Re: [asterisk-users] International dialing with GPX-2000 and early dial
I have been down this path with Grandstream but they (for reasons I don't
understand) want to upgrade the firmware to have a dial plan.
So the best you can do is use early dial, for all fixed length numbers in
the * dial plan this works reasonably well. International numbers vary in
length so apart from trimming the digit time-out there not much you can
do.
The GXP 2000 is a great phone is it's a pity that they don't want to
develop e the phone to make it even better.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
Other phones have a defined dialplan, just like an ATA the GXP is the only
phone I've seen like that!
I had a sudden stroke of genius, I haven't tested it, but I'm sure it
would
work. Define a DISA with no password at extension 011, and define a
context
where international calls can be dialed without 011, IE:
exten = 011,1,DISA
[gs-intl]
exten = _xx.,1,Dial(ZAP/g0/011${EXTEN})
and then asterisk can handle the timeouts
On 11/20/06, Anthony Kepler [EMAIL PROTECTED] wrote:
We are on the same page.
If you happen to find a solution - or know of a way that other phones
address these issues, please let me know.
Andrew Joakimsen wrote:
Ok, I actually GOT a GXP-2000. It does not have a dialplan. You
cannnot dial without the handset off-hook. I do not seem to find a way
to use early dial for international calls in a practical way, not
being able to dial international calls is not acceptable. Having to
dial # or send for domestic calls isnt either, and neither is having
to wait 4 or 5 seconds for domestic calls to complete
Or am I missing something?
On 11/8/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Early dial is a feature on the phone that makes use of the 484
(Address
Incomplete) response.
This is desired for in-office, local (PSTN), and long distance
dialing.
I'm really hoping to find a best-of-both-worlds solution to
this.
Andrew Joakimsen wrote:
Does the GXP-2000 not have its own dialplan? Use that and
disable
early dial
On 11/3/06, *Anthony Kepler* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I am trying to allow users to place outgoing international
calls
from a
GPX-2000 with early dial enabled, connected to Asterisk
1.2.12.1 http://1.2.12.1
http://1.2.12.1
I have the following extension line:
exten = _011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
When I attempt to place a call to a number in, for instance,
Kenya, I
dial 011254...etc.
and I get this on the asterisk console:
Executing Dial(SIP/1001-081fb718, Zap/g1/0112) in new
stack
-- Called g1/0112
It is attempting to dial out as soon as it receives a single
digit to
represent the .
What I need is for it to wait a reasonable amount of time
for
additional
digits.
I have tried using set(TIMEOUT(digit)=5), and I see the
following
in the
asterisk console:
-- Executing Set(SIP/1001-081fb718, TIMEOUT(digit)=5)
in
new stack
-- Digit timeout set to 5
However, this is printed far less than 5 seconds before the
dial out
attempt.
I assume there must be something relatively obvious I'm
missing
here...
if anyone can shed some light on this, it would be greatly
appreciated.
Thank you,
- Anthony Kepler
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
mailto: [EMAIL PROTECTED] |
SIP/Email
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RE: [asterisk-users] Queues without music on hold ?
Record a ring tone file as the default Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Queue option r, like so: Exten = s,1,Queue(somequeue|r) Try 'show application queue' at the CLI Wes Baehr -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ex Vitorino Sent: Thursday, January 11, 2007 5:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queues without music on hold ? Hello List, This must be an easy one... I'd like to setup a queue without music on hold - just give the callers the traditional ringing tones. However, not setting the musiconhold parameter in queues.conf does not seem to do the trick: it defaults to default moh class which: a) Gets played if it exists b) Doesn't get played if it doesn't, but the caller still gets no ringing tones Any ideas ? Thanks in advance, and kind regards, -- Ex Vito ___ --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 realize chief - secretary (or Manager - Assistant) setup with Asterisk?
Hi Michael, in practice I think that the managers extension should default to the assistant who can screen the call or call forward it. Call Forward - always or Call Forward - no answer would give you the flexability required. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hello, we are running a Asterisk (1.2) installation with about 80 snom phones (300,320,360). Now have the demand for a special manager - assistant setup for a few extensions. Since Shared Line Appearance is not available in 1.2 I´m wondering how to realize this... What we need is that the manager can decide whether he wants to get calls or not. If not he must have the possibility to redirect all incoming calls to his secretary. The secretary itself answers all calls and decides if the call is important enough to disturb the manager. If so she/he transfers the call to the manager. So the secretary can filter the calls for the manager... The only way I can imagine so far is via a redirect by AstDB on the manager extension. The managers phone has two different lines - the official and a secret one only the secretary uses... Or are there any other solutions? Any hint will be appreciated ... Michael ___ --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] International dialplans for Asterisk?
The + sign is grammatic only it just means your international dialing prefix + the country code etc. So for dialing a number from Canada to the UK you would advertize the number as + 44 xx etc. In Canada we dial 011 for international calls so I would actually dial 01144 xxx etc. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Wow what a mess! I can imagine how much easier it would be if the world adopted a country/area/exchange scheme like in the US with known length. It must be complicated in Germany just within the country. At least in the US we know what the length should be so if we don't have that we know the number is in error. Doug On Fri, 22 Dec 2006, Anselm Martin Hoffmeister wrote: Am Freitag, den 22.12.2006, 00:53 -0500 schrieb Doug Crompton: Question... What is the purpose of the + before the number? Does anyone actually have to enter it? If so how would you do it? It is not used in the US but do I see it come in on SIP lines CID. I assume the CID ignores it in the number as I do not see it on the display. It is however stored in asterisk and when doing CID comparisions it can be a problem. The + is replaced by the telco you are connected to - by whatever the local prefix for international call is. In the US and Canada it will be 011, in most parts of the world 00, and there is Russia with its exotic 08 wait for beep 10... The + should work in GSM mobile networks and most SIP providers seem to accept it. For callerid, there seem to be several cases. One of my providers (the others manage better and always give 00492281234567 formatted numbers) gives CID as +491601234567 for calls from one German mobile network, 491637654321 from a second network and 02281234567 from landline, so my dialplan has to cope with that such that my endpoints show the proper number. This is done by the following logic: If number begins with +, strip it. If number begins with anything but 0, prepend 00. If number begins with 0049, replace by 0. Although in Germany you can dial 0049 (region) (number), readability is better when there is only the 0 (region) (number) on the display - especially as numbers tend to get long, and e.g. Grandstream BT-100 only have a 12-digit display. BTW the longest number I _think_ is planned in Germany is 9 digits after the area code for 2- and 3-digit area codes, 8 for 4-, and 7 for 5-digit areacodes. There is one exception though that I know of: One of our ministeries has usually 55- numbers (55 being their number, then four digits DDI), but their fax numbers are 8-digit. Thus resulting in total in 011-49-228-55-87654321 from US, 18 digits. If you can, leave room for long numbers. BR Anselm ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.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] Re: Newbie Questions - Grandstorm phones?
