Re: [asterisk-users] snap a number now digium?
To: All Snap users and future ADA users
I hope you enjoy the product under Digium's leadership. During my time
working on Snap I learned a lot about the Asterisk community, building
software products, and meeting customer needs. From this experience, I
took Snap and made it as lightweight, stable, simple and easy to use
as possible. ADA, is that new version. It is a very promising start
for a serious desktop companion to Asterisk.
I wish Digium the best, and hope everyone benefits from the new change
in direction for Snap. I have since left Digium to focus on a new
company, Weavver™, with an initial focus in building a product around
Asterisk. The product is currently in a private beta and I'm looking
forward to launching it over the next few months. If you are a legacy
Snap customer and need any support (such as for a lost license key)
please feel free to e-mail our support team at support{at}weavver.com
for help.
If you want to keep track of the new company you can follow our
twitter feed at www.twitter.com/weavver.
Here's to a new year, new ventures, and a new president!
Thank you for using Snap and supporting it over the years,
Mitchel Constantin
p.s. Hope to see some of you in Miami @ ITEXPO and after that in
Orlando @ VoiceCon
On Thu, Jan 22, 2009 at 1:48 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Wed, 21 Jan 2009, John Todd wrote:
The SNAP dialer has just been renamed, and it's not only available for
AsteriskNow users - it's available for anyone using Asterisk, not just
AsteriskNow.
The website redirection is not ideal; I agree. We'll try to have it
pointed at a specific ADA page shortly, but for the moment the old
domain name goes to the digium.com page.
Here's a link which contains a location for download of the app, and
manuals:
http://forums.digium.com/viewtopic.php?t=66048
OK, Thanks.
Snap used to have a commercial version - is the full version now free, or
do we still have to buy licenses for the commercial version?
If it's not working as expected (i.e.: bugs) then you might want to
start a discussion of the specifics on the forum board
(http://forums.digium.com/viewforum.php?f=26 ) for comments.
Hate forums. Mailling list?
Gordon
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/
On Jan 21, 2009, at 2:47 PM, Dean Collins wrote:
So is it only available for AsteriskNow customers?
That's a bummer.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Mark Wiater
Sent: Wednesday, 21 January 2009 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] snap a number now digium?
isn't snapanumber turning into ADA?
http://dl1.digium.com/ADA/ADA_MIS.pdf
There's a forum too.
Looks an awful lot the same.
Can't get it to work with my Thunderbird contacts the way
snapanumber does though.
Maybe it's a work in progress?
Mark
Dean Collins wrote:
Wow they must have bought the company.
I use the software and love it - shame to see it disappear.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
+1-212-203-4357 New York
+61-2-9016-5642 (Sydney in-dial).
+44-20-3129-6001 (London in-dial).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Gordon Henderson
Sent: Wednesday, 21 January 2009 1:15 PM
To: Asterisk Users Mailing List Discussion
Subject: [asterisk-users] snap a number now digium?
Where's it gone?
Going to http://www.snapanumber.com/ goes directly to the digium
site
with
no indication of where it is ... Has it gone forever?
Gordon
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Re: [asterisk-users] Anyone use TACI.pl for a click to call app? - Doesn't seem to want to work for me
Hey Chris, You could try our Firefox plugin to acheive the same thing in an easier way (with no programming), it'll just work with whatever website you throw at it. -- www.snapanumber.com On 8/6/06, Christopher Aloi [EMAIL PROTECTED] wrote: Hello List -Asterisk Version: Asterisk SVN-branch-1.2-r38420MI'd like to create web GUI for basic internal outbound dialing.I came across TACI which, if I could get it to work, would fit the need. My goal is to provide the user a form with the following:OriginateUsers# _Number they wish to terminate to CallerID they wish to pass Context they wish to terminate through ___I'm sure there are many ways to accomplish this goal, I found this: http://www.azxws.com/asterisk/.The script seems to fail due to a missing priority, I added the extra space as outlined on voip-info, but the call still fails (doesn't even start) Debugging hasn't gotten me too far, Asterisk shows the app authenticate against the manager.conf file, then logoff.Anyone have this working? Anyone have another solution I may have missed?Thanks in advance, -- --Christopher T Aloi-- ___--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] Integrated T1
Yes it will support it, you should look up HDLC on the wiki...I went through this a year ago and had a hard time setting it up. It might be easier now though. I would recommend going another route and getting the data brought in seperately with it's own router. You'll also have better redundancy that way. Good luck, Mitchel On 10/12/05, Samy Antoun [EMAIL PROTECTED] wrote: Hi, We have a Data/Voice service supplied through an integrated T1. Does anyone know if Digium T1 card will support the splitting of the Voice and Data? Regards. __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Asterisk need in the way of a GUI?
