mail SMS over SIP Phone book - (1000 entries with photos) Incoming call prompt with picture View phonebook during call Enter sketch pad during call Adjust volume during call Auto-answer/flip answer Quick silence Turbo dial Manual/Auto/Earphone redial Call history (20 entries) Data Application Features POP3 E-mail client (SSL support) 100 full E-mails with attachments up to 200KB Document viewer for MS-Office and PDF files Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0 Instant messaging: QQ Multimedia Features Video format: MP4, 3GPP Audio format: MP3, WAV, MIDI, AMR Picture format: WBMP, BMP, JPEG, GIF Camcorder: QVGA, QCIF Media Player Audio: MP3 player Video: up to 30 frames/second QVGA MP4/3GPP PIM Features Calendar Schedule management Alarm clock Voice recorder World time Currency converter Anniversary Other Features English - Chinese dictionary Calculator World time Notepad Sketch pad File transfer Counter Timer ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP to IAX - forcing codec pass thru
Hi Peter, I don't see any codec allow=blah statements. If your end user has something like [gradwell] disallow=all allow=gsm Then you'll be forced to send them a GSM coded call. Why not force the codec at your end by only supporting one? If the customer then transcodes the call when it gets forwarded to his handset there's not much you can do about that but at least you'll have handed the call off in the best format you can source. Mark On Mon, 2006-05-22 at 09:57 +0100, Peter Gradwell wrote: hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP stream and maintained the same codec, so that all asterisk did was signalling conversion. sip.conf... --- [sip-router-1.gradwell.net] context=sip-inbound type=peer host=sip-router-1.gradwell.net [sip-router-2.gradwell.net] context=sip-inbound type=peer host=sip-router-2.gradwell.net --- iax.conf... [general] bandwidth=high disallow=lpc10 jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=lowdelay --- when a call comes in, we dial something like this, in our dial plan: -- Executing Goto(SIP/213.166.5.134-118f5310, sip-users|7770002|1) in new stack -- Goto (sip-users,7770002,1) -- Executing Dial(SIP/213.166.5.134-118f5310, IAX2/user:[EMAIL PROTECTED]/441376350002) in new stack -- Called user:[EMAIL PROTECTED]/441376350002 -- Call accepted by customerip (format alaw) -- Format for call is alaw -- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310 thanks peter ___ --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] WiFi VoIP Handsets..
Don't ya just love living in this technological backwater they call the USA? DECT technology was released almost 20 years ago. In most of the world it's been and gone. Anyone in the UK or Hong Kong remember Rabit and having to find a Green Dot Hot Spot by the train station or post office? When you got home the thing would mysteriously become part of your home phone system. On Sun, 2006-05-21 at 18:42 -0500, Eric ManxPower Wieling wrote: Are any of these FCC licensed for use in the USA. DECT in the USA is VERY new. I believe that DECT is approved for use here. Either that or Staples et al are selling loads of illegal multi handset DECT phones. Some with VoIP some without. Mark ___ --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: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
I wonder what they think VoIP is? Are they just port blocking? Could they be doing packet inspection? Do they think all UDP trafic is VoIP? On Mon, 2006-05-22 at 11:14 -0400, Julio Arruda wrote: From what I understand, T-Mobile UK just announced they would block VOIP earlier this month, but that is quite recent, and I don't recall seeing this 'globally' announced. ___ --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] I've broken voicemail
I know what he's done. He's installed my Alison Keenan wav files without converting them. Try downloading the sln files instead. BTW, G723 and G729 files going up tomorrow. Mark On Tue, 2006-05-23 at 00:49 +0200, Patrick wrote: On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote: [snip] May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100 Aren't Asterisk sound files supposed to use 8KHz? Did you perhaps forgot to transform the wav(s) you are using now to 8KHz versions? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] British English voice files are ready for download
Obviously a Radio 1 listener. 2) I was surprised to find that I didn't like the results. This is a purely personal thing, but I found Alison Keenan's delivery too redolent of a England that is gone. I instantly felt like a child again, being told slowly and clearly what to do. ___ --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] British English voice files are ready for download
Hi folks, With thanks to Alison Keenan (another Alison!) for the voice, Chris Bagnal for converting from 44k wav to sln and finally Terje Elde for debugging my HTML code, the British English files are now ready for download. They can be got from http://www.enicomms.com/cutglassivr/ Thanks and don't forget to practice safe IAX ;-} Mark -- Mark Phillips [EMAIL PROTECTED] ___ --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] test -please ignore
-- Mark Phillips [EMAIL PROTECTED] ___ --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] Broadvoice does it again
Hi folks, It seems that BV has messed it up yet again. I noted this weekend that any call going in or out had no incoming audio. All my other SIP providers seem to be OK. Is anyone else having this problem? Perhaps it's time to move on. What providers do you recommend that provide unlimited US/Canada and Western Europe? Thanks Mark ___ --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] British Voice talent records Asterisk prompts
Hi folks, I have British comedienne, Alison Keenan (another Alison!) coming in on Saturday afternoon to record the Asterisk prompts for me. Alison speaks with a posh boarding school accent. Finally we'll have a free British English female voice bank. As I have her in my studio (yeah right; it's a cupboard under the stairs) does anyone need anything doing? She's charging a Pound a word with a minimum of 20 quid. Interested parties should drop me an script email as well as relevant funds to my paypal account ([EMAIL PROTECTED]). Expect to see Alison's work up on g7ltt.com later next week in wav format. Have a good weekend all. Mark ___ --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: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ?
