[asterisk-users] Issue with RDNIS
Hello,
Does anyone know why I am unable to retrieve the Redirecting Number?
I've done a pri debug span 1/1 and can see the number being passed
correctly to Asterisk:
Redirecting Number (len=15) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Ext: 0 Presentation: Presentation allowed
of network provided number (3)
Ext: 1 Reason: Unknown (0) '7145551212' ]
Now in my dialplan I have the following two lines:
exten = 5914, 1, NoOp(${CALLERID(rdnis)});
exten = 5914, n, NoOp(${RDNIS});
The resulting output to the Asterisk CLI is this:
-- Executing [5...@test:1] NoOp(DAHDI/1-1, ) in new stack
-- Executing [5...@test:2] NoOp(DAHDI/1-1, ) in new stack
I am running Asterisk 1.6.0.6 and dahdi 2.1.0.4.
Thanks in advance,
Mitchel Constantin
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Re: [asterisk-users] Cisco acquires Jabber
Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. Dean, I'm right there with you. My money is on them using it as the first step in a larger strategy to provide a framework for applications to run on network without needing an operating system. Think Amazon Elastic Cloud (with P2P and presence built in) but for applications. On Sat, Sep 20, 2008 at 11:18 AM, Dean Collins [EMAIL PROTECTED] wrote: No I know they just bought the company and not the protocol basically they bought engineering bums on seats. http://deancollinsblog.blogspot.com/2008/09/cisco-acquires-jabber.html Cisco obviously didn't buy jabber.com engineers to implement a Cisco IM platform for their retail clients and that they must have something much bigger in mind. You could possible see different Cisco devices communicating with each other (or even using an api to communicate with other manufacturers devices) eg, you might have an XMPP api to 'discover' appliance functionality or to communicate status updates. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.cognation.net/profile -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of David Backeberg Sent: Saturday, 20 September 2008 10:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco acquires Jabber I wonder what this means in the long run for the open development of this platform? Not a darn thing, unless Cisco screws around and makes an incompatible version of a jabber server and client that doesn't play according to the protocol. Microsoft Java, anybody? We'll see how long this list stays true: http://www.jabber.com/CE/JabberXCPInteroperabilityOptions Cisco didn't buy the protocol, and literally dozens of open-source projects that use the protocol in various ways are not affected by this. They bought one commercial implementation of a Jabber server (arguably multiple implementations). ___ -- 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 ___ -- 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 -- Mitchel Constantin Weavver. Your voice, just better. Business Development: +1-714-726-8071 XMPP: mitchel.at.weavver.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom Voicemail emails
Depending on what e-mail server software you use, it may be easier to direct the voicemail to a specific e-mail address and have your e-mail software rewrite the subject, and then forward it on to your boss. On Thu, Sep 18, 2008 at 11:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Thursday 18 September 2008 09:54:39 Steve Anness wrote: So here is the deal. I have an Asterisk server here at work that I have recently taken over and the boss is wanting the server to do a lot of things that it didn't do before. I have already configured much of what he wanted including a voice messaging line where anyone can call in and leave a message and then he would get that message in his email. However, the boss wants his email subject to read something like This is an urgent message through the HISG voice messaging system so he knows that that message came through that number as opposed to his voicemail box that already gets forwarded there. The default is the [PBX]: New Message 10 in mailbox 0307. At second glance he would know which voicemail box is his line but he wants things to be different and so I am trying to make that happen. I know there is the 'emailsubject' option. I haven't tried this yet but my concern is that it will set the subject the same on every single box (obviously what the command is designed for). I can I customize a voicemail message so that if something comes in on our 0307 line it has a certain message and then we might get a message on 1942 line that we want a different subject. Currently, there is no such capability. Coding it would not be very difficult, however. I would suggest using the per-mailbox settings and simply adding the option to code an email subject per mailbox, and then default to the generic subject, if one is not otherwise specified in the mailbox. -- Tilghman ___ -- 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 -- Mitchel Constantin Weavver. Your voice, just better. Business Development: +1.714.726.8071 XMPP: mitchel.at.weavver.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Problems with SpanDSP
Steve, Are there any plans to get a newer version of spandsp working with Asterisk? On 8/29/07, Steve Underwood [EMAIL PROTECTED] wrote: Christian Peter wrote: Hi list, I'm running current SpanDSP http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.4pre6.tgz with Asterisk 1.2.22 somewhat successfully. Most Fax machines do work but I have problems with people having Tobit FaxWare and Shamrock CapiFax. http://www.tobit.com/login/mrd.asp?CategoryID=120 http://www.shamrock.de/ I've got black bars over the pages. In Tobit some content is Ok, other is covered by the black bars. Anyone else has simliar problems? I talked to Tobit and they said there should be an option somewhere in SpanDSP to disable Fax header crossbars. But I found none. Can anybody help me with this issue. Please no switch to Hylafax mails, because I'm very happy with SpanDSP, it integrates nicely and works most time. Thank you, Regards Christian Peter I assume if you are using spandsp-0.0.4pre6 you have adapted app_rxfax.c. and app_txfax.c to work with it. I haven't heard from anyone using tobit or shamrock software (who on earth wants to call their company tobit? weird). I have no idea what fax header crossbars might be. Do they have some kind of bicycle integration in their product? :-\ Is your problem when sending from Asterisk or receiving? Can you enable debug and e-mail me a log and (assuming its a receive problem) the resulting TIFF file. Steve ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speech Rec on Voicemail
Nuance offers an SDK to do something similar, I think they say you can only expect between 45-60% accuracy using it though. Total cost is about $6K to $8K for one server license. If there are enough people interested in pooling money I'd be willing to help set up a system to process voicemails and provide the Nuance converted transcript. However, I figure the low accuracy would be the biggest turn off from using Nuance. On 8/23/07, Stephen Bosch [EMAIL PROTECTED] wrote: Ryan M. Colbert wrote: I've had requests to processes incoming voicemails with voice recognition routine and add the output text to the body of the email message from * with the attached .wav file. Has anyone implemented this type of feature and willing to share some notes? I seem to recall that Lt. Cmdr Jordi Laforge did some stuff like this not too long ago. I get requests like this all the time -- but the technology is very far from being there. -Stephen- ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia cell connected to Asterisk
Jonathon, Are you talking about using the built in SIP client on some Nokia phones? I'm using an E90 with Asterisk and it works very well. I used Google for help and it returned plenty of results. Cheers, Mitchel On 8/19/07, Steve Totaro [EMAIL PROTECTED] wrote: If it is bluetooth and you don't mind running Asterisk 1.4 trunk, you should look at chan_mobile. Thanks, Steve Totaro From: [EMAIL PROTECTED] on behalf of Jonathan GF Sent: Sun 8/19/2007 6:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Nokia cell connected to Asterisk Hi folks, i've been looking for in many sources but i cannot see clear if the options i'm chasing is feasible with Asterisk. I understand that should be. I would like to connect a nokia cell to Asterisk but i don't know how exactly. Any ideas, inputs, docs or refs will be welcomed. Thanks in advance. Jonathan GF ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LumenVox Speech Recognition
Randulo, There is an extra letter in the url you provided, it should be: http://lumenvox.com/partners/integrator/digium/applicationzone/ I think that the LumenVox pizza and weather demo would sound much better if the prompts were professionally recorded. On 8/12/07, randulo [EMAIL PROTECTED] wrote: Nitesh, I've messed with the Lumenvox starter kit. If you are serious about this field, I think it's a must see. It was easy to set up and there are demos available. Their support is excellent. There is a quiet mailing list where questions are never ignored and most problems are solved AFAIK. Unfortunately, I have not had time to get to the next level of developing new demos for it, but I hope to do so some day. Take a look here for demos, etc. http://lumenvox.com/partners/integrator/digium/applicationzone/i /r ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LumenVox Speech Recognition