Yes thats the bottom line, its mostly the country code which can be 1-3 digits long. There is no rules based solution for this. Historicaly each country picked a number out of a hat except the US (which had to be number 1) because as we all know it's the centre of the universe. The former USSR had to go for 7 and Russia still kept this after it's break-up. All the other former USSR countries have settled on a 3 digit number but (as far a I know) can still be accessed by dialing 7. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Am Mittwoch, den 20.12.2006, 14:42 -0500 schrieb Doug Crompton: Anthony, Ok I understand. The 011 is unique though and I guess the problem is the length of the remaining digits. This could vary based on country?? and I suspect there is no unique rule that could be applied??? I have not studied this but is there any uniqness to the remaining digits? Doug There are no general rules for international number lengths. In certain countries, the numbering plan is very specific about how long a telephone number is - the US is the best example, where ANY phone number is area(3)+line(7). AFAIK Luxembourg and a few countries with a small number of telephones have rules as well. On the contrary, in Germany there are area codes between 2 digits (only a few, Hamburg, Berlin, Munich, Frankfurt) and 5 digits, and inside those cities numbering varies wildly. Old lines (registered pre-1960 or so) sometimes still have 3-digit numbers, especially in the countryside where there is no urge to assign new phone numbers. A friend of mine has the numbers 328 and 1653990 on the same ISDN line. And then, there are DIDs with varying number length. A company I worked for years ago had 9559-X where X might be 0 for central, two-digit 1X for department calling groups, [234]XX for individual phones and 9XXX for individual fax numbers. No rules there, bad luck. BR Anselm ___ --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 GXW-4108 8 port FXO
I would be very interested in getting an 8 port FXO myself. They are very new so I don't think there are any used ones out there yet. Does anybody out there in Canada stock them yet? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using similar devices (that is, FXO devices that connect to the Asterisk server via SIP over Ethernet). I am looking to connect at least 8 PSTN lines, and as many as 12 or 16 to Asterisk (Currently using Trixbox, but I'm also looking at either AsterixNow or just building from scratch on a bare linux box). Money is a major concern in my purchases, which is why I'm looking at the Grandstream (even used on ebay, I don't seem to be able to find 8-16 port FXO devices for less than the approx $50 per port the Grandstream will get me... plus it has a video input for a security camera which is just a plus to me as installing a web capable surveillance camera at the location is on my to do list). -chris www.mythtech.net ___ --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] Fast Busy
Sounds like you have a disconnect supervision problem. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada We currently have a pri coming into our asterisk system. Most of the time, the did numbers that we call into it work great. However, occationally, we get fast busies, but we noticed those busies were not due to anyone being on the line, etc... Any ideas what could cause this? Is this a congestion thing? Is there something I should add to the dial plan or configuration of the card to fix this? Thanks, Rob ___ --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] (no subject)
You might want to take a look at the new 4 port FXO from Grandstream I haven't had one yet to evaluate but assuming it works it is very price competative and off-loads all the analog (TDM) stuff from your PC Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I have been using the sangoma A200 with echo cancelation and I have been real happy. - Original Message - From: Todd- Asterisk [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 14, 2006 3:23 PM Subject: [asterisk-users] (no subject) Hello everyone! I'm planning on setting up a new system shortly and can't pick the right card... We will have 2 or 3 lines coming in and 7 extensions (GXP2k's). Should I just get 2 or 3 X100P cards? Or do I need the Sangoma A20200 or even the A20200D (Echo cancelation)... I was thinking I'd use a Dell 2.0 GHz machine as the server... If anyone has suggestions as to the benifits/problems of each card choice, I'd love to hear it. thanks Todd ___ --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] Is there any Asterisk controllable thermostat?