I think it is correct. Isn't that why they call it a Smart Jack? I've only ever seen a regular cat5 cable used from the Smart Jack to the device (router/PBX/CSU/DSU/whatever). I believe the point of the smart jack is, amongst other things, to allow for the use of readily available cables. I agree however that back-to back (PBX-PBX etc) you would need a cross-over cable. Mark On Sat, 2006-04-22 at 18:14 -0400, Steven Totaro wrote: The telco guys probably did something non-industry standard and reversed send and receive in the jack that they plugged the CAT5 into. Sure it works, sure it is easier, sure it is not the correct way of doing things. Thanks, Steve From: [EMAIL PROTECTED] on behalf of Lacy Moore - Aspendora Sent: Sat 4/22/2006 2:55 PM To: Paul Mahler; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: Pinouts for T1/E1 crossover cable WAS RE: [Asterisk-Users]whatcable to connect a legacy PBX to a TE410P ? att (formerly SBC, formerly Southwestern Bell, formerly ATT) just came out and installed my PRI. FYI, they used Cat 5e cable. No special T1 cabling that costs a fortune to buy somewhere, just plain old Cat 5e cable. Guess what guys? If they are using this as customers' sites, they are probably using it elsewhere. It's only as good as the weakest link, so you can go out and spend lots of money on T1 cable, or just use Cat 5e like the telco guys do. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 preventing Zap channels from disconnecting immediately after PSTN line hangs up (getting empty voicemails)
Likewise here. Using a 10 port FXO card and no problems detecting remote hangup. I'll grant you it can be a little slow sometimes however. On Mon, 2006-04-24 at 16:54 -0500, Rich Adamson wrote: Mike Garey wrote: As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing problems with asterisk not detecting when a channel has been disconnected? Thanks, Hasn't been a problem here with either the TDM400 or A200D cards (both are in use in same box). Just tested it again from an external pstn phone, calling into asterisk. When the pstn phone hangs up, asterisk recognized it and dropped the sip session that was handling the call (to a 7960). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone and Mac mini?
Just for shits and giggles, have you tried using a cross over cable? I'm not saying it's gonna work because everything I read says you're doing the right thing but it's worth a try. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dmitry Ivanov wrote: Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [asterisk-dev] RTP mixer in Asterisk
Erm ... isn't this what a conference does? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Leonardo (listas) wrote: I will implement a SIP application and I'm considering using Asterisk for mixing the media streams (audio). Does anybody know if Asterisk supports or contains a RTP mixer? If so, how to use it? Just to be a little more clearer: I will send to Asterisk more than one RTP stream and they must be mixed. The result must be a single stream to be forwarded to a SIP phone or to the PSTN. Thanks, Leonardo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LCDPROC cient for Asterisk
I think I've asked this before and think that Matt had said something about this. Is there an LCDproc client for Asterisk available and if so how can I get a copy please. Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Avaya IP Office 412
Do you have the right cable? You need a cross-over T1 cable and NOT a cross-over ethernet cable that people commonly try. This should satify the electrical requirements and turn the lights green. You're on your own with the rest. I do have a question however; why are you now speaking SIP to the IP Office? Did you not buy that extra server? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com zgor wrote: Hi! First at all, sorry for my bad english ... I m trying to connect an Avaya IP Office 412 to Asterisk using E1 I ve compiled/installed libpri - zaptel - asterisk correctly and now, im trying to get the link working. I think, first step is to have green light on the TE110P, isnt it? I setup zaptel.conf: span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 defaultzone=es So, i think: clock will be generated by Asterisk But after making ztcfg -vv , i see that all channels are correctly setup, but running zttool, always i have RED Alarm Any idea ? Thanks you very much ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How can I force Asterisk t not override my codec order?
Actually no. As far as I understand it, the receiving station gets to dictate the codec used. You call and offer up your list. He selects his preffered from your list and off you go. in your case you will always have gsm from 12 becasue 2 has a prefference for GSM. Try it back the other way. You should get an alaw connection Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Álvaro Palma wrote: I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no): softphone 1 - PCMA - Asterisk - GSM - softphone 2 But, strange for me, Asterisk forces both sides to GSM. I guess that this feature is done to avoid the problem of users setting always the more bandwith consuming codec against its administrator desires. However, is there a way to bypass this feature, so Asterisk set as codec order the same offered by the softphone? Something in sip.conf like: use_client_codec_order = yes (no by default)??? Thanks a lot. ___ --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] How do I install speex for asterisk?
Do you have allow=speex in your codecs list in either sip.conf or iax.conf? if not this this could be the reason. Also, Speex won't get selected if its not the primary codec on either side's call initiation. In other words you allow list should look like this disallow=all allow=speex allow=blah allow=blah When you make a SIP call you will be able to force the other side into speex if they suport it. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Elaborating a little more I checked for files suggested by Matthew Roth: If the build goes as planned, the /codecs directory will contain three speex-related files: - codec_speex.c - codec_speex.o - codec_speex.so Then ran the show modules command and now codec_speex shows as loaded by asterisk! But still cannot make or receive calls using speex. I am investigating with my VOIP provider.. Thanks to all of you. -Original Message- From: Jesus E Zepeda [mailto:[EMAIL PROTECTED] Sent: Friday, February 17, 2006 09:54 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? Mark: I did so, but that did not make asterisk to integrate speex. Do I have to tweak something in speex after installation? This is some of asterisk output when I try to use speex: -- Accepting AUTHENTICATED call from 192.168.2.32: requested format = speex, requested prefs = (), actual format = speex, host prefs = (speex|ilbc|gsm), priority = mine -- Executing Macro(IAX2/ext2-2, outbound|14802012944) in new stack -- Call accepted by 66.234.228.160 (format speex) -- Format for call is speex -- IAX2/66.234.228.160:4569-5 is circuit-busy -- Hungup 'IAX2/66.234.228.160:4569-5' Feb 17 09:20:42 WARNING[1811]: chan_iax2.c:1717 attempt_transmit: Max retries exceeded to host 66.234.228.166 on IAX2/66.234.228.166:4569-9 (type = 6, subclass= 1, ts=8, seqno=0) -- Hungup 'IAX2/66.234.228.166:4569-9' == No one is available to answer at this time (1:0/0/0) Feb 17 09:20:52 WARNING[2508]: pbx.c:2405 __ast_pbx_run: Timeout, but no rule 't' in context 'internal' -- Hungup 'IAX2/ext2-2' -- Registered IAX2 'ext1' (AUTHENTICATED) at 192.168.2.31:4569 Feb 17 09:30:42 NOTICE[1811]: chan_iax2.c:5673 update_registry: Restricting registration for peer 'ext1' to 60 seconds (requested 300) -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I install speex for asterisk? If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --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] How do I install speex for asterisk?