Dean, Are you aware of any better options for speech recognition? (though I'm sure more expensive) On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Dean... will update you on the progress... Cheers, Nitesh Dean Collins wrote: Hi Nitesh - yep great place to start. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Saturday, 11 August 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Thanks Dean and Steve, I am planning to use for my IVR notification application which is built using PHPAGI and A2Billing (Callback, Calling Card). I saw the $50.00 Starter kit does it provide some functionality? Cheers, Nitesh Dean Collins wrote: Hi Steve, no I'm no expert at all I do however (or did) have an interest in building a far more comprehensive solution for an ASP solution combining other solutions that would have helped the asterisk community however could never get it off the ground. Nitesh to answer your original question...Lumenvox is great value for the money and works well - however there are limitations but for 90% of applications will work great. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, 11 August 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean Collins is probably the list expert on this. Thanks, Steve Totaro Nitesh Divecha wrote: Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] LumenVox Speech Recognition
Dean, Hmm.. I was hoping something that could be used with Asterisk on the machine locally... Nuance doesn't seem to offer that. On 8/11/07, Dean Collins [EMAIL PROTECTED] wrote: Nuance etc. and Steve to answer your questions - lumenvox just doesn't have the engine or phonetic capabilities that some of the the larger systems have. Like I said before - I've been stunned considering how cheap it is how good it is but. if you are looking for a less defined utterance structure it has limitations. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Saturday, 11 August 2007 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean, Are you aware of any better options for speech recognition? (though I'm sure more expensive) On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Dean... will update you on the progress... Cheers, Nitesh Dean Collins wrote: Hi Nitesh - yep great place to start. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Saturday, 11 August 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Thanks Dean and Steve, I am planning to use for my IVR notification application which is built using PHPAGI and A2Billing (Callback, Calling Card). I saw the $50.00 Starter kit does it provide some functionality? Cheers, Nitesh Dean Collins wrote: Hi Steve, no I'm no expert at all I do however (or did) have an interest in building a far more comprehensive solution for an ASP solution combining other solutions that would have helped the asterisk community however could never get it off the ground. Nitesh to answer your original question...Lumenvox is great value for the money and works well - however there are limitations but for 90% of applications will work great. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, 11 August 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean Collins is probably the list expert on this. Thanks, Steve Totaro Nitesh Divecha wrote: Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh ___ --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 ___ --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 ___ --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 ___ --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 ___ --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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
Re: [asterisk-users] LumenVox Speech Recognition
Nitesh, They claim to support numbers on their website so I would say yes. On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Dean, Can the LumenVox Speech Recognition engine understand numbers? Sorry to ask stupid questions but kinda curious... as for my application all I want is to the software to understand the numbers and provide me the output. Cheers, Nitesh Dean Collins wrote: No they have a standalone solution - lol NLVR is a whole separate server (or server farm) in most onsite installations. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Saturday, 11 August 2007 7:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean, Hmm.. I was hoping something that could be used with Asterisk on the machine locally... Nuance doesn't seem to offer that. On 8/11/07, Dean Collins [EMAIL PROTECTED] wrote: Nuance etc. and Steve to answer your questions - lumenvox just doesn't have the engine or phonetic capabilities that some of the the larger systems have. Like I said before - I've been stunned considering how cheap it is how good it is but. if you are looking for a less defined utterance structure it has limitations. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of mitcheloc Sent: Saturday, 11 August 2007 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean, Are you aware of any better options for speech recognition? (though I'm sure more expensive) On 8/11/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Thanks Dean... will update you on the progress... Cheers, Nitesh Dean Collins wrote: Hi Nitesh - yep great place to start. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Saturday, 11 August 2007 11:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Thanks Dean and Steve, I am planning to use for my IVR notification application which is built using PHPAGI and A2Billing (Callback, Calling Card). I saw the $50.00 Starter kit does it provide some functionality? Cheers, Nitesh Dean Collins wrote: Hi Steve, no I'm no expert at all I do however (or did) have an interest in building a far more comprehensive solution for an ASP solution combining other solutions that would have helped the asterisk community however could never get it off the ground. Nitesh to answer your original question...Lumenvox is great value for the money and works well - however there are limitations but for 90% of applications will work great. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, 11 August 2007 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] LumenVox Speech Recognition Dean Collins is probably the list expert on this. Thanks, Steve Totaro Nitesh Divecha wrote: Hello All, While looking for solution to solve my Callback DTMF problem, I came across LumenVox Speech Recognition software. Has anyone tried out? Need some feedback before I purchase it... Please help... Cheers, Nitesh ___ --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 ___ --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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
Re: [asterisk-users] OT - Callto:// tags inside web pages
Ollvier, You could use the Firefox plug-in for Snap. It will auto detect numbers on a webpage and make them dialable. Cheers, Mitchel On 8/7/07, Olivier [EMAIL PROTECTED] wrote: Replying to myself, I got this : http://en.wikipedia.org/wiki/URI_scheme Anyway, I'm still wondering which is the best way to go, for standard and usage compliance. It seems that callto: was initialially used by netmeeting before being by Skype software. I could find a tab in XP Internet Option configuration panel, where you can either select Skype or Netmeeting to be launched as default software whenever a callto: tag is clicked. If you had to design a web site, which scheme would you adopt ? cheers ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial from Phonebook of Evolution or Thunderbird
Alexander, Check out Snap. It plugs into Thunderbird on Windows. On 7/29/07, Alexander Topolanek [EMAIL PROTECTED] wrote: Hi, does anyone know about a plugin that allows dialling a contact from the phonebook of evolution or T-bird? -- Alexander Topolanek http://www.topolanek.at ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Wiki
Using the CLI is another good way to find that information quickly: nox*CLI core show application Playback -= Info about application 'Playback' =- [Synopsis] Play a file [Description] Playback(filename[filename2...][|option]): Plays back given filenames (do not put extension). Options may also be included following a pipe symbol. The 'skip' option causes the playback of the message to be skipped if the channel is not in the 'up' state (i.e. it hasn't been answered yet). If 'skip' is specified, the application will return immediately should the channel not be off hook. Otherwise, unless 'noanswer' is specified, the channel will be answered before the sound is played. Not all channels support playing messages while still on hook. This application sets the following channel variable upon completion: PLAYBACKSTATUSThe status of the playback attempt as a text string, one of SUCCESS | FAILED nox*CLI On 7/27/07, Bruno De Luca [EMAIL PROTECTED] wrote: http://www.voip-info.org/wiki-Asterisk+cmd+Playback you can use google asterisk cmd playback.. bilal ghayyad wrote: Hi List; I am trying to use wiki via the link (http://www.voip-info.org/wiki/index.php?page=Asterisk) in effective way to find the needed resource for me, but still it is hard to arrive for the needed information. For example: what is the best (shortest) way to search for information related to the command playbak()? Using the backlines, it make the eyes feel hard by keep reading without alphapatic orgnaization, any advise how to search fast in this website? Regards Bilal Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545469 ___ --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 -- Bruno De Luca, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02 997663.12, Fax: 02 91390172 ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best and easiest soft phone for my Dad..