We've collaborated, and are going to work on an advanced GUI client with a web interface to compliment it, it will be an all in one type of system. I would really appreciate feature requests on what you would like in a windows/linux form client. Please be creative =). We already have a very feature rich list planned! Thank you, Mitchel On 5/25/05, admin [EMAIL PROTECTED] wrote: Here are a couple of items I hear people asking for regularly. - Multi-tenant functionality - Allow users to change their own preferences via web (call forwarding, MoH, etc...) We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does Asterisk need in the way of a GUI?
We are two programmers who are passionate for Asterisk and we will be dedicating the next three months towards programming for Asterisk and would like to get some input from everyone on what they feel Asterisk is lacking or needs based on what is not currently a part of it or available through third parties. Hopefully, by asking up front we won't be wasting our time on something nobody wants or needs. Specifically I am asking in the way of GUI's (web-based or not), not in backend programming as Mark and others have that well under control! Thank you for your suggestions, Mitchel Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over
Matt, This isn't meant as a flame, rather I'm curious about what other people think about the following situation...maybe it's just the philosopher in me, what happens when the load balancer fails? Thanks, Mitchel On Tue, 29 Mar 2005 13:47:58 -0800, Matt [EMAIL PROTECTED] wrote: you can use dual T1, each on a separate pbx. and use a load balancer for fail over. see http://www.xgforce.com/loadbalancer.html for affordable models. Best Regards Matt - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 7:11 AM Subject: RE: [Asterisk-Users] Fail over No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forwarding calls
I think from what I remember you have to use agi to do this, so you can send the command once the call is bridged. I don't know how off the top of my head though but I do think this is the route to look at. mitchel On Wed, 30 Mar 2005 01:48:14 -0600, Paul [EMAIL PROTECTED] wrote: I have setup the menu system, it works fine, but I can't get it to forward the call to another outside number. The sites you gave me are on setting up the IVR. Any thoughts? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wilson Pickett Sent: Wednesday, March 30, 2005 00:19 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Forwarding calls connected to one of them. Basically my goal is to have someone call into the incoming POTS line and be presented with a menu where they would select an exten = 1,2,Goto,cellphone|s|1 Nice try, but take a look here: http://www.voip-info.org/wiki-Asterisk+tips+ivr+menu or here http://users.pandora.be/Asterisk-PBX/IVR.htm or here http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/d ocs-html/x720.html all of which were found using google interactive voice menu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Developing an IP Phone
I just thought this link might be interesting to some of you. I know it's m$ware but please hold back the flames. http://msdn.microsoft.com/library/en-us/wcetarget5/html/wce50oriDevelopingVoIPPhone.asp mitchel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco phones config over internet
This doesn't solve the clear text issue, but how about an access list based on the mac addresses? That'll secure tftpd a little more. mitchel On Mon, 31 Jan 2005 14:04:55 -0800 (PST), David Newman [EMAIL PROTECTED] wrote: On Mon, 31 Jan 2005, Gregory Junker wrote: There should not be any, except for the occasional rekeying. That's right. If you can, try capturing traffic on either side of the VPN tunnel endpoints to see what's creating all those packets. dn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice outgoing works - now tackle caller ID
I've tried it and didn't get it working, I also asked when calling their support and they said chances are relatively high that they won't let you do it if you ask nicely (to management) and that it is for sure disabled by default. mitchel On Tue, 11 Jan 2005 18:50:41 -0500, digium-list [EMAIL PROTECTED] wrote: My question was about setting the caller ID to be something other than the broadvoice phone number. That way if I want to put an 800 number on their display I can do so. I also need this because I am setting up incoming DID numbers to the pbx. I need to set caller ID to show the correct callback number. Otherwise I might as well stick to using vonage for outgoing calls. One thing I like about them so far is that they can set the caller ID name for you in the database. All I had to do was ask. The change shows up now in my web login. Maybe in a few days