Did you rebuild asterisk after your speex install? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do I install speex for asterisk?
If you did a make install with speex then everythings where it should be. Just do a make; make clean with asterisk and all will be fine. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jesus E Zepeda wrote: Huuu! I never expected you had to recompile asterisk to add a codec. But if that is what it takes, we'll do it. I noticed that asterisk makes reference to some speex.c in the makefile file. In some of those references I saw the actual speex.c file in the paths specified. A couple of them missing by the way. That could be why speex was never taken by asterisk. Mike, does speex have to be copied to a specific directory, then compiled and installed before re-compiling and re-installing asterisk? I appreciate you took your time to reply. Regards, Jesus -Original Message- From: Mike Pollitt [mailto:[EMAIL PROTECTED] Sent: Thursday, February 16, 2006 15:22 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] How do I install speex for asterisk? You need to recompile Asterisk itself after installing Speex. Do a make clean, make, make install. I usually stop asterisk before that last step, by the way! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jesus E Zepeda Sent: Friday, 17 February 2006 5:58 AM To: Asterisk User List Subject: [Asterisk-Users] How do I install speex for asterisk? Hi, everybody: I enabled speex in my asterisk box (iax.conf), but no phone call went throug. At the asterisk console, I used the show modules command and it did not show the speex codec in the list. So, I downloaded the speex codec from speex.org, v1.0.5, compiled and installed in my asterisk machine. What I still don't know is: what do I need to do from the asterisk side to make it available? I just downloaded it to a directory, compiled and installed thinking that by doing a restart to asterisk it would some how know where to load it from. Any hints are appreciated Regards, Jesus E. Zepeda ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk native sounds now available!
Yes, it seems that I was somewhat in error. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: In my remember, when playback a file. The Asterisk will automatically choose the audio file with the lowest conversion cost. Not always looks the filename.gsm. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, February 09, 2006 5:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk native sounds now available! Yes you can copy them into the same directory as the current files. Kris recommends that you move your existing files for safety only. The mode (ULAW, GSM etc) is selected by Asterisk depending upon what mode the current caller is using. Have you noticed that you don't have to put a file extension on the end of a Playback instruction? This is because Asterisk looks for filename.mode when trying to play a file. In the event it can't find filename.mode it looks for filename.gsm. If the file it's playing is not encoded using the current mode it has to transcode the gsm file into whatever is required. This not only adds computing overhead to the call in progress but degrades the quality of the file as all such transactions are lossy. Understand? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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 ___ --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 native sounds now available!
Alex, I've been looking for someone whom speaks both with a Welsh accent and also the language. Ya'think you could persuade someone to speak Taff for us? As for my VM files, Kris is gonna send me the updated list and I'm gonna re-record them. I have a new Samson USB Condenser mic I'm dying to try out. Not a bad price at $79. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Alex Barnes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: 07 February 2006 19:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk native sounds now available! Kirs et al, I did this already. It's on my website. Your most welcome to use them Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Kristian Kielhofner wrote: SNIP P.S. - Do you have a full set of prompts, but with the Queen's English and a british accent? If so, send me the WAVs, I'll do all the work and even host them for you! Contact me off list. Cool. -- Kristian Kielhofner Hi Kris + Mark Sorry I don't think I can sent out the prompts as they were bought from a private company (http://www.westany.com/) £75 for a set I thought was quite reasonable for a commercial deployment. We did actually have Marks prompts for a while but at the time there were a few needed ones missing (bit of a strange mix of English bloke to American woman to welsh girl going on :P ). But the biggest draw to switch to Westany was very easy to get the custom welcome messages done, Welcome to BLAH you call might be recorded.. Thanks for the info though I will have a go at converting them this weekend. Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk native sounds now available!
Is a panoplie legal in Wales? I thought they did away with those at the same time as the Wooly Mountainside Brothels? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Wilson Pickett wrote: I've been looking for someone whom speaks both with a Welsh accent and also the language. Check this: http://isdnvoice.com he says he has access to a whole panoplie of Welsh speakers here: http://isdnvoice.com/services.htm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk native sounds now available!