It is possible that the hotel is only allowing certain incoming/outgoing ports as well (i.e. just allowing DNS and HTTP traffic). A VPN *might* help with that. On 7/21/07, WipeOut [EMAIL PROTECTED] wrote: Hi, Here is the situation.. My Dad is working on contract in overseas.. He has internet access in his hotel.. He wants to be able to talk to my Mum but the calls are expensive.. I have an asterisk box setup for my business and it has a public IP etc.. My Mum has access to a working phone extension on this box.. I got my Dad to install X-Lite but for some reason it won't register and trying to talk him through working out whats wrong is proving to be difficult.. Also I haven't used a softphone in years.. It could be the NAT in the hotel, it could be a firewall or any number of things that can cause these issues.. It could even be X-Lite or something running on his PC.. So I am looking for a softphone thats really simple to setup and as foolproof as possible.. If SIP is likely to be problematic to setup then I have no problem getting him to use IAX but will need suggestions of which IAX softphone to use and also how to configure it in the iax.conf (haven't done this before).. Any suggestions welcome and appreciated.. Thanks.. ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Single ringer phone for incoming calls, that anyone can answer
Tom, It sounds like this is what you need: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups Cheers, On 6/23/07, Tom Lanyon [EMAIL PROTECTED] wrote: On 23/06/2007, at 6:01 PM, Gordon Henderson wrote: Can't you give the ringer phone a different ring tone/tune ? Gordon Gordon, Well, we can give the 'ringer phone' whatever ring tone we wish, but that's not the issue at hand. We need a way to answer the incoming call on our personal phones without explicitly directing the call to dial every phone (so they don't ring). Regards Tom ___ --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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] getting call status using Manager API
Matthew, If you hold open the connection to the manager api after initiating your call you should see packets returning from Asterisk which you can parse to get the call state. I'm not sure if there is a way to do it later after the fact. You can execute commands like Show Channels through manager though. On 5/16/07, Matthew M. Boedicker [EMAIL PROTECTED] wrote: I am originating a call using the Originate action in the Manager API. It calls one party, then when they answer does the Dial application and calls another party and connects the two. Is there a way using the Manager API to check back later on the status of this call (is it still up, etc.)? I have found the Status API action, but I don't know how to get what to pass in for the channel parameter. Thanks, Matthew Boedicker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk M$ SQL Server
I've never heard of M$ SQL Server? On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could point me to a howto / guide or relate their experiences with this, that would be great ! Thanking you all in advance. Callum P.S. Before anyone asks; it's a legacy database that has to be kept on SQL Server for other business reasons. ___ --Bandwidth 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 M$ SQL Server
Oh. Got it now. Well, in this case I think you are looking at it backwards. I imagine most users with this requirement write AGI scripts that talk to their databases then communicate back. You can use FastAGI and run your code on a windows server, or you could use any other programming language (PHP, MONO) to acheive the same, but then place it locally on the Asterisk side. On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Oh Microsoft SQL Server for those unfamiliar with the term M$ ;) mitcheloc wrote: I've never heard of M$ SQL Server? On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could point me to a howto / guide or relate their experiences with this, that would be great ! Thanking you all in advance. Callum P.S. Before anyone asks; it's a legacy database that has to be kept on SQL Server for other business reasons. ___ --Bandwidth 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Automated Outbound Messaging
It sounds like what you want is called a predictive dialer? There are several listed on the voip-info wiki. On 3/20/07, Lee Jenkins [EMAIL PROTECTED] wrote: Cory Andrews wrote: I have a client application looking for an Asterisk based solution. Client wants to deliver pre-recorded messages for a variety of clients. Wondering if anyone is offering an middleware for Asterisk for management of outbound messaging? Someone can correct me if I'm wrong, but I think a friend of mine mentioned that TrixBox has a gab cast function. It also shouldn't be that difficult to put together a script to do this. I actually have plans to do this myself, but no need for it just yet... -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Microsoft launches first PABX
Is that FUD really necessary? On 3/19/07, C F [EMAIL PROTECTED] wrote: Wow. Does that mean that someone calling into the system will be able to use the Embedded voice recognition technology and halt the system by saying stop now? Or will it just do that without anyone saying it? On 3/19/07, Dean Collins [EMAIL PROTECTED] wrote: http://www.crn.com.au/story.aspx?CIID=76033eid=4edate=20070320 The company developed Response Point to work alongside traditional phone systems or voice-over-IP systems. Continuing its recent foray into the market for digital communications products, Microsoft on Monday introduced its first packaged digital phone system for small business. Anyone know anything about it? Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +1-917-207-3420 Mb +61-2-9016-5642 (Sydney in-dial). ___ --Bandwidth 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Jajah.com like script?
You are missing something. Initiate the call to channel for the first user (i.e. ZAP/g1/phonenumber), and have their destination extension the second phone number. On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: So does this mean that we have to dump the two callers into a Meetme context??? The problem there is what if one of the callee's doesn't accept the call (call screening). There is no easy way to kick the other user out of meetme and dump him to a vmail context. Am I missing something? R On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Thanks Steve! I will give it a shot. R On 3/16/07, Steve Edwards [EMAIL PROTECTED] wrote: Search on voip-info.org for call files. On Fri, 16 Mar 2007, Ritesh Agrawal wrote: Hi Folks, I am planning to create an internal portal where the users can enter two phone numbers (theirs and the party they are trying to reach) and connect the two of them by initiating two calls from Asterisk. Any pointers on how to initiate two calls and then bridge them (without using meetme?). Ideally, I would like to do a call screening as well. Thanks for your help. R Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] While the VoIP-Info.org site is down...
Just a heads up guys. I'm currently attempting to recover the website through spidering the Google cache. I'll let you know how it turns out. On 3/15/07, Drew Gibson [EMAIL PROTECTED] wrote: Stephen Bosch wrote: Patrick May wrote: On Wed, Mar 14, 2007 at 10:05:20PM -0400, Matt wrote: Yikes.. you'd think a server would be running RAID. At any rate.. Please feel free to visit http://www.voip-wiki.us I have set this up to be able to hold information for the Asterisk community. I will also gladly allow others to mirror it. It is sitting in a climate controlled data center in Central PA on a server with RAID. Additionally, it is at the end of 95Megabytes/second on a BGP redundant connection. Please feel free to use it, if the community feels it can be useful... additionally, I would love to setup some rsync mirrors with others so that we can have redundant backups of this very valuable information. The previous message to the list was they lost 3 of 4 drives in the array. I'm not sure of any RAID that can sustain 75% hardware loss and still function. As somebody else has already pointed out -- There must be more to it. Let's say three of four drives failed -- the odds of them failing at the same time are vanishingly slim; but if you're not paying attention, and you operate with a degraded volume, well... then you get what you deserve. RAID or no RAID, the site should have one or more mirrors. -Stephen- The odds of multiple drive failure are a lot higher than you think. Failing power supplies or power spikes are common to all drives and controller failure on a drive can throw noise back onto the SCSI bus causing corruption on other drives. Although the other affected drives are not physically damaged, your data has evaporated none the less. We have already had one multi-drive RAID failure on our main file server (only one drive was physically failed) and a single drive and power supply failure on our Asterisk box. RAID 1 and redundant power supplies saved the day. Spring and Fall are the special Hardware Failure Seasons! Seems to affect power supplies, hard drives and light bulbs in particular. regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voip-info.org is back!