The Message Waiting Lamp (neon) on these phones requires a 90v signal which is generated and switched to the phone via a special station card on an analog PBX. This feature was developed mainly for Hotel and Motels but I doubt there are any manufacturers who would develop this functionality for any ATA's as this technology is very old. your best bet is to use the stuttered dial tone or buy (as a previous person has suggested) a cheapo Grandstream (you can re-spay them any colour) Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --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 Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.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] Attended Transfer
Thats quite correct however if you have a multi-line phone like Grandstream GXP 2000 or Aastra 480iyou can put the call on hold manually. As for using an ATA, You can program the ATA using the vertical service codes. In this case you can use a code to tell the (SIP) ATA that you want to do a transfer. You must make certain that the code you choose doesn't conflit with Asterisk's feature codes. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry, according with voip-info.org, attended transfer is While on conversation with another party, you dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while putting the other party on hold. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hangs up and you will be back to your original conversation. The callee is put on hold automatically Eric, attended transfer is only possible with an ATA?? On 12/5/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Henry.L.Coleman wrote: Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the * and transfer by #. But obviously one is limited to the vacant digits on the dial pad (DTMF). With an ATA you would use FLASH (aka RECALL) ___ --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 -- Arlen Nascimento ___ --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] Attended Transfer
Attended transfer is really four functions 1. Put the caller on Hold while you dial another number 2. Speak to the dialed number (announce the call) 3. Patch the call on hold to the other party using transfer button. 4. Disconnect (otherwise this would be a 3 party conference) How these functions work depend on what type of device the operator is using. SIP phones have this functionality ie a hold button, a transfer button and multi-line appearances. If you are using an ATA with an ordinary phone and standard dial-pad then you may be able to put a call on hold by using the * and transfer by #. But obviously one is limited to the vacant digits on the dial pad (DTMF). Note: If your analog (POTS) phone has a hold button this will not work as the hold button simply applies a resistive load to hold the loop in an off-hook status. Hope this helps... Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Dear List, I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable attended transfer feature. but i just can't do it work. I've already set atxfer = * (and many other combinations) and all extensions on extensions.conf have the t and T option. But when I'm going to test, it doesn't work. Is there any other file that i have to configure in order to make it work? I've already looked at google so many times and nothing Does anybody have an idea?? Regards -- Arlen Nascimento ___ --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] any possibility of Vonage Integration
This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM ___ --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
[Fwd: RE: [asterisk-users] any possibility of Vonage Integration]
I stand corrected! However you do get my point ... The bigger the company the worse it is. Having to deal with these guys is a nightmare. The company that brings me out in spots is Rogers Cable (24/7). They have this electronic air-head called Gertrude or something, (an android) who can't understand the word NO and has trouble with YES (actually like my ex-wife now that I think about it) but anyway, the point is that these companies spend millions of dollars on advertizing how much they care about you and your dog/cat/rabbit/beaver/etc. but won't spend an extra few bucks to have another person in the call center. My future policy is make a bogus call to the call center before you buy the companies product. TTFN Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada LOL.. Sorry, had to point this out: I think you meant Timbuktu... http://www.thesalmons.org/lynn/wh-timbuktu.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry.L.Coleman Sent: Tuesday, December 05, 2006 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] any possibility of Vonage Integration This 24/7 mantra that companies keep promoting to us is often just the ability to subject us to endless hours of their lame MOH while you wait for the one service specialist to answer the phone from Tinbuckto. My apologies if you live in Tinbukto. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You login to your vonage account on the web and set the bandwidth saver option. That is the most you can do with a locked ATA. Vijay Gandhi wrote: Thanks for all the feedback on the message, if i do the vonage integration using FXo card, is there any possibility of working on G729 or GSM codec, because linksys boxes by default use G711, which consumes hell lot of B/w. Regards Vijay Gandhi -Original Message- *From:* Al Bochter [mailto:[EMAIL PROTECTED] *Sent:* Tuesday, December 05, 2006 4:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] any possibility of Vonage Integration Brad Templeton, Thats a very good point. Best regards, Al Bochter Bochter Services http://www.BochterServices.com/?t=Email (VOIP PBX) 1-866-638-1254 (Voip PBX) Free World DialUp: 780-217 WebSite: http://www.freeworlddialup.com/ We have Toll Free DID's instock * * * NO MONTHLY FEE - LIMITED TIME ONLY * * * http://www.bochterservices.com/?t=TF(NM)did BUY Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security Paul wrote: Brad Templeton wrote: On Tue, Dec 05, 2006 at 11:36:12AM -0500, Al Bochter wrote: And if you get someone over at Vonage that knows that to do you can connect without the FXO It is like FWD you have to get the KEY from Vonage for this to work. And more to the point there are so many VoIP providers out there, most of them cheaper, who do not require you to use a locked ATA, and thus work great with Asterisk. I number will speak IAX or SIP at your desire. Don't be fooled by the flat rates of the locked-box providers. The real rates are so low these days most people pay less paying per minute than paying a Vonage style flat rate. In addition people report if you start making really heavy usage of your Vonage flat rate so that they are losing money on you, they notice and try to stop it. $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) Vonage has 24/7 support. When my DID is out I don't want to wait until Monday morning. ___ --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 Inbound (clean). Database: 0653-5, 12/05/2006 - 12/5/2006 3:56:28 PM -- -- ___ --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
RE: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14
Hi Scott, I have the following firmware 1.1.0.16 1.1.0.11 1.1.1.9 1.1.1.14 1.1.2.6 1.1.2.13 Some of these were not from the official website but they were all an improvement 1.1.2.13 is very stable apart from the 56 button ext, unit support. Let me know which ones you want and I can send them to you. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks for your help Claudemir, I look forward to the response. Seems odd that they don't post an archive of their old firmware versions on their website, or at least ones that are required to get to the latest release from whatever is in the field already. Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudemir F. Martins Sent: Saturday, December 02, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] upgrading grandstream GXP-2000 from 1.0.2.13 to1.1.1.14 Hi Scott, I have direct contact with a support person from Grandstream. I will ask him about that and tell you what did he say as soon as possible. Please just wait. Regards Claudemir On 11/30/06, Scott Keagy [EMAIL PROTECTED] wrote: So I've got phones with ancient firmware, and the release notes for 1.1.1.14 say read the previous release notes and first upgrade to 1.1.0.16 The 1.1.0.16 firmware is not available for download from the grandstream website (at least I haven't found it). Any pointers on where to get this intermediate image? I already tried googling to no avail (didn't help that I was using a link with 2000 ms latency). Plus, any overall pointers for making this upgrade process a success would be appreciated. Regards, Scott ___ --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] T1 PRI not announce this is long distance call, please add 1 for this call...
Using the PSTN in Toronto ie 416 NXX X all calls to 647 and 416
exchanges are local. 905 is an over-lapping area code, most excahnges are
local, however Whitby (905 430 ) is Long Distance while 416 428
(Ajax) is not. You can find out which ones are long distance (from the
CRTC web site) and modify your dial plan to add the 1 to the dialed number
or route the numbers to a DID with your friendly ITSP like Unlimitel for
termination.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
do something like this in your extensions.conf:
exten = _NXXNXX,1,Dial(ZAP/g0/1{$EXTEN})
exten = _222NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _223NXX,1,Dial(ZAP/g0/{$EXTEN})
exten = _224NXX,1,Dial(ZAP/g0/{$EXTEN})
Where 222, 223 and 224 are local area codes.