Yes you can copy them into the same directory as the current files. Kris recommends that you move your existing files for safety only. The mode (ULAW, GSM etc) is selected by Asterisk depending upon what mode the current caller is using. Have you noticed that you don't have to put a file extension on the end of a Playback instruction? This is because Asterisk looks for filename.mode when trying to play a file. In the event it can't find filename.mode it looks for filename.gsm. If the file it's playing is not encoded using the current mode it has to transcode the gsm file into whatever is required. This not only adds computing overhead to the call in progress but degrades the quality of the file as all such transactions are lossy. Understand? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Adrian A wrote: If I understand this correctly, this sounds package is a subset of the Asterisk sounds package. Can I just copy the native sounds (eg. ulaw) in the existing sounds directory and Asterisk will automatically use them instead of the default gsm ones? How does Asterisk pick which one to play, does it know about the .ulaw extension? Doug, It looks like you have installed asterisk-sounds. asterisk-sounds is not included in the Asterisk Native Sounds Package. That is a separate collection of prompts arranged by John Todd and contributed to the community. I have already talked with him about that. Other people have brought this up too. Basically, I'll consider re-doing (and paying for) the sounds in asterisk-sounds based on the donations I receive for what is provided so far in the Native Asterisk Sounds package. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk native sounds now available!
Kirs et al, I did this already. It's on my website. Your most welcome to use them Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Kristian Kielhofner wrote: Alex Barnes wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: 06 February 2006 17:48 To: Discussion of AstLinux - Asterisk on Compact Flash; Asterisk- [EMAIL PROTECTED]; [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk native sounds now available! Hello everyone, As I promised at eTel last week, I have finished up work on my Asterisk Native Sounds project. Here's a little diddy from astlinux.org: Hi Kristian, This sounds like a great step forward. However since am from the UK we have to use a private set of prompts. The company that did them provided WAV format as well as GSM but I didn't really think about it and simply used the GSM pack provided as I assumed that was the recommended option. Could you give me a little detail on what the best format settings are so that I can convert my UK set into uber ulaw processor codec. Also if you have a nice linux script to take out some of the effort that would be fantastic but if not I am sure the sox man page will help me out. *I did try simply calling the .wav using Playback() but asterisk wasn't having any of it. Thanks in advance Alex Alex, Your WAVs are probably 16bit with a 44.1 (or 48kz) sampling rate. Asterisk can't resample (that's probably for the better). You need to resample them with sox. See my (basic) scripts here: http://mirror.astlinux.org/sounds/scripts/ Once you have your prompts in 8bit, 8khz wav, you can use the convert module here: http://redice.krisk.org To convert to anything you want. P.S. - Do you have a full set of prompts, but with the Queen's English and a british accent? If so, send me the WAVs, I'll do all the work and even host them for you! Contact me off list. Cool. -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911 and ISDN PRI
I dunno about your provider but I know that 2 of my 3 MCI PRI circuits have no 911 abilities. MCI tells me this is becasue I have no local dialing plan on them. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michael Collins wrote: 911 **should** work on a PRI. If you are getting a hangup and you don’t see a valid hangupcause, it might be best to get your carrier on the line and have them monitor the circuit while you dial 911. They might be able to tell you what the problem is. -MC *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Joe Pukepail *Sent:* Tuesday, February 07, 2006 10:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] 911 and ISDN PRI Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to FWD
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes
extensions.conf
; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion
[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup
Try this and see how it goes.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bayrouni wrote:
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD
iax.conf
[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr
register = xx:[EMAIL PROTECTED]
[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes
when I call the 613 number (echo FWD service), I have this
message from my PBX:
Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new stack
-- Called iaxfwd-outbound/613
Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest:
Auto-congesting call due to slow response
-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
== Everyone is busy/congested at this time (1:0/1/0)
Please, how can I resolve this problem?
Thank you very much
___
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Re: [Asterisk-Users] Asterisk native sounds now available!
Erm ... sorry. That should read Kris et al Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Mark Phillips wrote: Kirs et al, ___ --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] Better i18n for Asterisk?
The same 7 sound file is used to indicate both time and quantity. The sound file could be easily recorded to say sept heure but then every time the VM system tells a user that they have 7 messages they'll hear something like vous avez sept heure notification (excuse my schoolboy French). Perhaps rather than writing a VM AGI one could have a French language patch to the sources? In general I think the French way is better (I can't believe I just said that). I tell the time using the 24 hour clock. 7:45AM is correctly expressed at 7 hours 45 minutes using the 24 hour system. Could we have run into another Americanism here? OK, back to being English and bashing the French ;-} Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Jean-Michel Hiver wrote: Hi List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: message received at seven 30 am might sound good in English. But: message recu a sept trente apres-midi sounds terrible in French, because you *need* to say sept heure trente and not sept trente. Is there a way to fix this / improve the situation (other than write own voicemail AGI)? Cheers, Jean-Michel. ___ --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] change languages from an IVR
I've come across this in my dealings with my customers in Toronto. As an Englishman I find it most infuriating. French is after all, the most hated language in the world from an Englishmans perspective ;-} Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Derek Whitten wrote: Colin Anderson wrote: But, AFAIK, when they get to voicemail, the greeting is not based on the language setting, so you have to record it in those 3 languages, which makes a pretty long greeting This is common in Canada which has 2 official languages. The convention here is to intersperse the secondary language with the primary language so a non native English speaker can follow what is going on: Hi, no one can take your call right now / Bonjour, personne ne peuvent prendre votre appel en ce moment / Please leave a message and I will return your call as soon as possible / Veuillez laisser un message et je renverrai votre appel aussitôt que possible 3 might be a stretch though. ___ --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 maybe break the languages into smaller pieces? for french, press 1... for english, press 2... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] change languages from an IVR
Aha!! why didn't I think of that. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Gonzalo Servat wrote: On 2/6/06, Mark Phillips [EMAIL PROTECTED] wrote: A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish? Maybe once they've selected the language, set their default language? ie: exten = 1,1,Set(LANGUAGE()=en) exten = 1,2,... exten = 2,1,Set(LANGUAGE()=es) exten = 2,2,... exten = 3,1,Set(LANGUAGE()=fr) exten = 3,2,... Hope this helps. Cheers, Gonzalo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free IAX login
Try adding insecure=very to the guest user account in iax.conf. This should not do a user/pass challenge on the incoming call. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com kevin ling wrote: Not sure answer your question? Try to write some html code and let user register the username password online. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk guy Sent: Tuesday, February 07, 2006 7:31 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Free IAX login how to set up iax.conf , so IAX clients with any user name and any secret can login to * ? ( no authorize for login ) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk to FWD
I forgot to add that you must have an IAX acount with FWD. A regular SIP
account won't let you then use IAX. You have to register for it.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with
asterisk.