That's awesome, we were nearly done with the spider too! On 3/15/07, Sean Bright [EMAIL PROTECTED] wrote: Looks like the site is back up. Don't all hit it at once, it might go down again ;-) Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] XM Radio Stream to Asterisk
Oh Dovid, You always seem to be up to something! All these strange projects ;). On 2/26/07, Dovid B [EMAIL PROTECTED] wrote: Hi Guys, Anyone figure out a way to have XM radio work over asterisk (Not thru an audio card but over the internet) ? Either where a sip phone dials an extension (i.e. *202 will go to XM channel 202) or maybe with a confrence room so multiple people can call in and listen. Thanks for any and all ideas. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Cordless SIP Phones
You can usually meet this by using an FXS adapter along with a standard cordless phone and thus achieve the same result. If you *really* want a voip-cordless phone, check this out: http://www.voiplink.com/Aastra_480i_CT_p/aastra-480i-ct.htm p.s. I've had good experiences with VoIP Link (no I don't work for them!) On 1/28/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I share your frustration. Might I suggest a Grandstream HT-386 (or 486, etc) gateway to a regular cordless phone? On Sun, Jan 28, 2007 at 09:52:19PM -0500, Edward Halman wrote: Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you. Ed ___ --Bandwidth 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] windows mobile 5 softphone for square screen devices
I've been trying the SJPhone with no luck. Where did you download the Xten version from? On 1/16/07, Anton Krall [EMAIL PROTECTED] wrote: Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] WIFI SIP- The Best phone
Wait for the iPhone...seriously. On 1/9/07, Jerry Glomph Black [EMAIL PROTECTED] wrote: I've had the E70 for about a month. The first few days were not fun. But now that I've learned the gotchas and the workarounds, it is GREAT. You -can- configure it, and asterisk, to work perfectly together, every time. With automatic failover to conventional GSM phone behaviour if not in 802.11 land. I would be happy to give these to blinking 12:00 users, if -I- preconfigured them. This thing is great. Far exceeds expectation. Battery life with 802.11 is great, not discernibly worse than without it. _ On Tue, 9 Jan 2007, Stephen Davies wrote: On 09/01/07, Nigel Kendrick [EMAIL PROTECTED] wrote: I've had a play with a Nokia E70 - the 'bar' version of the E61 and gave up! Menu navigation is dire - I went through hoops trying to get SIP working - I know from others it can be done, but I bailed out when I realised that to put these phones in the hands of inexperienced users would be a recipe for a lot of frustration and support calls. Ironically I was going to recommend the E70. It is true that the menus are complex but once configured it does do what it says on the tin - provide a very effective merging of SIP over WIFI and GSM all in one unit. 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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 SIP- The Best phone
Those wifi phones are neat but I'd rather not carry around two devices, does anyone know of any good dual-mode GSM/SIP phones? I'm using a T-Mobile MDA right now and it is way too slow. Apparently the Nokia e61 has a built in SIP client, but there might be a new model around the corner (worth the wait?) Suggestions? Thanks! On 12/30/06, Matthew Mackes [EMAIL PROTECTED] wrote: Well, We have found that it will stay on line for around 6-7 hours with a full charge and no talk time, and we can get about 2 hours of straight talk time. We have given these units to our cashiers that are located out in parking lots, and our location mangers who walk a large location (Outside and Inside) and they are able to carry a phone for about 5 hours if they talk on it for a total of an hour during that time without charging. Our goal is to replace our current Radio's that we use for on site communications. Over all, I have been very happy with the charge vs talk time So much so, it hasn't ever been an issue. We have around 280 phones out at our locations at any one time, 150 in use.. So, for every phone we have in use, I have a spare waiting in a charger... that way a phone is always waiting in case. However I have only had to do that because we run a 9am until 11pm day, and our staff requires 100% phone uptime while they are on the move. Our managers until recently carried company issued cell phones- however with the full corporate move to Voip, Asterisk, Aastra Desk Voip phones and the Zulty WIP 2's I have had to ensure 100% uptime by throwing hardware at the project. However we really don't need allof the redudent units, my users are just a bit unreasonable. Their cell phones don't have any better talk time then the Zultys. However by connecting all of our locations with Voip over our PRN (Quest) we have been able to save so much on Telco Charges that I have a large budget Available to keep us up- Hence the overkill However--- For a Normal Office At our Headquarters I have also deployed about 25 of these phones to normal Cube workers, and they have NO trouble with them. They put them in the charger when they think of it, and they always have a charged phone... They only need one, and it replaced the Desktop Digital Nortel phone they used to have. They can take it to any of our locations, and boom, instantly on line with the same Number, Extension and services. And Since we are in 3 States, we use several collocated Asterisk Servers to create a Private Toll bypass network of Voip. Illinois, NY and Pennsylvania are all local PSTN calls no matter what state you are in. Also, if you are interested, we use a Alcatel Wireless Controller unit to provision, route, and control our 600 + Alcatel Wireless Access Points. Because all of the AP's are routed back to one controller, it helps with all IP roaming, not just WIFI Voip. I have also used Cisco 1200 AP's and those also work very well. I just wish I could find a DEPENDABLE service for DID and Termination via SIP I current am working with Vitality, and they are so, so. I hope that helps. Matt Mackes Delta Sonic Car Wash Systems Noah Miller wrote: HOWEVER- The Zultys WIP 2 is an INCREDIBLE WIFI B/G SIP PHONE- IT IS EXCELLENT IN ALL RESPECTS. Thanks for the tip! I hadn't seen these advertised before, and I've been searching for some time for a Wifi SIP phone that can handle multiple line appearances. One Question: Really only 12-13 hours on the standby time? That seems pretty short in comparison with all the other wifi phones. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Checking voicemail from outside
You could be using an older version of Asterisk that doesn't support it?
On 12/28/06, Phil Finkler [EMAIL PROTECTED] wrote:
Rob,
Interestingly enough, I'm using that same sample macro, and that line is
indeed in there, yet when I hit *, I hear the tone to leave a message. Any
ideas?
Phil
Phil,
Add this to your extensions (I have mine in a macro)
exten = a,1,VoicemailMain(${ARG1}); If they press
*, send to
Voicemail
so it should look like...
exten = s,1,Dial(${ARG2},13,${ARG3})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) exten =
s-BUSY,1,Voicemail(b${ARG1}) exten = a,1,VoicemailMain(${ARG1})
I have a few other things in there as well, but those are the lines that
should do what you want. When you press *, you are prompted for a password.
Rob
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Re: [asterisk-users] Happy X-mas
Ditto, Happy Holidays everyone! On 12/23/06, Carlos Rojas [EMAIL PROTECTED] wrote: Hello everybody HAPPY and Merry Christmas to all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Click to dial apps always show from asterisk
You can use the CallerID parameter of the Originate command to override the default caller id. It's listed on the wiki with examples: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate Cheers On 11/27/06, Eric Bishop [EMAIL PROTECTED] wrote: We have calls that originate click-to-dial apps that use the manager interface. As most of you know these apps first ring your handset so that you pickup the handset and then place the outbound call once you have picked up. When they first ring my handset (before me picking up the handset) the call shows as being from asterisk. Is there any way to change this from name to something the average joe understands such as PBX? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Click to dial apps always show from asterisk
Not that I know of. Try Snap it has a setting to let you force the Caller ID. On 11/27/06, Eric Bishop [EMAIL PROTECTED] wrote: I am trying to do it with FOP and Calling Circles. Both have closed code. Anyway to do it from Asterisk? On 11/27/06, mitcheloc [EMAIL PROTECTED] wrote: You can use the CallerID parameter of the Originate command to override the default caller id. It's listed on the wiki with examples: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate Cheers On 11/27/06, Eric Bishop [EMAIL PROTECTED] wrote: We have calls that originate click-to-dial apps that use the manager interface. As most of you know these apps first ring your handset so that you pickup the handset and then place the outbound call once you have picked up. When they first ring my handset (before me picking up the handset) the call shows as being from asterisk. Is there any way to change this from name to something the average joe understands such as PBX? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Question about TFTPD server
Remember to do a service xinetd restart after changing disabled to no in /etc/xinetd.d/tftp. On 11/16/06, JOAO CARLOS MOURA [EMAIL PROTECTED] wrote: Check is: very good http://www.it4u2.com/asterisk2.htm#SIPmacaddress http://www.loligo.com/asterisk/cisco/79xx/current/ - Original Message - From: Edwin Lam [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 15, 2006 9:22 PM Subject: Re: [asterisk-users] Question about TFTPD server Christian wrote: I have installed this package onto my Debian and placed the files i want the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to work. Are ther any special settings I should do to this server? Many thanks for all your help, which tftp server package did you installed? make sure your /tftpboot directory and all the files inside is at least readable by everyone. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 ___ --Bandwidth 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Survey: In what ways do you use Asterisk at yourhouse?