On 12/4/06, Isaac Xiao [EMAIL PROTECTED] wrote:
Can any one help? In Toronto, we can't identify if a number is long
distance call or not. If long distance call, we have to prefix with 1.
We
should hear a voice prompt as above to indicate that it is not a local
call.
However, we hear the normal ring back tone (indicating the phone had
been
connected, but actually not) when we call this long distance call
without
prefixing 1.
Here is the message shown in CLI.
Requested transfer capability: 0x00 - SPEECH
-- Called g0/9056671191
-- Zap/1-1 is proceeding passing it to SIP/9188-0e6a
-- PROGRESS with cause code 127 received
-- Zap/1-1 is making progress passing it to SIP/9188-0e6a
Thanks in advances.
Isaac
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Re: [asterisk-users] Hold calling channel and ask called channel beforeconnect???
Hi Nigel, If I understand your question correctly, you can accomplish what you need in Trixbox/FreePBX by having your calls answered by a queue. When the caller is in this queue, he will hear music on hold until the call is answered by an agent. When the agent answers the call a recorded message can be played ahead of actually connecting the caller. With this feature I can be notified that the call is originating from a certain channel or line. This functionality could probably be modified to report the CLI of the incoming call. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada you can find an example on the wiki here: http://www.voip-info.org/wiki/view/Asterisk+cmd+dial On 12/1/06, Nigel J. Terry [EMAIL PROTECTED] wrote: I posted this a week ago and have had no response. Can someone tell me if I am asking a stupid question, i.e. is the answer either obvious or impossible? Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel J. Terry Sent: Wednesday, November 22, 2006 10:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Hold calling channel and ask called channel beforeconnect??? I am a newbie. Just got my Asterisk working and I love it. I want to do the following, believe it should be possible, but can't work out how: When I get an incoming call, I want to answer and just send ringing to the calling channel. Then I want to call the destination channel, send a message asking if they will accept the call, get a response (1 or 2) and then either connect the parties (1) or send the calling channel to voicemail (2). Any ideas, thanks Nigel ___ --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] Grandstream GXP2000 -- What's the Catch?
I have deployed the Grandstream 2000 with very little hardware problems. Early firmware was petty rough but from 1.1.1.9 onwards is very robust. Frankly it represents the best bang for your buck. The only thing that I would like to see is a dial plan (which would speed up dialing). Most IP-phones don't have this anyway so it's not a big deal. The only other IP-Phone that I would consider is the Aastra 480i which is of a higher overall quality but the display is not as bright as the GXP 2000 and is difficult to view. PS they haven't ironed out all the bugs with the sidecar (56 button BLF/DSS) Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Doug, Just a note on this subject: I have a Snom 320 at home, and it's got a nice orange MWI that's pretty visible (especially if the apartment is dark). At the office I have a Polycom 501. It's got a great red light right at the top of the phone in the middle. It's very visible unless the phone isn't facing you at all. Alex On 11/15/06, Doug Crompton [EMAIL PROTECTED] wrote: Well I have a Grandstream 200 in a home application and so far I have been happy with it. My biggest complaint is that 99% of these IP phones are black!! One of the reasons I bought the 200 was because it has a bright red, see across the room, message waiting indicator. I have not seen that spec'ed on other phones. That doe not meant they don't have it, it is just not spec'd. I imagine the multiline LCD's have it on the screen, but you would not see that unless you specifically walked over and looked. I would be interested if any other phones have message waiting indicators as visible as the GS 200. Doug On Wed, 15 Nov 2006, Tom Vile wrote: They brake easy. Speaker phone is not very good. Overall sound not good compared to a Snom, Polycom or Cisco phone. Drop registrations with Asterisk randomly. Power supplies die. Had 4 out of 10 go bad within a year. LCD backlight died on 2 that I deployed. We only do the Snom 320 or 360's now and are just as easy to configure and have alot of great options as well. On 11/15/06, Jeronimo Romero [EMAIL PROTECTED] wrote: We are doing a medium sized office in NYC with 80 phones. The customer originally requested Polycom 601 phones. The COO also authorized us to purchase 2 Grandstream GXP2000 phones for the mail room. We find these phones much easier to configure and work with asterisk . They support BLF intercom right out of the box. They can also be centrally managed and provisioned. They also sound great and work in a very intuitive way. We don't have real life experience deploying this phone so I'm just going to ask: Is there a catch? Why the huge price difference? These phones seem to do everything a busy corporate office would need. Is there a big qualitative difference between this phone and Polycom501/601?? Is there a major problem with this phone not disclosed by the manufacturer or vendors. Some feedback from people who have deployed them would be great. Thanks In advance. JR ___ --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 -- Tom Vile Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.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 -- Alex Robar [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 ___ --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] Newbie Questions . . .