Mine (which works!) looks like this
iax.conf
register = user:[EMAIL PROTECTED]
[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes
extensions.conf
; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion
[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup
Try this and see how it goes.
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Bayrouni wrote:
Hello all,
Here is my problem,
I try to place a call to FWD (free world dialup) trough my asterisk PBX.
my config is as follow:
extensions.conf
[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD
iax.conf
[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr
register = xx:[EMAIL PROTECTED]
[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes
when I call the 613 number (echo FWD service), I have this
message from my PBX:
Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new
stack
-- Called iaxfwd-outbound/613
Feb 7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest:
Auto-congesting call due to slow response
-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
== Everyone is busy/congested at this time (1:0/1/0)
Please, how can I resolve this problem?
Thank you very much
___
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___
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Re: [Asterisk-Users] change languages from an IVR
Log live the Python crew!! Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Colin Anderson wrote: unfortunately the federal government in Canada mandates this and in Quebec if you don't do it, you can be charged with a criminal offense. French Canada farts in your general direction. -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 07, 2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] change languages from an IVR I've come across this in my dealings with my customers in Toronto. As an Englishman I find it most infuriating. French is after all, the most hated language in the world from an Englishmans perspective ;-} Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Derek Whitten wrote: Colin Anderson wrote: But, AFAIK, when they get to voicemail, the greeting is not based on the language setting, so you have to record it in those 3 languages, which makes a pretty long greeting This is common in Canada which has 2 official languages. The convention here is to intersperse the secondary language with the primary language so a non native English speaker can follow what is going on: Hi, no one can take your call right now / Bonjour, personne ne peuvent prendre votre appel en ce moment / Please leave a message and I will return your call as soon as possible / Veuillez laisser un message et je renverrai votre appel aussitôt que possible 3 might be a stretch though. ___ --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 maybe break the languages into smaller pieces? for french, press 1... for english, press 2... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change languages from an IVR
A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish? Thanks Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] OT O'Reilly Asterisk TFOT
Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool box. I can't always access the net whilst on site. If I get stuck doing something I can look it up in the book. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dave Cotton wrote: I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. ___ --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] OT O'Reilly Asterisk TFOT
It does indeed. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com James Ronald wrote: Does the printed version have an index? -- JR Whilst it can be downloaded I find that a paper copy is easier to read. I bought it for that reason alone. I also find it's a usefull addition to my tool box. I can't always access the net whilst on site. If I get stuck doing something I can look it up in the book. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dave Cotton wrote: I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH sourced from a sound card?
This looks like the solution. I'll let you know how I get on. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: Hello, MP Can I couple this to the sound card in the Asterisk server and then have MP it play into the MOH? If so how? Yes, it's possible. I've tried it last week. 1. Add the following into musiconhold.conf: [default] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein In /var/lib/asterisk/mohmp3 should be at least one mp3 file. 2. Create script file /usr/sbin/ast-playlinein and make it executable: #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw 3. Then you need to configure your mixer to turn on LINE-IN capturing. You may plug into line-in FM-tuner or external audio player. Don't forget to reload (should be enough) asterisk. -- Grigoriy Puzankin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MOH sourced from a sound card?
How does the customer maintain the message if I have to capture it every time he changes it? This is not the solution. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Peter Fern wrote: Using the classic MoH, use a custom moh player (see http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf) and sox with the alsa pseudo-filetype, and output to stdout with the correct bitrate and samples... see the sox manpage for instructions. Untested, but I think that should do the job for you... Mark Phillips wrote: I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server and then have it play into the MOH? If so how? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MOH sourced from a sound card?
I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server and then have it play into the MOH? If so how? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Wifi phone set-up
Throw it in the trash now. There's next to no support for these. No firmware upgrades. The are VERY SLOOW in responding to network calls too. All in all not a very astute purchase. I should know; I've had 5 of them. I use the UTStarcom F1000 currently. Much better but still not good. Mark Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ronald Wiplinger wrote: Nabeel Jafferali wrote: I got some troubles with my wifi phone. What phone is this? pulver phone ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dial out and message playback
An example of this would be Outcall Voice Mail? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Danish Samad wrote: Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of AGI or application? I have looked into the autodial out feature but I am thinking of a more flexible or optimal solution. Any help will be appreciated. Regards, Danish ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and NAT - best practices?
Most often the simple addition of nat=yes in the relevant sip.conf stanza is all that's required to make a remote SIP phone work from behind a firewall. for example [2201] user=blah secret=blah auth=blah allow=blah host=dynamic nat=yes I've been running 4 remote SIP phones across the internet from my families houses all over the world in this manner. The only issues I get are those of bandwidth availability or rather occasional lack of it. Hosted PBX's are no different. The hosting service should be providing a similar mechanism (although it might not be Asterisk based). Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Michaël Gaudette wrote: Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect the ports of the router as well. Or you can configure your SIP phone to use a STUN server. Please read in voip-info.org about SIP NAT, there are good suggestions. regards On 1/20/06, Michakl Gaudette [EMAIL PROTECTED] wrote: Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk box that needs to translate the SIP and RTP port? In other words: if my SIP phone is behind a Linksys router, do I need to configure the Router for any reason? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone interested in getting a basic training course together for the greater NYC area?