On 11/13/06, Dovid B [EMAIL PROTECTED] wrote: How much did the hardware cost you to set this up for your door ? From memory... the strike was around $95, the relay board between $25 and $50, and the power supply was only a few dollars, so you could do it all for under $200. -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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: FW: [asterisk-users] Desktop integration
Snap will do this for you. (Check my signature) On 11/13/06, Ondrej Valousek [EMAIL PROTECTED] wrote: Hi Dean, I will check that site - thanks for the hint. The biggest problem I see with authentication and I do not think mexuar could help me here (and I am definitely going to pay $2000 for it :-) But it is another story... Thank you! Ondrej Dean Collins wrote: Ondrej, You could do it using Mexuar Corraleta but this is a commercial application for Asterisk (US$2,000) http://www.mexuar.com/products_sdk.shtml http://www.mexuar.com/downloads/Press/CorraletaLaunchPR-1-branded.pdf However it has a whole heap more functionality than what you are looking for. If you just want to do 2 legged outbound calls check out 'call files' on www.voip-info.org Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ondrej Valousek Sent: Monday, 13 November 2006 6:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Desktop integration Hi all, I am interested in integrating my telephone system (I am using hardphones and Asterisk) with my desktop - something like this: 1. someone sends me his/her phone number via email/icq 2. I cut/paste the number in some application/web page (?) 3. my phone starts ringing and when I pick it up I will get connected with the remote party. Now I know I have read some discussion about this possibility but I can not recall where. Many thanks for any point. Ondrej ___ --Bandwidth 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Survey: In what ways do you use Asterisk at your house?
For about a year and a half now I've had Asterisk set up to unlock my front door at my house when calling a certain number. I locked it down by using caller id (not the most secure, but hey nobody knows the phone number to my door). Speed dialing your front door is one of the coolest things you can do. On 11/13/06, Dave Fullerton [EMAIL PROTECTED] wrote: Earle Clubb wrote: snip The main reason for this e-mail is to see what other people are doing. - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. snip I'm using a plain old POTs line for my termination. I have a Pentium III 450 w/192 megs of RAM, a TDM400 with 2 FXS and 2 FXO ports, and asterisk 1.2.13. The two FXS ports provide an extension for upstairs cordless phones and another extension for the downstairs cordless phone. I also have a budgetone 102 in the office. One of the FXO ports terminates my POTs line and the other is connected to a Dock-n-Talk device that I use to plug my cell phone into when I'm home. I don't get great signal at my house, so I dock it in the location where I get the best signal. Any calls coming into my cellphone then ring all the extensions in the house. I can call out on my cellphone by prefixing any number with a 9 from any phone in the house. Asterisk provides voicemail and emails a copy of the voicemail to me at work. I have asterisk linked to another asterisk box where I work (via IAX over the internet, which is cable) so I can call home from from my desk or dial any extension at work from home and connect directly. Works great for talking to the wife on her day's off. I also have it connected (in a very limited fashion) to some X10 equipment. I've used it to turn on and off Halloween decorations in separate rooms all at once. Hope that's the kind of info you were looking for. Asterisk really is cool once you get it figured out. -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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Survey: In what ways do you use Asterisk at your house?
I did not write a how to, but it essentially involves installing a door strike that I purchased from smarthome.com, running some wiring, and a serial port relay controller hooked up to the Asterisk server. I then used the System command and a script to send the signal through the board to send power and thus unlock the door. I would highly recommend getting the strike professionally installed even if you are the handy type. However, there seems to be an unspoken code among lock smiths not to do such things for the average consumer On 11/13/06, Yu Safin [EMAIL PROTECTED] wrote: I am wondering how you do your door. is there a howto? It will also be cool if I can speed dial my garage door so I can take deliveries when I am not home. -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Manager API - Originate Call - Need Help
If I understood your question correctly, you just need to reverse everything. Channel = OUTGOING TRUNK i.e. ZAP/00982166501553 Context = default Exten = internal extension that points to - 0041435215301 Priority = 1 CallerID = 0041435215301 This will first initiate the call to the number 0041435215301 and then connect it to the internal extension you specify in Exten that points to SIP/0041435215301. Cheers On 11/1/06, Ehsan Khosrowshahi [EMAIL PROTECTED] wrote: Hi all, How can i originate a call from someone outside my sip-network (for example my PSTN home number) to one of my SIP number? I can originate a call from my SIP-network using this parameters in Originate call command : Channel = SIP/0041435215301 Context = default Exten = 00982166501553 Priority = 1 CallerID = 0041435215301 this works with out any problems I initiate a call from one of my network sip clients (0041435215301) and call someone at anyside of the world, but Can I initiate a call from (00982166501553) to one of my sip 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Snom or Cisco Phones?
My vote is definitely for Snom, I've worked with Cisco phones for years, but the Snom is much better integrated, and the feature buttons can be retooled for any environment, making custom installs very easy. On 10/31/06, Conrad Wood [EMAIL PROTECTED] wrote: On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote: Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia cars Not really. Both are very good phones. * My Clients prefer cisco because it looks more business-like. - The new snom phones do look better though and the side car rules. * The Cisco phone 'feels' very good in your hand, and the voicequality is superb. (I'd say slightly better than that of the snom 360) * Technically, I find the snom phone more advanced and I can do more cool stuff with it - Cisco doesn't seem to like giving features away in SIP. * Snom phones, for example, have freely programmable buttons that can park/retrieve/transfer calls, show line status etc. I can't get that to work with Cisco phones at all. * Putting custom ringtones (and choosing which ones to use) is a no-brainer with snoms and real trouble with ciscos. * On ciscos, I find the upgrade path from sccp to sip a totally unnecessary annoyance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Fxo box for asterisk ?
Check out the SPA-3000 from Sipura (www.sipura.com). On 10/30/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi do you know if they have external Box (not internal card) for connect Analog Line and Pri/Isdn to asterisk for incomming and outgoing calls ... 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 -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Escape from Voicemail
Here you go, from the voip-info.org wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail Also. during the prompt if the caller presses: '*' - the call jumps to extension 'a' in the current voicemail context. This needs an example '#' - the greeting and/or instructions are stopped and recording starts immediately. On 10/20/06, Jason Walker [EMAIL PROTECTED] wrote: I used to have fonality and I could press * when I got to someones voice mail to go back to the menu. I assume I add that to the dialplan but how? Thanks BTW I went back to 1.2.12 and transfer works and DTMF works and it seems to be much better for now. Thanks for you help Jason ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 SIP won't update?
I am experiencing the same issue. However, I have not tried the VersionStamp field and will do so tomorrow. If you find an answer please post it to the list. On 10/13/06, Tim Connolly [EMAIL PROTECTED] wrote: Does anyone know what triggers the 7970 to update its config? I was able to get it to update to SIP, but the config I used initially won't go away. I am making small changes to the SEPxxx.cnf.xml file and rebooting the phone, the phone is downloading the (TFTP) new config file, but I don't see any change on the phone itself. I've looked at the VersionStamp and incremented that, but still no go. Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel Constantin Snap - A desktop user interface for Asterisk www.snapanumber.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] Digium GUI?