By the time you purchase PCI cards for you extensions (FSO ports)you would be better off purchasing SIP phones like Grandstream GXP 2000 this will give you a fully featured PBX IP phone for about the same cost or less than FSO ports. Asterisk will have no problem running 25 or more SIP phones Personally I would reduce the incoming analog lines to 4 (FXO) ports and add some DID lines. This way you will only have to buy one PCI board with 4 FXO ports Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php . Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck On 11/14/06, Jason Flatt [EMAIL PROTECTED] wrote: Hello all. My company currently has an older Executone PBX system that we are outgrowing. Rather than wait until the last minute to make a hasty decision, I thought it would be a good idea to do some research and compare options first. My expertise is in computers and networking, and telephony systems are mostly foreign to me. What we currently have are 5 incoming POTS lines and 25 stations and are wanting to add 1 or 2 more stations. I think we might have added at least one more incoming line, except that the phones we have only support 5 lines (so I'm told). Our PBX system has room for 5 more stations, then it's time to buy a new one. I'm assuming I need to add some hardware in order to make Asterisk work with our existing setup, but I'm not entirely sure what. Based on the reading I've done so far and my limited understanding, if we wanted to use it in place of our existing PBX system, I would need to get an analog interface card (several, actually), like Digium's TDM400P, like so: 2 - Wildcard TDM04B cards for FXO and 7 - Wildcard TDM40B cards for FXS -or- 1 - Wildcard TDM04B card for FXO and 1 - Wildcard TDM22B card for FXO FXS and 7 - Wildcard TDM40B cards for FXS I might as well use the top configuration for future expansion. If I am correct, that is 9 PCI cards in a PC. I don't know of any motherboard that supports that many cards, so either I'm wrong, or I'll need different cards, or I'll need to utilize 2 or more PCs in conjunction with each other. I haven't yet found any mention on the last two options, so I'm assuming I'm wrong and I need a little enlightenment. Thank you for any information that will help me better understand this. -- Jason Flatt Father of Six: http://www.flattfamily.com/ (Joseph, 13; Cramer, 11; Travis, 9; Angela; Harry, 5; and William, 12:04 am, 12-29-2005) Linux User: http://www.sourcemage.org/ Drupal Fanatic: http://drupal.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 -- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket * ___ --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 v IAX2
Hi Jon, Well Skype was one of the reasons I started my Asterisk based business. I first came across a VoIP demo about 12 years ago in a teleco carrier in Altanta GA. At that time the technology was very primitive (most people still had dial up lines). Anyway, to cut a long story short it wasn't until I many years later that I tried Skype, then I knew the technology had finally arrived and was good enough for business communications. Here in Canada, long distance is realitvely inexpensive so cheap calls are not very important Most of my clients are sold on the feature set in Asterisk and the ability to have extensions in multiple sites/offices without any line costs. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. That will be Skype then ;-) -- Jon Farmer Telford, Shropshire, UK ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Newbie Questions - Grandstorm phones?
I am at ver 1.1.1.9 and I will update to 1.1.1.14 pretty soon. Strangly enough I have just picked up an Aastra 480i looks real nice! Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I am agree with you. Do you use the latest version of firmware? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry.L.Coleman Sent: Wednesday, November 01, 2006 7:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Newbie Questions - Grandstorm phones? I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada After doing some research on the Internet and studying all the major IP phones, I have came to a conclusion that Grandstream GXP-2000 has the most features of all the phones for the least price of all. I don't know how they are managing to manufacture their product for such a cheap price, but they're doing it well for sure. Each and every other phone has something missing in it, but Grandstream GXP-2000 has every necessary thing in it. Even if they sell their product at 2x the price, it'll still be a fair price. So Grandstream GXP-2000 is the best phone to go with. I only wish if they could make its face look a litter more like Polycom, that would be better. Aastra 9133i is the second best option. Good price for the features they have. A lot of lines, PoE, dual ethernet etc. Looks very professional, same design as those of existing non-VoIP office phones, which people are used to look at as office phones. This is becasue Aastra once used to make phones for Nortel, so they have the same designs for their IP phones as well. It gives more professional image. The only drawback could be smaller LCD. They could make it a little bigger. I am testing it these days. Third best option is Linksys 942. They have two lines, you pay extra for the adapter and pay extra for other two lines. This all make them more than twice expensive than GXP-2000. But then they come at the same level with GXP-2000. Good thing is the big display. I am also testing this phone these days. Polycom are best looking, expensive, but configuration a little difficult, and don't have backlit LCDs? And also they have limited lines. Mostly no PoE. Snom are good, ok looking, expensive and limited lines, either no PoE or no backlit LCD. But very configurable. And an important advice: Don't buy a phone which doesn't have backlit and non-tiltable LCD, or you'll regret later. ___ --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] Re: Newbie Questions - Grandstorm phones?
Hi Andrew, I can highly recommend using the Granstream GXP 2000. Upgrade the firmware to ver. 1.1.1.14 and you won't have any problems. The 4 line buttons are not actual lines they are calls queued up on an extension so you can have as many incoming lines as you want. The first call comes in on line 1 second simulatanoius call on line 2 etc. The main features that make this a great deal is POE if you want it and dual ports (so you can plug a computer into the back of the phone, plug the phone into the LAN and away you go!) The 7 buttons down the side can be programmed as DSS/BLF, Speed dial buttons or just to show if an extension is registered which is very useful if you use softphones. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All that being said, any comments on the Grandstorm phones? I've ordered the GS-101 for my house, and I'm seeing the GXP-2000 is VERY inexpensive for a business solution. I see it has room for 4 lines with 7 programmable buttons. I assume I can put a few more lines on the programmable buttons (we have 6 lines at our main location). One last newbie question, I assume if I have an Asterisk PBX at 2 locations in different states, I'll be able to transfer a call that comes into location1 to a user at location2. Thanks again for the quick responses help. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Wednesday, November 01, 2006 5:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: Newbie Questions Ken If these are older comdials then they are just analog phones with extra signaling. The extra signaling could be on the main twisted pair (likely) or on the next twisted pair as data (9600 baud modem) like some of the nortels do. Always remember that it would cost the companies a ton to make every system totally closed That being said, the entry price for IP phones or ADSI phones can be much lower than you think. Find a good consultant in your area, get an ATA, a TDM card, and an Aastra/SNOM/Polycom/Granstream to play with. You can order the Aastra phones from your local electrical supply company (the place with a long counter and lots of electricians drinking coffee ordering their parts.). Andrew On 10/31/06, Ken Williams [EMAIL PROTECTED] wrote: I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject. That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection. That is: Comdial Phone - Comdial System - Asterisk PBX (FXO?) - Internet - Asterisk PBX (FXO?) - Comdial Phone I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available. Thanks for any help. ___ --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 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --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] Which IP phones have best voice quality, preferably under $150
I strongly recommend you upgarde to the latest firmware for the GXP 2000. I have been using them for almost a year now and while the early firmware was poor they are now very stable and working fine (from 1.1.1.9) onwards. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some don't work at all (probably defective). What good phone are out there which will work perfectly and will not be expensive. Should be $150 or maximum $200. -- 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] Re: Newbie Questions - Grandstorm phones?