Contact me off list if interested. -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] video development
This is a great idea! You could have an IVR presented by a computer generated figure. You could play viewzak to folks on hold. Or how about the company promo reel when waiting for you turn in the call center queue? I'm loving this idea!! In a previous life I used to be a video editor for the BBC. If you want me to knock up some video stuff for you lemme know! Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Fran wrote: I would like to develop a video file player tool inside Asterisk. When calling to an extension answer and Play a video file (H264). With the applications PlayBack is not possible to give a video extension (only sound file extension). is it posible? How do u start in this development? With AGI scripts is not possible to send a video stream...(i tried to send images but with SIP channel doesnt work. I am testing with SEREyeBeam ) greetings and thanks in advance. Fran ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eid Mubarak
It has to be said that Eid is a funny and possibly suspect celebration though. As I understand it (from one of my Muslim underlings) 3 Mad Mulahs have to look for a particular phase of the moon. When they see this phase they declare the start of Eid. They apparently get 3 nights in which to look for this moon phase. I guess my question is what happens if its cloudy on all 3 nights? Another thing I thought about is this; If we could get the Faithfull whom are attending the Haaj this week to suddenly apply their brakes do you think they could stop the world from turning? Better yet if they all jumped into the air at once would the resultant landing knock us off off our regular orbit? Talk about death to Ifidels! They could do it in one fell swoop! I wonder if Al Quaeda has spent any research money on this? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Rusty Dekema wrote: On 1/10/06, *Carlos Alperin* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: And, as I said before, I'm not a religious man, but I don't like other people trying to be funny with somebody else traditions or believes. Personally, I like being funny about traditions and beliefs a whole lot better than I like being overly serious (or worse yet petty) about them! I have never heard of Eid Mubarak, so I thought it was kind of interesting to learn a little bit about it. But if I were to bring up a religious subject in a technology forum (or list), I would consider myself to have gotten off rather lightly if the worst response I got was simply a funny little play on words such as the got my goat comment. Now, if someone came on here and started a I believe in Religion X and you believe in Religion Y, so YOU ARE DAMNED TO HELL FOR ALL ETERNITY spiel or on the other hand started a DON'T YOU (so-and-so) IDIOTS KNOW THERE'S NO GOD spiel, then sure; that would be annoying and offensive. But nobody did that. I don't understand why people get so worked up about this kind of thing. To me, the idea of you being offended by the got my goat joke makes about the same amount of sense as the idea that someone would have been offended by Rehan's original post that started this thread. Although one person might not like to hear jokes about religion, other people might not like to hear about religion at all. I think that either way it's not worth getting particularly upset over. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non-PRI T1
Are they configured for inbound calls? If so how? Usually the telco sends the last 4 digits of the called phone number down the line. This means you'll need an exten=blah setup in the context that handles the T1. Hope that helps. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com David Sampson wrote: Hello – I have a non-PRI T1 setup and have been making outgoing calls for several months no problem. I have Zapata.conf setup for fxs_ks on these channels. How do I take incoming calls on these same channels? Do I need to change the signaling? Any help is appreciated. Thank you, Dave ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Debugging
They're not? They have no business in an open source world then ;-} Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Douglas Garstang wrote: Not everyone is a C programmer extraordinairre. -Original Message- *From:* Alyed Tzompa [mailto:[EMAIL PROTECTED] *Sent:* Thursday, January 05, 2006 11:59 AM *To:* Douglas Garstang; asterisk-users@lists.digium.com *Subject:* RE: [Asterisk-Users] Asterisk Debugging Then stop looking for easy solutions and get your hands dirty changing your c files Alyed Well, I want the output that the NoOp's generate. I want to be able to manually log lines to a file through some mechanism. I just wish I could do it without all the extra NoOp stuff at the front. I just tried using: mylogfile = verbose in logger.conf but all I got was the startup/shutdown asterisk messages. Besides, this isn't what I wan't. I don't want Asterisk internal generated log messages. I want my OWN log messages, that I specify. Doug -Original Message- *From:* Alyed Tzompa [mailto:[EMAIL PROTECTED] *Sent:* Thursday, January 05, 2006 11:18 AM *To:* asterisk-users@lists.digium.com *Subject:* re: [Asterisk-Users] Asterisk Debugging I don't find the console output ugly, maybe messy, but never ugly :P If u don't like those NoOp, just take them away from ur extensions.conf. BTW, to save the console output to a given file, just edit your logger.conf file. Say you only want the console output, then just add to your filename the verbose option . The file will be saved wherever is defined in the asterisk.conf (the default is /var/log/asterisk) after editing the file you'll need to do either an Asterisk restart or input CLI logger rotate at the Asterisk console. i.e. ;logger.conf [logfiles] mylogfile = verbose Alyed I'd like to have Asterisk log useful messages during operation. Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra Executing NoOp(SIP/pstn.voip.com-08a28bd0, crud at the front of each line. I'm not sure how to save console output anyway. Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] M0n0Wall traffic shaping rules
Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] name that vendor...
My apologies, Cory. I am mistaken. I was not trying to imply that Voipsupply.com supplies sucky equipment either. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Cory Andrews wrote: Mark - we have never sold this device...just FYI. The only not well known 4FXO device we sell is the ClipComm 4FXO gateway. The rest of the 4FXO devices we offer are from well established companies like Mediatrix and AudioCodes.I deal with the product management side of our business, and from the looks of this device I am not familiar with it at all. Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Mark Phillips wrote: Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap FXS/USB terminal SE-B2K, can it work with asterisk?