it will actually show up on phones I call. I can't get that feature from the other voip providers I am trying or from my cell provider. I have it on my pots line. If I set the callerid to my pots number when placing outbound calls through other providers, that number and my business name is displayed on the called party phone. Again, has anyone been able to set caller ID on outgoing broadvoice calls using asterisk? Daryll Strauss wrote: Outgoing caller-id seems to work fine. BroadVoice appears to send the name that is on the account and the phone number. My dial plan uses SetCallerID and SetCIDName, but the later is definitely ignored and the former may not actually be required. - |Daryll ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phones with two ethernet ports
I just wanted to point out something cool about the hub in the Cisco 7940 and 7960 (maybe others too). With the older SIP versions of the phones when you would restart them they would disconnect the workstation while they were rebooting, the newer SIP image on them uses a Universal Boot Loader that runs the image it downloads ontop of itself and thus can unload and reload it without restarting the phone and losing power to the workstation. This is very minor but somewhat annoying when your using your laptop plugged into the second port so you can use the lan connection to debug the phone. Mitchel On Tue, 04 Jan 2005 14:30:43 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: On Tue, 2005-01-04 at 10:08, Race Vanderdecken wrote: Each 4-pair wire has 8 wires in the blue wrapper cable. You only need 2 pairs, 4 of the wires, for 100MB Ethernet. You could split the wire at the wall jack and at the switch end where it goes through the punch down thingy (the name escapes me at the moment.) You only need to run 1 blue/4 pair wire cable to each desk. You could put a small hub on each desk to split our more sockets. For the life of me I will never understand why people believe that each cubical/desk needs it own 4-pair cable, CAT5 cable connect back to the server. There are limits to this idea of hubs, in that you can only cascade so many. If I remember rightly it's called the 5,4,3,2,1 rule. 5 segments, 4 hubs, 3 populated, 2 unpopulated, make 1 network. No doubt someone will correct me if I am wrong :) -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel ppp HDLC Receiver Overrun messages
I have the same issue, but I've been running this for several months on a faster (2.8ghz) computer with no loss to data or harm of any kind, so I would have to say it's probably safe to ignore. mitchel On Tue, 21 Dec 2004 16:37:17 -0600, Mark Farver [EMAIL PROTECTED] wrote: I have a pair of sites tied together with a T1 line running zaptel PPP on either end. The system works, but I keep getting these messages in the kernel logs, and users are reporting connection problems (TCP timeouts, and hangs) especially under high usage. --snip-- HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) HDLC Receiver overrun on channel WCT1/0/1 (master=WCT1/0/1) --snip-- My initial though was interrupts, the LAN was connected via eth0, which is sharing with the t1xxp card running the WAN link. (The t4xxp card is for a future project and is idle) but I downed eth0 and switched the LAN to eth1. /proc/interrupts: CPU0 0: 140133158 XT-PIC timer 1:125 XT-PIC keyboard 2: 0 XT-PIC cascade 9: 157178 XT-PIC eth1 10: 1401370796 XT-PIC t4xxp 11: 1408600199 XT-PIC t1xxp, eth0 14: 32528 XT-PIC ide0 15: 27273 XT-PIC ide1 NMI: 0 ERR: 0 My only other idea is the box is underpowered for the job. It is a P3-350 mhz... Any other suggestions? Mark Farver ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7940 help
You have to use tftp to do this, use www.voip-info.org it should have everything you need. Thanks, Mitchel On Tue, 30 Nov 2004 10:20:47 -0800, Michael Levenson [EMAIL PROTECTED] wrote: Does anyone have any simple documentation on converting a 7940 to SIP and making it function with *? I have been beating my head on a wall on this and have gone no where. Thanks Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
This is a great idea, I've got that phone and it really would be an amazing feature, I have a really nice asterisk system set up in my house and a $10/month broadvoice line, but linking everything together would definately be a really nice touch. I've got $15 towards a bounty ;). mitchel On Thu, 23 Sep 2004 19:54:53 +0200, Andy Powell [EMAIL PROTECTED] wrote: On 23/09/2004 at 13:36 Joe Antkowiak wrote: There are quite a number of positive (for end users) implications of doing this too... just think about all those cell providers that offer unlimited mobile to mobile calls, and then all those unlimited LD packages from landline and voip providers. This has huge potential for people who use their cell phones alot... Not to mention the fact that you wont be microwaving your brain... :D Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