No it's not, it's supposed to just be a framework for developers and resellersto create GUIs that can go on the appliance. On 9/19/06, shadowym [EMAIL PROTECTED] wrote: I am talking about the GUI that was announced as part of the new AsteriskAppliance.Sounds like it is going to be a full featured GUI like FreePBX. -Original Message-From: Noah Miller [mailto:[EMAIL PROTECTED]]Sent: Monday, September 18, 2006 8:11 PMTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct.Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access?Do you mean this?http://svn.digium.com/view/asterisk/trunk/static-http/ On 9/18/06, Don Fanning [EMAIL PROTECTED] wrote: You mean the menuselect ncurses screen?If yes, then yes... it's a gui. :) -Original Message- From: shadowym [mailto:[EMAIL PROTECTED]] Sent: Monday, September 18, 2006 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct.Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access? ___ --Bandwidth 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 listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel ConstantinSnap - A desktop user interface for Asteriskwww.snapanumber.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] Digium GUI?
You are incorrect. The GUI you are referring to is the framework I already mentioned. The webpages are static html _javascript_ (AJAX functionality). Asterisk has a simple built in HTTP serverin trunk now which will be used to serve the webpages up and keep the footprint on the server to a minimum. There is no PHP,no CGI, or anything like that. On 9/19/06, shadowym [EMAIL PROTECTED] wrote: There is the underlying framework for developers to do their own thing but Digium has also made their own GUI. It's a GUI, a REAL GUI! it's in the FAQ's and press releases. In other words it's public knowledge. A GUI like FreePBX. In other words, a GUI! Did I mention it's a GUI! Not just a framework for a GUI but also an actual GUI. Did I mention that it is a GUI! A REAL GUI. It was on display at VON! A GUI as in point the mouse and click kind of thing. I believe that is called a GUI. It's graphical, and the user interfaces with it. They call that a GUI. From: mitcheloc [mailto:[EMAIL PROTECTED]] Sent: Tuesday, September 19, 2006 4:20 PM To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Digium GUI? No it's not, it's supposed to just be a framework for developers and resellersto create GUIs that can go on the appliance. On 9/19/06, shadowym [EMAIL PROTECTED] wrote: I am talking about the GUI that was announced as part of the new AsteriskAppliance.Sounds like it is going to be a full featured GUI like FreePBX. -Original Message-From: Noah Miller [mailto:[EMAIL PROTECTED]]Sent: Monday, September 18, 2006 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct.Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access?Do you mean this? http://svn.digium.com/view/asterisk/trunk/static-http/ On 9/18/06, Don Fanning [EMAIL PROTECTED] wrote: You mean the menuselect ncurses screen?If yes, then yes... it's a gui. :) -Original Message- From: shadowym [mailto: [EMAIL PROTECTED]] Sent: Monday, September 18, 2006 4:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Digium GUI? So the press announcement said that the new Digium GUI will be available in v1.4 sometime in Oct.Is the GUI already there in Trunk or is there some other branch of development that the general public cannot access? ___ --Bandwidth 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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mitchel ConstantinSnap - A desktop user interface for Asterisk www.snapanumber.com ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Mitchel ConstantinSnap - A desktop user interface for Asterisk www.snapanumber.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] linuxdevices.com: Trolltech woos developers with open Linux phone Who'll be the first with * on a mobile?
Forget about running Asterisk on it. It looks like it will be a good phone to use.I would buy one, especially in hopes that if we support a company like this then maybe they will produce phones like the Treo 650 or Motorola Q but in full open source fashion. On 8/19/06, Dinesh Nair [EMAIL PROTECTED] wrote: On 08/16/06 23:35 Robert Michel said the following: I think the BCM chip is for the GSM stuff, for GUI and applications the XScale chip - so for running asterisk, the XScale will be the processor. why would you want to run asterisk on the phone ? ideally, it should berunning a softphone and connecting back over WiFi or 3G (HSPDA ??) to anasterisk installation.--Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/+==oOO--(_)--OOo==+ | for a in past present future; do|| for b in clients employers associates relatives neighbours pets; do || echo The opinions here in no way reflect the opinions of my $a $b.| | done; done|+=+___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] VoipNow 1.2.0 Beta
On Sat, 29 Jul 2006, Tom Vile wrote: Did you look on the site? http://www.4psa.com/products/voipnow/demo.php Ughh.. it's PLESK! Looks like the entire thing is written w/ the PLESKuser API. NEXT! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST - Show quoted text - Instead of flaming, you could accept that not everyone makes software with just you in mind. ___ --Bandwidth 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] Manager interface
You could try out Snap -- www.snapanumber.com, it has the features you need. We also do custom developement, so this may help get your project moving along faster. On 7/27/06, Tielin Xu [EMAIL PROTECTED] wrote: There are many ways to do the screen pop, I'd like to do this way:1. Build the manager interface as an event server, which collect agentconnet events.2. Build a Java applet with the constant connection to the event server, each agent starts the Java applet at first task of each day3. The event server sends the connect info to the computer which theagent registed,4. The applet launch (pop up) the web based CRM application on agent computer with the caller's information5. Agent terminates the CRM application when the call is termianted.Tielin [EMAIL PROTECTED] 07/27/06 2:16 PM On 27 Jul 2006, at 11:47, Lee Archer wrote: This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it.I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on.There a number of ways to do this:1) run an application on each workstation which speaks themanagerprotocol andpops a screen as needed. This doesn't scale easily to large numbers, you need toinstall an application on each workstation and need some sort ofmanager proxyas asterisk does not like many manager connections.2) run an IM client on each workstation and have a central serverthat talksthe manager protocol to asterisk, sending messages to IM clients whennewcalls come in.3) have each user point their webbrowser at a web server whichtalksthe manager protocol to asteriskand have the webpage poll the server (using AJAX)4) embed a softphone in your application (or web page) and sendcalls to it. Configure the softphone to pop the screen when a callcomes in.We do 4) . which you chose depends on your needs/skills. Tim. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.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 Tim Pantonwww.mexuar.com___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Putting a call recording into a mailbox
This is a good idea, good use of technology. You should be able to do this by looking at the way voicemails are already being stored, just add the file in and make the txt file with the relevant information. Look in for hints: /var/spool/asterisk/voicemail/context/mailbox/ On 7/4/06, Marc Rohlfing [EMAIL PROTECTED] wrote: Hi, I don't understand what you are asking: what's the difference between sending out the email as an attachment so it ends up in a user's mailbox versus having it in the user's mailbox. Aren't they the same? Oops, my bad: I'm talking about the user's *voicemail* box here - should have been more precise there... Marc Rohlfing ___ --Bandwidth 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 is the best user GUI ?
Is Centile a solution built ontop of Asterisk? It looks similar according to their feature list. http://www.centile.com/solutions-intraswitch-platform-systemmanagement.php and http://www.centile.com/solutions-intraswitch-platform-advancedfeatures.php On 6/20/06, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to customise an end user application like Centiles's callpad software ( http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ? I've read www.voip-info.org User interfaces section ( http://www.voip-info.org/wiki/view/Asterisk+GUI). 23 softwares are listed. Which one is your favorite for that ? Why ? Cheers ___ --Bandwidth 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] voiceone?