Take a look at voipsupply.com. They have a number of devices that allow a wireless phone to be connected to a * server. One of their units has a built in ATA and another is compatible with X-Pro/X-Lite. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dan Elder wrote: I've been searching for clever ways to add a wireless phone to our asterisk install, I could setup ATAs on each station, but I'm wondering if something like the SE-B2K (as seen at http://www.skype-phone.net/) can be configured to work w/asterisk something like SJPhone. Anyone ever played with any of these products? I've ordered the B2K, and have the SE-P1K, but I haven't been able to find any non skype info on these devices... the B2K looks like it'd be a great way to do this, could it work? The sales specs on various sites that sell these say it'll do SIP, but I haven't been able to figure out how. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] name that vendor...
Judicous application of my Staples Easy Button reveals this to be a no name special I Googled it and found the device badged under Ipeya, BossLAN and a whole host of others. Until recently it was on Voipsupply.com too. This is one of the devices that was recently discussed a being a sucky device. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5846258648 The seller refuses to tell me who the vendor is. Anyone know? -Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Allison on Free 411
Yes they do use Asterisk for some of their facilities. However, Alison is a contractor and so whomever pays her money gets her voice. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Joe Pukepail wrote: I heard on the radio about 1-800-FREE411 and tried it out, I was very suprised to hear allisons' voice for the digits. Not sure if they are using asterisk for the backend on this or not. Try it out its Free! http://www.snopes.com/inboxer/nothing/free411.asp (not afflicated with it in any way). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Allison on Free 411
I had heard the one about the Microsnot style strong arm tactic. Perhaps we should lay this at the door of the MythBusters. The really spooky thing is when one calls a company in the UK whom you have never had dealings with before and hear your own voice talk to you. This happened to me about a week ago. I guess that's what happenes when you GNU your voicemail files. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dean Collins wrote: Alison is a contractor and so whomever pays her money gets her voice. Apparently not according to some people on this list she was 'unavailable' for another voip project about 6 months ago. I don't remember/care about the details but that was the story. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, December 29, 2005 6:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Allison on Free 411 Yes they do use Asterisk for some of their facilities. However, Alison is a contractor and so whomever pays her money gets her voice. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Joe Pukepail wrote: I heard on the radio about 1-800-FREE411 and tried it out, I was very suprised to hear allisons' voice for the digits. Not sure if they are using asterisk for the backend on this or not. Try it out its Free! http://www.snopes.com/inboxer/nothing/free411.asp (not afflicated with it in any way). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iterfacing with a Mitel PBX
This would not be the chosen method. Also, you have connected an fxo device to an fxs device. This will produce the results you have encountered. Connecting 2 fxo's or 2 fxs's together would not produce anything as you have discovered. The prefered method would be via a cross-over type T1/E1 interface. A dial plan is then placed on each PBX so that all routes are known and as many features as possible are maintained. Both PBX's think that the other is the PSTN. You could continue with you TDM400 card but you'd need 2 dial plans. 1 from the Mitel to the * box and vice versa. This would require you to have an fxo to fxs connection from each box. this would limit you 2 a total of 2 calls in each direction. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Tom Conklin wrote: I am testing [EMAIL PROTECTED] V2.2 I want to interface with our PBX via a FXO card (TDM400P). I have one extension hooked up right now, and I can call into the Asterisk system from both a PBX connected phone, or through a DID number, but I can't dial from an IP phone out to our PBX system or out through a PSTN line (9 on the extension in the PBX gets an outside line). I can call other extensions that are set up within Asterisk. I have configured the Zap trunk, and set up an outbound route, but the best results I have gotten so far is 'connected' to dead air, or a fast busy tone. Are there good instructions posted somewhere for using a PBX extension? Thanks, Tom C ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broken sound Music on hold, , voice prompts good
sound is all broken? WTF is that meant to mean. Does it play or doesn't it? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Zeeshan wrote: Hi, When I call to my asterisk server, voice prompts play ok but when it goes to music on hold, sound is all broken. Why is that, is there some ports which Music on hold uses which are not configured properly, or there is some other reason. Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Callerid
Assuming its a SIP based device [110001] user=something allow=whatever callerid= lateef Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Code Lover wrote: Hi all, How i can change the CallerId format in plan id? for the example i can see the value of CALLERID variable like lateef 110001 I want to let asterisk do in plain id like lateef any idea? -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budge Tone 102
This is a stable, well used firmware version. It fixes a load of faults that have plagued users. You should be fine Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Tomislav Parcina wrote: I have Grandstream Budge Tone 102 with Software Version:Program-- 1.0.5.18Bootloader-- 1.0.0.21HTML-- 1.0.0.42VOC-- 1.0.0.7. I'm planning to upgrade it with Firmware 1.0.6.7. My question is, does anybody has any ishues with this firmware version? Should I put this or some older firmware? ___ --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] What hardware fits my needs?