Our prayersanswered? (http://www.phonelabs.com/prd_blue01.asp) mitchel On Thu, 23 Sep 2004 12:10:11 -0500, Jay Milk [EMAIL PROTECTED] wrote: When I installed my first home-PBX three years ago, I was looking at cellsockets -- devices which will accept certain cellular phones and provide an RJ11 jack, generating the ring-voltage and recognizing DTMF, which in turn makes your cell-phone look like a CO line. Pretty cool stuff, in theory, but it just didn't seem to be worth the cost, especially since it locks you to a particular cell-phone. Since then, I've moved to Asterisk. I looked at at cell-sockets again recently, but they haven't really gotten any cheaper... And on top of that, I'd now require a precious FXO interface for *. I looked at some developer documentation for my particular phone (S/E T610) while connecting it to my PC via Bluetooth. For those who are unaware, all GSM phones have a built-in set of AT modem commands. Not surprisingly, I was able to place calls as well as receive ring-indicators, caller-id information and call-progress information via the virtual serial port that the phone provides over bluetooth. But what's more, I was also able to utilize my PC as a handsfree speakerphone -- and all this over bluetooth. As I see it, all the pieces are available -- we got full phone control, some form of digital audio going back and forth, call-progress reporting. I know there's at least one bluetooth stack for linux, so *technically* we're there, no? I foresee a chan_blue which allow Asterisk to utilize a bluetooth/GSM cellular phone as a CO line, connecting by nothing more than a $5 bluetooth dongle and 5ft of air. Who's up for the challenge? If there's enough interest in the community, I'll be the first to add a bounty on this -- it would be worth at least $100 to me to have this functionality. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Channel Bank
Yes you can do it, I've done it with a T100P and an Adtran 612, if you need specific help let me know, look up adtran on the wiki for a similar example. Mitchel On Fri, 13 Aug 2004 20:16:20 -0300, Daniel Bichara [EMAIL PROTECTED] wrote: You can use VoiceTronix boards. Joe Pukepail wrote: Since it doesn't look like any of the FXS cards supported by asterisk support analog DID trunks, would it work if I used a T100P connected to an adtran channel bank (atlas 550?) with an FXS card installed? Anyone ever try this configuration? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using Cisco SIP Phones with Asterisk
Which hurdles are you talking about specifically? These phones work great with asterisk (as long as you install the SIP image on them). mitchel On Wed, 4 Aug 2004 15:57:11 -0400, Gary Carr [EMAIL PROTECTED] wrote: Are they still hurdles using Cisco phones with asterisk as mentioned at http://www.voip-info.org/wiki-Cisco+Phones ? We are looking for some cisco phones to test with. Gary ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Get MWI from Telco's voicemail
Scott, This may not be the point of your message, but just a side note, I believe that echocancel and echotraining are turned off when a fax is detected automatically regardless of whether or not they are enabled. mitchel On Wed, 04 Aug 2004 14:59:22 -0700, Scott Petersen [EMAIL PROTECTED] wrote: On Wed, Aug 04, 2004 at 04:37:32PM -0400, Seth Remington wrote: On Wed, 2004-08-04 at 14:21, Scott Petersen wrote: Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco. Maybe I am misunderstanding you but why does this force you to use telco voice mail instead of * voice mail? You can also free that third line up for voice if you use faxdetect. The concern from the client is that they want to have 2 people on the line plus a fax and still not give anyone a busy signal. Using asterisk voicemail does not allow this unless they pay for another line. Voicemail is a less expensive option ($10/month) than another line (~$50/month). I am looking at getting a DID from a VOIP provider to try and make the price point a little better but, being in Victoria,BC the options are non-existant at the moment. Vonage and Primus are the only two I have found that provide local (area code 250) DID's, but neither support integration with asterisk. I discussed faxdetect but, as they are a law firm, they live and die by the fax and never want a situation where they can't send or recieve a fax. As well, I couldn't figure out how to dynamically disable echocancel and echotraining on a line. My experience is that fax is less reliable with those enabled. Cheers Scott Petersen Xavier Solutions Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 backlight and list etiquette?