I came to the same conclusion. There is one thing however that the GXP2000 needs in my opinion. There is no dial plan avaiable in the configuration, this means that when dialing a number there is a slight delay before it actually dials. With a dial plan the dialed number is sent immeadiately the pattern is match ed so it saves a second or two. Maybe they will fix this? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada After doing some research on the Internet and studying all the major IP phones, I have came to a conclusion that Grandstream GXP-2000 has the most features of all the phones for the least price of all. I don't know how they are managing to manufacture their product for such a cheap price, but they're doing it well for sure. Each and every other phone has something missing in it, but Grandstream GXP-2000 has every necessary thing in it. Even if they sell their product at 2x the price, it'll still be a fair price. So Grandstream GXP-2000 is the best phone to go with. I only wish if they could make its face look a litter more like Polycom, that would be better. Aastra 9133i is the second best option. Good price for the features they have. A lot of lines, PoE, dual ethernet etc. Looks very professional, same design as those of existing non-VoIP office phones, which people are used to look at as office phones. This is becasue Aastra once used to make phones for Nortel, so they have the same designs for their IP phones as well. It gives more professional image. The only drawback could be smaller LCD. They could make it a little bigger. I am testing it these days. Third best option is Linksys 942. They have two lines, you pay extra for the adapter and pay extra for other two lines. This all make them more than twice expensive than GXP-2000. But then they come at the same level with GXP-2000. Good thing is the big display. I am also testing this phone these days. Polycom are best looking, expensive, but configuration a little difficult, and don't have backlit LCDs? And also they have limited lines. Mostly no PoE. Snom are good, ok looking, expensive and limited lines, either no PoE or no backlit LCD. But very configurable. And an important advice: Don't buy a phone which doesn't have backlit and non-tiltable LCD, or you'll regret later. ___ --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] Fixing the Caller-ID Problem, by John Todd for O'ReillyNet
Obviously we (as an industry) have to start to take notice of this spoofing. otherwise big brother will start to legistrate against it. This will give the CRTC or FCC another excuse to spend a lot of tax payers money on something which is of marginal value. My position is that there are only two reasons for wanting to change an outbound CID: 1. to deceive the called party 2. to validate the calling party I don't know how much notice people take of CID but obviously if if it can be used to mis-represent or for fraudulant purposes then it will become useless. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada I have a couple of useful bits that could be tacked on to this.. 1. Telcos required to offer the ability to set the outbound caller id. 2. Telcos required to offer the ability to write to the CNAM database, in near-real or short time. 3. Telcos required to forward the ANI you provide to the 911 wire center, instead of the trunk number of a PRI. -Ejay -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Tuesday, October 24, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fixing the Caller-ID Problem,by John Todd for O'ReillyNet This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models predicated on the ability of Caller ID to be transmitted to the PSTN with a value that is not expected; call centers are an obvious example, where customer-support staff make outbound calls with a Caller ID that may be from one of many possible clients. More troubling is the possibility that Caller ID may be used to trick unsuspecting call recipients into certain actions or beliefs, and it is this concern that's currently creating a legislative threat I believe must be averted. ... Congress is currently considering legislation titled The Truth in Caller ID Act, which certainly sounds noble. Who doesn't want correct Caller ID when receiving a call? The truth is that this bill is redundant--the Wire Fraud Act already covers this issue, and adding more wording seems to be merely a re-statement of a certain circumstance or type of Wire Fraud. While the wording of this legislation does not effectively change the amount of power a prosecutor currently has, I believe it will certainly create confusion and fear in the technical and investment community because of the uncertainty it promotes. It's like saying, I want you to not break the speeding laws AND I want you to not go over the speed limit! A legal staff could spend a week--at $200 an hour--explaining that to a CEO, despite the consistency. === http://www.oreillynet.com/pub/a/etel/2006/10/18/solving-the-caller-id-proble m.html Cheers, -- jra -- Jay R. Ashworth [EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] SIP v IAX2
As I understand it the main advantege IAX has over SIP is the number of port it uses and therefore its ability to traverse router/switches and firewalls Also the higher number of simulatanious SIP calls travelling through these devices adds a higher overhead than IAX with it's single port. Personally I like IAX but I there simply isnt enough hardware out there to use it exclusively. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Thursday, October 26, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote: with SIP qualify, I can specify, what time in delay I will accept, with sip and setting qualify=3000 I can circumvent this anoying messages (bacause delay in reply is about 2000ms, and I accept 3000ms) with iax, qualify is working different, so setting qualify=3000 will ping peer every 3s, quite inconsistent, imho So are you saying that in your world two different things, created by totally different people, must have the same configuration settings. - You will find DUNDi configuration a lot easier with IAX, although you can use SIP. - If you use SIP to route calls between Asterisk boxes, you will lose your caller id as SIP uses the From: number to authenitcate with. You will have to store the original caller id in an extra SIP header, and then pluck it out an the other end, if you want to preserve caller id. Yuck. IAX doesn't have this problem. 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
RE: [asterisk-users] SIP v IAX2
I suspect that IAX has less overhead but when we get into voice bandwidth then the answer gets very complex for any given codec. Andrew mentions SIP concurrency but I doubt that this buys very much. In reality, in a single processor world everything gets processes serially. For *2* IAX would be my choice every time, but for IP phones 95% are SIP so it's academic as to which protocol is best (as you have no choice). The only way I see of changing this is to manufacture IPphones with both protocols in firmware but that would only be possible if the demand for IAX devices increased. Its a bit like the VHS vs Beta war, both systems have their good and bad points In the end, sales/marketing perception will always win regardless of better technologies. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada On Thu, 2006-10-26 at 13:14 -0400, Henry.L.Coleman wrote: As I understand it the main advantege IAX has over SIP is the number of port it uses and therefore its ability to traverse router/switches and firewalls Also the higher number of simulatanious SIP calls travelling through these devices adds a higher overhead than IAX with it's single port. Personally I like IAX but I there simply isnt enough hardware out there to use it exclusively. What about the bandwidth used for both protocols? Is IAX using less or more bandwidth than SIP? Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Thursday, October 26, 2006 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP v IAX2 On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote: with SIP qualify, I can specify, what time in delay I will accept, with sip and setting qualify=3000 I can circumvent this anoying messages (bacause delay in reply is about 2000ms, and I accept 3000ms) with iax, qualify is working different, so setting qualify=3000 will ping peer every 3s, quite inconsistent, imho So are you saying that in your world two different things, created by totally different people, must have the same configuration settings. - You will find DUNDi configuration a lot easier with IAX, although you can use SIP. - If you use SIP to route calls between Asterisk boxes, you will lose your caller id as SIP uses the From: number to authenitcate with. You will have to store the original caller id in an extra SIP header, and then pluck it out an the other end, if you want to preserve caller id. Yuck. IAX doesn't have this problem. 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 -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --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] Linksys SPA3102 - PSTN hangup detection in the UK
You are welcome. Please let me know if this makes any difference. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Yep, just swop the two wires. Sometimes the Tip and Ring get reversed and most loop start interfaces don't really care (they work either way). It's worth a try since if the disconnect is a reverse polarity flash then the card may see not see this condition as it is already reversed. I have a similar problem with Foriegn Exchange line (FX) but I haven't had time to visit the client to check this out yet. Thanks Henry. I'll definitely give this a go. Faris. ___ --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] Add second account to Xlite 3.0
I believe you have to buy the non-freeware version to have this enabled. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote: I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on initiate time, but I can add second sip proxy account, which is very critical to my testing. I installed Xlite 3.0, which I could not add second account on SIP account settings. After I add the first one, the Add button is grayed out, I can not do anything to add extra account. Does anyone know how to get second account added in? http://support.counterpath.com/viewtopic.php?t=7919 And see also http://www.xten.com/index.php?menu=Productssmenu=compare (scroll down to where it says Multiple Accounts) ___ --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] Linksys SPA3102 - PSTN hangup detection in the UK
Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada You have polarity reversal detection and I do not (I did try with it on, but it didn't help even though there I have measured a polarity reversal on disconnect) FWIW: I once had a nasty DSL filter that broke polarity reversal detection. You have 3ms On hook speed, I have less than 5ms. You have Line In Use Voltage 30 and I have 25. You have Ring Validation 100Ms and I have 256. You have Ring Indication Delay of 256 and I have 0. I had problems with my (old) phone ringing briefly at some stage, so I experimented a little. I will now try your settings to see if it helps with my next big problem --- I'm not getting a CLI number. Instead I get the Username I've allocated to my SPA. ah. Do you have callerid from BT (bt line?). I signed up for something called BT Privacy or so which is free and gives you callerid. If you turn on logging (debug) on the sipura it'll log the received callerid via syslog. Also helpful to check under info Last seen number or so. Conrad ___ --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] Linksys SPA3102 - PSTN hangup detection in the UK
Yep, just swop the two wires. Sometimes the Tip and Ring get reversed and most loop start interfaces don't really care (they work either way). It's worth a try since if the disconnect is a reverse polarity flash then the card may see not see this condition as it is already reversed. I have a similar problem with Foriegn Exchange line (FX) but I haven't had time to visit the client to check this out yet. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Henry.L.Coleman wrote: Just a thought ... try reversing the Tip and Ring Henry L.Coleman CEO Henry, Apologies for answering the wrong message in my last post. I thought I was answering the one from Conrad. Sorry! By reversing the Tip and Ring you mean physically in the wiring or somewhere in the SPA? I can see Forward/Reverse settings for Line1 in the config, but nothing on the PSTN side? Thanks, Faris. ___ --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] (no subject)
Hi all, the lists seems to be littered with disconnect problems using various equipment (TDM 400,Linksys etc etc.) My question is very simple and could make for good solution to Asterisk users. Since * can detect various tones according to different country standards would it be possible to disconnect on the 'off-hook' warning tone? This tone is: 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz, at a cadence of 0.1s on, 0.1s off. is it very easy to establish if this tone is present on the line simply ask the non-asterisk end to hangup and wait on the line if you hear a loud warning tone then that is the disconnect tone!. If this tone could be detected and issued as the # then * would see this as a dialled digit and force a disconnect. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada ___ --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] say Asterisk to answer
The latest X-lite version has autoanswer button on the front.. marked AA Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi Greg, Idefisk support Auto-answer only in a biz version I suppose you got free version.. You will find more details http://www.asteriskguru.com/idefisk/free/ Cheers, Giovanni 2006/10/19, Gregory Duchatelet [EMAIL PROTECTED]: Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on Accept button. Greg ___ --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 -- Giovanni Miano ___ --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 activate recording (automon)
Hi Andrea, Try the following: featuredigittimeout=1500 ; Slow down digits for the record [featuremap] automon = *0 ; One Touch Record Use both option switches(wW) Check that the dial plan on your SIP phones doesn't preclude this feature code. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw under General Setting two fields, denoted Asterisk Dial command options and Asterisk Outbound Dial command options Here the help says something about w and W options, but every combination of this options does not produce anything Anyway, apart from FreePBX, what I have to check ? And moreover, what are the correct actions to do to record a call ? Let's say extension 555 calls extension 567, 567 answers the call and then press *1 and no other key ? I am trying with at320 sip phones and snom 320 sip phones thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] Reception Console
I have a bata site we can use to test your software. Please contact me [EMAIL PROTECTED] Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Secure multi-tenant partitioning capabilities? What is your distribution intentions, commercial or GPL? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Sunday, October 15, 2006 10:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Reception Console We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales -- Paul Hales Technical Manager AsteriskIT www.asteriskit.com.au bus: 03 8320 8106 mob: 0434 673 529 ___ --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] two SIP phones as one line