4 T's for 100 users? That's almost a line each. Talk about overkill and expense! Whatever happened to the 3:1 rule of thumb? That would require 33 lines. Obviously this would be a little difficult to produce so 2 T's for 48 lines would be what I'd install. If you have 100 users and they are all on the phone at the same time you either have a slammin' HellDesk or you have a big discipline problem within the firm. I have many customers in the 100+ user range and with the exception of 1 whom runs a HellDesk they all have only 2 T's with no reports of congestion either in or out. Could it be you are over scaling things somewhat? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com [EMAIL PROTECTED] wrote: Jami, Providing a specific response to your question is rather difficult without a more meaningful list of parameters. 1. You say you have 20,000 distinct DIDs already. Are these provisioned through an existing telephony switch using multiple PRI lines (E1/T1)? Ideally you would need 4 PRI lines to support the average load of 100 users (4xE1=120 channels or 4xT1=96 channels). However in reality there will be usage peaks - thus you may need to consider designing a system that can cope with double or treble that many simultaneous users in order to handle peak loads. 2. Providing that many mailboxes and offering the functionality you describe is feasible using Asterisk. However you will undoubtedly need multiple servers - though again the number of servers and their specification is dependent on many additional factors. 3. What is the nature of the service. i.e. Is it mission critical and do you need to ensure high-availability? This will impact the architecture/hardware configuration you choose. Also do you plan to locate all of the lines/servers at a single site or do you want to have redundancy spread across multiple sites in the event of an outage within your Central Office? 4. How many messages of what maximum length do you anticipate each user being allowed to store? Again this will impact storage requirements. 5. The www.digium.com site lists the cards they offer for interfacing to E1/T1 PRI lines. As for server hardware - you will ideally want to use fast multi-processor servers for your service. Again - the exacting specification is difficult to suggest without knowing more about what you are seeking to achieve. 6. Asterisk is robust and powerful. However there is a learning curve spanning anything from many weeks to a few months depending on your available skills/resources. Setting up a production grade service on this scale will certainly require a deep understanding of both Linux/UNIX and Asterisk. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of S.Ammad Jami Sent: Thursday, December 22, 2005 8:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] What hardware fits my needs? Hello: I want setup an asterisk based VoiceMail Server(IVR). I have around 20K distinct users(DIDs) dialing to my system through telephones/mobiles. The users can dial to their mailboxes and listen/delete voicemails sent to them by others. The users can also recordsend voicemails to other users. I expect to have 100 simultaneous users to my system. Please suggest me the hardware configuration I need to have: the cards, peripherals, no. of extensions, hardware server etc. Thanks Jami __ Yahoo! for Good - Make a difference this year. http://brand.yahoo.com/cybergivingweek2005/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make Asterisk to generate and terminate calls
This is the how long is a piece of string question. It all depends on the hardware Asterisk sits on, the codecs in use, the dialtone provider (SIP vs IAX vs T1/E1) etc. Do a wiki search and you'll find some examples of what folks have found. As for originate on one and terminat on another; thats doable. Your phone device will have settings for an outbound proxy. Set this for the outbound Asterisk server. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Ravi Shankar wrote: Hi, I would like to connect two linux machines running asterisk and then originate SIP calls from one asterisk and terminate it on the other asterisk. Terminating the call is not a problem because I can give the call handle to say AGI application on the terminating asterisk. How do i originate a call from the asterisk ? Is this possible using AGI ? Any pointers in this regard would be of great help. This type of application can be used two simulate bulk calls and find out what is the maximum limit for the asterisk in terms of CPU utilization, memory, etc. before it can be deployed in production environment. thanks, Ravi ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budge Tone 102
I don't know if the registration bug has been fixed but I've not seen the registration problem for some time. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Bob Goddard wrote: On Friday 23 Dec 2005 08:03, Tomislav Parcina wrote: I have Grandstream Budge Tone 102 with Software Version:Program-- 1.0.5.18Bootloader-- 1.0.0.21HTML-- 1.0.0.42VOC-- 1.0.0.7. I'm planning to upgrade it with Firmware 1.0.6.7. My question is, does anybody has any ishues with this firmware version? Should I put this or some older firmware? There is a major registration bug with all of the BT releases which Grandstream are refusing to fix. We no longer have anything to do with them because of this. B ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does broadvoice modify caller ID name?
I'm not sure what you mean here but you do realise that BV don't allow you to set the caller ID. They fix it in there system. I guess it gets displayed s whatever the minimum wage keyboard operator enters it in as. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Paul wrote: The logs for my broadvoice BYOD always show caller ID name with capitalization like Abc Towing although every other provider I test against will show it as ABC Towing. They even do this to substituted values when their CNAM lookups seem to fail. They will have Old Town ME when every other provider and phone I have shows OLD TOWN ME. It looks to me like they are doing some type of auto capitalization in their systems. I need comments from some experts before pursuing this further with broadvoice. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Merry Xmas to everybody!
What's wrong with us that celebrate Kwanza? Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Dmitry Ivanov wrote: On Friday 23 December 2005 10:22, Mauro Zanin wrote: Hi everybody, no issues this time. Only stopped to say: Merry Christmas and Happy New Year. Yes, Merry Christmas, Happy New Year and Hanukkah :) Just received nice postcard from Digium :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone doing NAT through m0n0Wall?
Hi Folks, I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues with Asterisk's NAT abilities. Here's my setup, A bunch of hardphones (various types) littered around the house. SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about. Grandstream BT-101 at my dad's house via our cable modems. Until replacing the Linksys with the m0n0Wall everything was working fine and dandy. I have externip=g7ltt.dyndns.org set in my sip.conf file. Without it I could not make my dad's phone work. With the m0n0Wall in place and the externip setting set I can make no calls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work but the external ones don't. I've done some tracing with both firewalls and have noted the following; Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phone in use. In other words it offers up the external address for internal calls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server. m0n0Wall: externip set as above and the firewall drops the packets. externip not set and the * NAT doesn't work. I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all calls to use its public IP address when externip is set? Surely this doubles network traffic; one packet goes to the router. another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff? I had assumed that the addition of a nat=yes statement in the relevant phone stanza would turn on or off the NAT reqirement for that phone device but this doesn't seem to be the case. Any ideas would be greatly appreciated. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