This sounds great in my opinion, I am looking forward to hearing more about it, as far as list etiquette it may be off topic but many of us do use the ciscos and it could be a great feature we can offer our clients. mitchel On Thu, 29 Jul 2004 09:28:09 +0200, Holger Schurig [EMAIL PROTECTED] wrote: I've taken apart a 7960 to fit a backlight to the LCD. Would others on the list be interested in this as a project when I've finished (i.e. should I document and photo all the stages)? It would be nice if you could make some pages on http://www.voip-info.org. You can upload and refer to pictures in the wiki :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple days on a GotoIfTime command?
I still don't see why you can't use a script and an array to simplify this, that way you don't have to work with extensions.conf, just work on your file, possible php and an array with a loop to check everything. -mitchel On Wed, 7 Jul 2004 20:49:48 -0400, William Suffill [EMAIL PROTECTED] wrote: well then lever it db driven and set the #'s in the db and update that to the proper call order as needed On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos [EMAIL PROTECTED] wrote: The problem is, there is no pattern. It´s not an open/close scenario. This month I need to call NUMBER1, NUMBER2 and NUMBER3 on those days. Next month, who knows? I´ll receive another schedule to implement on asterisk. I see no way to avoid changing those lines each month. What I´m trying to do is reduce the number os files involved. Gelson brian wrote: I see the pattern.. let me think for a second.. and I'm sure I can get you something that's simpler than 31 gotoif's bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of brian Sent: Tuesday, July 06, 2004 5:24 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] multiple days on a GotoIfTime command? You're making this WAY too complicated its simpler than you can even imagine. Mind answering my original question first? WHAT THE HECK is the pattern your logic? What times are you open.. what times are you closed? What? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roger Gulbranson Sent: Tuesday, July 06, 2004 4:20 PM To: [EMAIL PROTECTED] Cc: Roger Gulbranson Subject: Re: [Asterisk-Users] multiple days on a GotoIfTime command? On Tue, 2004-07-06 at 17:03, Gelson Dias Santos wrote: brian wrote: What are you trying to do? What is the end result and what hours are you open? Exactly what I said. Need to call a number if time and day matches what is on the rule. This month I have to: call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29 call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31 call NUMBER3 if day = 7,13,16,19,22,24,25,28 I have it working now using 31 GotoIfTime lines, one for each day of month but I would like to optimize it. If I could group all days related to a number somehow, I would end up with just three GotoIfTime lines. You are making this way too complicated. Use DBget to retrieve a number which is the extension you want and then dial that extension. Have a cron job (or something similar) set the extension you want via DBset. You can put all of your time logic into the cron job. There may be even simpler solutions. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple days on a GotoIfTime command?