Neil, I have not tried it yet, but I wanted to say this to those that don't realize it: VoiceOne is GPL http://www.voiceone.it/documentation/licence/ I just thought that was interesting... it doesn't look like it from the first look. On 6/20/06, Neil Adona [EMAIL PROTECTED] wrote: Hi! anyone from here, who uses voiceone as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help from here. That's all, thank you, Neil ___ --Bandwidth 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 Softphone on Thinclient?
I currently use NTAVO thin clients w/ Thinstation and I would love to put a soft phone on them, but I don't think that would work well (they use RDP), or do you all know if there is a smooth way to make the interface work? I don't really picture my users switching between an RDP session X-Windows (i.e. ALT-F3/ALT-F4) On 6/20/06, Idris AVCI [EMAIL PROTECTED] wrote: Hi Steve, We are running X-Lite on Wyse V90 terminals. They have Windows XP Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the onboard audio chip is very poor on voice quality. I guess X-Lite has Windows CE version. Check on www.counterpath.com. Idris -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 20, 2006 1:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SIP Softphone on Thinclient? Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth 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 Softphone on Thinclient?
Vitaly, That is good news, but I'm afraid that switching between screens will be a bit too much for my end users to handle. On 6/20/06, bails [EMAIL PROTECTED] wrote: Steve Totaro wrote: Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We have both kphone and xlite running on thinterms using LTSP nad running them as a local app, however it uses portaudio with OSS and i have noticed that different audio modules/soundcards give very different audio quality. eg CMIPCI = very good VIX82XX = very poor Bails ___ --Bandwidth 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] Transfer call via AMI or dialplan
If you want something ready to go I have implemented this in Snap. Here is a screen shot of the transfer feature: http://www.snapanumber.com/portals/0/transfer.png and website: http://www.snapanumber.com/ As well, you may also want to look at ADM (Asterisk Desktop Manager): http://adm.hamnett.org/ On 6/19/06, Moises Silva [EMAIL PROTECTED] wrote: This would be more like a blind transfer :) On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Thanks for the tip. No idea why I missed this. Off the top of your head, does this support attended xfer, or is it a blind xfer facility ? Julian. Moises Silva wrote: Piece of cake Julian: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect Regards On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: At the moment when one of our users wants to transfer a call, they press the transfer button on the phone, enter the extension and away they go. Is there any way to do this via the AMI or dialplan ? I want them to push a button on the screen rather than using the phone itself. Julian ___ --Bandwidth 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.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 ___ --Bandwidth 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] Easiest (best?) linux distribution for dedicatedAsterisk box?
You could also look into the official distribution from Digium called Pound Key. http://www.rpath.org/rbuilder/project/asterisk/ On 6/14/06, Mike Fedyk [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Wed, Jun 14, 2006 at 01:15:03PM -0700, shadowym wrote: FreePBX or AAH(aka trixbox) requires 256MB of RAM minimum to run properly. Just sitting there doing nothing on my test system it is using 170MB. How exactly do you meassure memory usage? E.g: on my laptop: $ free total used free sharedbuffers cached Mem:483056 476320 6736 070820 169360 -/+ buffers/cache: 236140 246916 Swap: 976744 3048 973696 Technically you could say that it only has 6.7 MB free. But actually some 220MB are used for buffers and caching by the kernel (because unused memory is wasted memory). BTW: where did you get the idea that Asterisk == FrePBX? Asterisk is not known to require MySQL and Apace to run. My home debian linux box runs, iptables, asterisk, apache2, exim, dovecot, dhcpd, proftpd and tftpd on a PPRO 200Mhz with 64MB ram for two users. Yep, it swaps a bit, (especially when running something big like emacs), but it does what it needs to do and it doesn't have a high load on it. The only thing I'm not running is mysql and astmanproxy. I doubt centos requires much more memory than debian. ___ --Bandwidth 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] 100 lines PBX + system config - repost
Hello Varun, Every system is different and simply suggesting a botherboard or cpu just isn't enough... You have two good options, your first is to do a lot of reading and research to determine your needs, and the best place to start is here: http://www.voip-info.org/wiki-Asterisk+hardware+recommendations The other option is to hire a consultant to help you build your system. If you have no experience with Asterisk, this will be the best place to start and have them guide you through it, that way you know your system is built correctly. On 6/14/06, varun [EMAIL PROTECTED] wrote: Hello, We are planning to biuld a 100 lines PBX based on asterisk. How do you decide on the system config, e.i motherboard, cpu , how much ram , etc ? We will have all 100 phone plugged in. But we expect about 20 calls at any given moment. Thanks Varun ___ --Bandwidth 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: Digium pound key software appliance opinions
Pound Key does seem to be a professional distribution, and considering Digium engineers work on it, I'd have to say that is pretty exciting. It doesn't include FreePBX, however it should be simple to install on top of. I'm looking into setting up a semi-production machine sometime next week to test it out. On 6/9/06, shadowym [EMAIL PROTECTED] wrote: So what are peoples thoughts about the new Digium software which appears to combines Asterisk, FreePBX(?), and Linux into one release to eliminate inter dependency issues and emphasizes stability. Seems like a much more professional way to go compared to Trixbox. Anyone using/testing it. I am curious what sort of opinions people have about it. http://www.rpath.org/rbuilder/project/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] click to call features on asterisk
You could check out Snap, there is a Firefox extension for it. You won't have to program webpages or anything as the phone numbers are automatically detected and handled without needing anything extra from the web designer. http://www.snapanumber.com On 6/9/06, Sharon Lim [EMAIL PROTECTED] wrote: Hi colin, I am doing on php. But i would glad that you can share the codes as i will explore it. Thanks. On 6/9/06, Colin Anderson [EMAIL PROTECTED] wrote: I have, using Active Server Pages + Flash. See: http://new.landmarkmasterbuilder.com and click on Contact Call Us Online. I can post the .asp and .fla somewhere if someone is interested in it. -Original Message- From: Sharon Lim [mailto:[EMAIL PROTECTED] Sent: Friday, June 09, 2006 6:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] click to call features on asterisk Hi there, anyone in the community has manage to configure click to call features? Care to share. I have tried on this manual , seem got some software error like http://www.voip-info.org/wiki/view/Asterisk+click+to+call Software error: Unable to determine call statusMessage: Originate with 'Exten' requires 'Context' and 'Priority' For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. Thanks in advance. ___ --Bandwidth 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] Integrating Asterisk
Just be sure that if you ditch your POTS line that you have a proper way to terminate 911 calls! On 6/3/06, Martin Joseph [EMAIL PROTECTED] wrote: On Jun 3, 2006, at 2:04 PM, Dakota Burns wrote: What I was attempting to visualize is the following case: 10 people in an organization pick-up their phones to make an outbound call. Before integrating Asterisk, all calls route through their current non-VoIP based phone provider. After integrating 1 trunk from a VoIP service provider into their system that provides 4 simultaneous calls (Teliax's Corporate plan), and dropping 4 legacy lines, if 10 people make calls simultaneously, some will be VoIP and some will be legacy based. Based on the above example, I'm questioning whether it would be best to configure a Sipura 3000 for every analog phone (I'm guessing the non-profits will want to keep their existing analog phones), or utilize another device (or devices) to connect the company's internet service into their existing Trunks or POTS. I think the former would be easier something I know how to do, but the latter may be smarter more cost effective. So the latter is what I'm questioning whether either of you have experience implementing. Personally I think it's better to get rid of the POTS lines and got to a real VoiP terminator. I am really an experimenter only, but my initial goal was to setup a way to share my existing PSTN line via an FXO like the wellgate 3701a. This turns out to be quite a bit of a problem due to crappy hardware (I started with the HT-488 but found it to be useless) and problems with my local loop (ie echo). Even with all the fussing I have done, I still have a very bad echo for the first few seconds of some calls, until the echo can. trains and knocks the echo out. Conversely, with Voip providers like Teliax (very good), Nufone.net (very good), I found that there are no such issues, and the most serious QUALITY issues are due to the routing of my data over the public internet to these companies. SO, in conclusion. Just because a particular Voip terminator is good, doesn't mean they will work well for you. Check the routes to them! Having said that, I found a third Voip call terminator that is very close to me (sellvoip.net), and have configured that as my primary terminator (asterisk will fail over to nufone and teliax if needed). This arrangement works great, allows for inward dialing, and is very cost efficient. If I had realized this to begin with, I would have skipped the whole PSTN aspect of my setup. Asterisk is SUPER flexible. you can set it up to route calls based on many criteria. For example, my setup routes 7 digit calls through my PSTN, because I already pay qwest 18$(us) per month, so these calls are free. If I dial 10 digits (US long distance) the calls are routed through sellvoip.net. If I dial an Israeli cell phone, the calls are routed through teliax (better rate). Hope this helps a bit. Marty ___ --Bandwidth 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] Simple windows / web Asterisk user software?