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___ --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] two SIP phones as one line
The quirk of your old PBX is in fact exactly what happens when you put any two analog phones on the same line. The easiest way to duplicate this is to connect another analog phone to your ATA. Some analog phones can indicate when the other is on the line and can put a call on hold locally. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi, I am looking to replace a quirk of our old PBX system functionality with asterisk but after searching, archives, wiki, etc.. I cannot figure out how. Here is what I would like to do: PhoneA is a SIP hard phone, phoneB is an analogue phone connected to a SIP ATA. When an incoming call comes in, I would like to ring both phones, but if phoneA is answered first, I would like phoneB to be answered as well and left in a off hook state so that when someone picks up the receiver of phoneB, they can hear and participate in the conversation between the calling party and phoneA. I believe I would have to put both phones in a MeetMe conference, but how to I auto-answer phoneB when phoneA has answered the call? I suspect that this may not be possible with asterisk, but would like confirmation of that. Thanks in advance. -m ___ --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] Call bridged, but no sound
I have had this problem before and it always turns out to be the fire wall. You SIP registration and signaling (port 5060) is going thru okay but the audio signals use a range of different ports which (if blocked) will cause the problems you experience. Try putting * in DMZ to test this theory Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hello everybody, I have a problem and already browsed the mailing list archives but didn't find any help. So I ask here My new * Box ist up runnig. Got access to the SIP server of my Internet provider (Userid, password, phone number, ...). And yesterday I tried my first calls to the outside world. (Internal calls work). Now when I call from the SNOM-360 connected to Asterisk to my cellphone (or to any other number), the call is set up, but both sides cannot hear each other. The asterisk console says: -- Executing Dial(SIP/1-08182b48, SIP/[EMAIL PROTECTED]|30|r) in new stack -- Called [EMAIL PROTECTED] -- SIP/inode-outbound-081906e8 is ringing -- SIP/inode-outbound-081906e8 answered SIP/1-08182b48 -- Attempting native bridge of SIP/1-08182b48 and SIP/inode-outbound-081906e8 With SIP DEBUG, somewhere in the tons of output i finde the following lines: Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.201:56190 Found description format pcma Found description format telephone-event Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing sip:[EMAIL PROTECTED]:5060;line=vz5y8h67 for address/port to send to set_destination: set destination to 192.168.1.201, port 5060 Transmitting (NAT) to 192.168.1.201:5060: ACK sip:[EMAIL PROTECTED]:5060;line=vz5y8h67 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK559b80fb;rport From: sip:[EMAIL PROTECTED];user=phone;tag=as2bff66b8 To: Chef sip:[EMAIL PROTECTED];tag=3mzvp0gi42 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 From this I understand that both sides agreed on a common codec (alaw). As soon as the connection is up and the receiver is lifted on both sides, the leds of the DSL Modem between Asterisk and my ISP, and the leds of the switch between Asterisk and the SNOM phone start rapidly flashing. So I assume there are lots of data packets on the wire. But no sound in both receivers Could it still be a firewall problem? Any hints or ideas? Norbert ___ --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 big is *your* dialplan??
Frankly waiting for the box to break will loose you the client. I would change the box but use the original Hard Drive, it only takes a couple of minutes on a small system. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada On Wednesday 11 October 2006 15:16, Douglas Garstang wrote: Are you serious? Would you really just wait until a system looked like it was on shaky ground before deciding to build a new one? What about if some other component failed? What about the myriad of other failures you didn't think of ahead of time? Do you really think that it's ok for a system that hosts less than 50-60 users to be unavailable while a new system is built? We're talking about VOICE service here, not someones email access. People can do without email for a period of time but they are very sensitive to a lack of dialtone. How many small to medium businesses do you know with redundant Meridian hardware on the shelf? Hell, how many small to medium businesses do you know that have their KSU or PBX on a UPS? Btw, I showed this email to a senior telecom guy here in the office. Initially his eyes widened, and then he laughed and said he'd certainly never get you to build his telecom infrastructure. I'd love to see a senior telecom guy in a small to medium business. His eyes would fall out of his head and he'd be escorted out of the building laughing. I completely understand your need for redundancy, but most places that Asterisk is being installed in to don't have that kind of redundancy in place in the first place. -A. ___ --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] Test Call Script
I can think of a couple of ways to achieve testing of a PSTN line but this
would seem to be the easiest.
Attempt to call an incoming PSTN/SIP/IAX line from your outgoing PSTN
trunk, answer the call at a vmail box and notify you of a message via
email.
insert a delay of x minutes and do it again.
Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada
yes.. actualy use 1 did for each proxy to check..
then inbound for each use the method he described..
On 10/12/06, Mojo with Horan Company, LLC [EMAIL PROTECTED]
wrote:
on an analog Zap PSTN channel, you have no real way of determining if
the remote side answered, because, as you discerned, it IS considered
answered as soon as asterisk opens the channel.
How about you contact another asterisk server through the PSTN, and dial
through to an extension on that remote asterisk server that, in turn,
notifies the first asterisk server maybe via the internet that it was
received?
for example, consider the following php script accessupdate.php on
primary asterisk box:
?php
if (!strcmp($_GET['update'], 'true'))
{
touch(/etc/asterisk/secondary_server_last_access);
}
?
then primary calls secondary box through PSTN, and through the magic of
DISA or CID or what-have-you, dials through to an extension that
executes
System(wget -q -O /dev/null
http://primary-server/access_update.php?update=true)
then hangs up. then primary server checks the last-access time of
/etc/asterisk/secondary_server_last_access to make its decision, via
cron script or bash script triggered through the dialplan subsequent to
the initial dial-out.
This is of course a very rudimentary on-the-fly thing I came up with,
but think outside the box and this may be the easiest way for you to do
what you want.
Moj
John Kane wrote:
I am trying to write a script to attempt to make a call on a Zap
channel, and if it fails, send an alarm. I can generate the call, but
because the Zap channel accepts the call, even though the other end
never answers, it sees it as a successful call, which it isn't.
Anyone have any ideas on this? Thanks.
!DSPAM:500,452d7fa8199221504517840!
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!DSPAM:500,452d7fa8199221504517840!
--
Mojo [EMAIL PROTECTED]
Office Manager, Horan Company, LLC
(907) 747- x112
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--
Mike
Sales Manager
http://www.theclubvoip.com
Making it happen
1.877.807.VOIP (8647)
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