Just an idea, I'm sure Asterisk supports an optimized version of what you want to do, I haven't delved into that area yet, but why not use an AGI script and some arrays to simplify everything? -mitchel On Tue, 06 Jul 2004 18:03:20 -0300, Gelson Dias Santos [EMAIL PROTECTED] wrote: brian wrote: What are you trying to do? What is the end result and what hours are you open? Exactly what I said. Need to call a number if time and day matches what is on the rule. This month I have to: call NUMBER1 if day = 1,2,3,4,5,8,14,17,18,20,23,26,29 call NUMBER2 if day = 6,9,10,11,12,15,21,27,30,31 call NUMBER3 if day = 7,13,16,19,22,24,25,28 I have it working now using 31 GotoIfTime lines, one for each day of month but I would like to optimize it. If I could group all days related to a number somehow, I would end up with just three GotoIfTime lines. Gelson bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gelson Dias Santos Sent: Tuesday, July 06, 2004 3:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] multiple days on a GotoIfTime command? I´m trying to setup a dial rule where I need to evaluate the day of month. Here is an example: exten = 4,1,GotoIfTime(16:01-07:59|*|14,17,18,20,23,26,29|jul?6) I found it doesn´t work. Is it possible to specify more than one day on the same line, or do I need to include one line for each day? I known I can use ranges but even then I´ll end up with around 25 lines for each month. I´m trying to simplify maintenance of this rules, because I´ll have to change it each month. Thanks for any tip/suggestion. Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie's doubt on sip.conf
Your probably going to get this url (www.voip-info.org) thrown at you by a few other people too...check there if you haven't already for more information. -mitchel On Wed, 7 Jul 2004 00:45:53 -0400 (EDT), kaiduan xie [EMAIL PROTECTED] wrote: Hi, I have some doubts on sip.conf. 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more than one sip phone, each with different FWD number? [fwd.pulver.com] type=friend secret=mypassword username=my fwd number host=fwd.pulver.com 3) Can anyone explain the meaning of peer, friend, user in more details? For each case, what is the role of Asterisk in SIP world, a UA, a proxy, or others? 4) If we only use SIP phone as extensions in Asterisk, the SIP phone doesnot associate with outside proxy, does Asterisk act as a proxy for inter-extension call between the SIP phones? In this case, for the outgoing call originating from SIP phone to other network, say, PSTN, does Asterisk act as a gateway? (PSTN connection with Asterisk is assumed.) Any comments are welcome, thanks, kaiduan __ Post your free ad now! http://personals.yahoo.ca ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing out of a voicemail message?
I believe this has already been done, it's the o extension. Check out the samples with the newest cvs of asterisk. That should answer any questions you have. -mitchel On Tue, 06 Jul 2004 08:27:42 -0500, Daniel Jimenez [EMAIL PROTECTED] wrote: Anyway to make hitting `0` during a voice mail dial an extension? The bosses used to have that feature and love it. Their VM prompt would say: Hello, My name is blah blah. I am currently unavailable. If you would like to speak to an operator press 0 now, otherwise leave me a message. Extension 0 exists, but dialing it during a VM prompt does nothing. Thanks, -- Daniel Jimenez djimenez[at]pobox[dot]com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: dialing # on a crisco (was: Divert to arbitrary number)
I do it like this:
exten = _*XXX, 4, VoicemailMain2(${EXTEN:1})
That way the user just dials * first then their extension number. The
only thing they are asked for is their password. This makes it easier
on the user so that they don't have to dial i.e. extension 800 then
dial in their mailbox number followed by their passwordone less
step.
On the cisco phone you need to set in the tftp configuration the
messages_uri setting to the number you want dialed..i.e. *123 for
extension 123.
messages_uri: *123
Mitchel
On Mon, 5 Jul 2004 18:46:49 -0700, Randy Bush [EMAIL PROTECTED] wrote:
Is it possible to have a speed dial on a cisco 7960 which dials the voice
mail number and then dials the extention and password so a user can
just push a single button to get their voicemail?
see Message Button under
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
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[Asterisk-Users] (no subject)
Robb, I wrote up a small tutorial on setting up the standard tftp server for linux check it out on my site. http://asterisk.titaniumsoft.net/ Mitchel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Robert Boardman Sent: Thursday, May 13, 2004 2:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] recommend a Linux based TFTP server Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb Do you Yahoo!?Yahoo! Movies - Buy advance tickets for 'Shrek 2'
RE: [Asterisk-Users] pattern matching w/ Cisco dialplans
I don't know specifically about your question, however you can do a MATCH="*" for all matches that don't match anything (no pun intended). Mitchel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Sent: Thursday, May 13, 2004 4:38 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] pattern matching w/ Cisco dialplans I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx- for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx- Currently my dialplan looks like so This DOES work - I can call LD using NuPhone and call local numbers that start w/ a 2 - however when I dial local numbers that start w/ a 2 I have to wait 10 seconds for the call to be initiated.. ie pressing 9xxx-, pause 10 seconds, initiate call. Looking over the SIPDefault.cnf I'm not finding a value that I can enter that would shorten this time. I'd like to have a pattern match in say 5 seconds as opposed to 10. Any ideas on how I can accomplish this? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Do you Yahoo!?Yahoo! Movies - Buy advance tickets for 'Shrek 2'