Hello Steve, Yes transfer is supported, when you are on a call just right click the status window and click transfer. Here is a screen shot of the feature in action: http://www.snapanumber.com/portals/0/transfer.png Take care, Mitchel Constantin On 5/29/06, Steve Totaro [EMAIL PROTECTED] wrote: Looks cool, thanks for posting this. Pro pricing is pretty decent too. Thanks, Steve Rod Bacon wrote: I liked the look of it, but the documentation didn't mention transfer capability. Does it do transfers? On Tuesday 30 May 2006 10:27, Paul Hales wrote: Have you given SNAP a go? http://www.snapanumber.com/Home/tabid/53/Default.aspx ___ --Bandwidth 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] Simple windows / web Asterisk user software?
Bruce, There are a few tricks to do what you are looking for, I'd need more details though on your situation, could you contact me off-list? Thanks, Mitchel On 5/30/06, Bruce Reeves [EMAIL PROTECTED] wrote: With SNAP can I write the configuration to a file and send it to a user? Or is the some way to deploy it in mass? On 5/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello Steve, Yes transfer is supported, when you are on a call just right click the status window and click transfer. Here is a screen shot of the feature in action: http://www.snapanumber.com/portals/0/transfer.png Take care, Mitchel Constantin On 5/29/06, Steve Totaro [EMAIL PROTECTED] wrote: Looks cool, thanks for posting this. Pro pricing is pretty decent too. Thanks, Steve Rod Bacon wrote: I liked the look of it, but the documentation didn't mention transfer capability. Does it do transfers? On Tuesday 30 May 2006 10:27, Paul Hales wrote: Have you given SNAP a go? http://www.snapanumber.com/Home/tabid/53/Default.aspx ___ --Bandwidth 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 -- Bruce Nortex Networks ___ --Bandwidth 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] Snap for Asterisk
Bartosz, When set up correctly the phone on your desk should ring and then when you pick it you will be connected to the number you dialed. This is all done via the origination command. Did you configure the Asterisk management interface both in Asterisk and Snap? The best approach to debugging is to log into Asterisk via asterisk -vr and watch what is happening when you try to dial. Best of luck to you, Mitchel On 4/11/06, Bartosz Piec [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: I've been working on a project for Asterisk for some time and it is finally ready for a beta release. Any feedback is well appreciated. At the basic core it's a Dialer for Windows. I'll be adding more features quickly, but I wanted to keep everything simple and stable in this first release. What is this for? I have set it up, trying to dial some number, a balloon tip says it is dialing but nothing happens. What am I doing wrong? -- Best regards, Bartosz Piec ___ --Bandwidth 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] Snap for Asterisk
Hopefully it's okay to *announce* this here. I've been working on a project for Asterisk for some time and it is finally ready for a beta release. Any feedback is well appreciated. At the basic core it's a Dialer for Windows. I'll be adding more features quickly, but I wanted to keep everything simple and stable in this first release. Check it out at http://www.snapanumber.com Thank you, Mitchel Constantin ___ --Bandwidth 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] Recommended rack-mountable server anyone?
Alexander, Perhaps I'm wrong, but I have a server here next to my desk (IBM e325) and I tried to fit a normal pci card into it. The slots are completely different and the card would not fit.. this was just a pci dvi video card. The server specs say that it is using PCI-X technology for the slots so this leads me to believe that they are not compatible as one would think. Cory Andrew, I will look into the supermicro servers again, I'm not keen on the handles up front on them though, that makes for awkward handling (imo). Wow Stagg, Thank you for that first hand knowledge. These are things you just can't learn until you buy a product and experience it first hand. I'm not so sure that we want to Frankenstein our own cable for this configuration though! (yikes!) Hopefully some other people will pipe up too with some more server suggestions! On 2/22/06, Stagg Shelton [EMAIL PROTECTED] wrote: We just installed asterisk for a customer using a Dell 2850. It has 3 pci slots. My customers configuration contained a TE411p Quad Span PRI, and a TDM400P with 4 FXS Modules. The only problem that we had with the 2850 was getting power to the TDM400P. We located a power connector on the backplane that supplied the required 12v. I think it was originally intended to power a tape backup drive. We ultimately sacraficed a power supply to get at it's 12V P4 connector. We then used a voltmeter to put together a pinout for the dell power port, and frankensteined together a cable that could be used to power the TDM400P.Aside from the power issue, the platform seems rock solid. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth 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] Recommended rack-mountable server anyone?
Yusuf, Could you find out the brand/make of the servers you used? That would be very helpful for me. Thank you! Stagg, Those servers sound like they should be avoided at all costs... Thank you for the heads up ;). On 2/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, we have used about 6 Intel rack mounted servers, with dual Xeon processors, (I have forgotten the exact make.) we used them with Digium quad span pri cards, and some with Sangoma cards, works well . These servers are still pretty solid. they come with two onboard NICs, and with the Digium cards, one of the NICS had to be disabled cause of interrupt clashes. Other wise they have worked well. I forgot about one other issue we had with the 2850. The integrated NICs caused interrupt issues with the TE411P. We had to disable the integrated NICs, and installed dual port gigabit intel NIC. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Alexander, Perhaps I'm wrong, but I have a server here next to my desk (IBM e325) and I tried to fit a normal pci card into it. The slots are completely different and the card would not fit.. this was just a pci dvi video card. The server specs say that it is using PCI-X technology for the slots so this leads me to believe that they are not compatible as one would think. Cory Andrew, I will look into the supermicro servers again, I'm not keen on the handles up front on them though, that makes for awkward handling (imo). Wow Stagg, Thank you for that first hand knowledge. These are things you just can't learn until you buy a product and experience it first hand. I'm not so sure that we want to Frankenstein our own cable for this configuration though! (yikes!) Hopefully some other people will pipe up too with some more server suggestions! On 2/22/06, Stagg Shelton [EMAIL PROTECTED] wrote: We just installed asterisk for a customer using a Dell 2850. It has 3 pci slots. My customers configuration contained a TE411p Quad Span PRI, and a TDM400P with 4 FXS Modules. The only problem that we had with the 2850 was getting power to the TDM400P. We located a power connector on the backplane that supplied the required 12v. I think it was originally intended to power a tape backup drive. We ultimately sacraficed a power supply to get at it's 12V P4 connector. We then used a voltmeter to put together a pinout for the dell power port, and frankensteined together a cable that could be used to power the TDM400P.Aside from the power issue, the platform seems rock solid. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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 message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth 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:
[Asterisk-Users] Recommended rack-mountable server anyone?
Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
