Re: [asterisk-users] two level administration tool for Asterisk
I worked on something like this about a year ago. It was a multi tenant web gui that virtualized pretty much everything from extensions to sip/iax users to voicemail. What I found that worked fairly well was to use a prefix for everything user specific with a format that could easily be parsed. I used a double underscore followed by a username followed by a double underscore for the prefix, it worked pretty well. I had a couple of functions that would add the prefix when writing the configuration to a database, and strip it when displaying everything on the web. I used a database schema for storing the configuration data and when it was updated it would write out the static asterisk files. The configuration files get a bit more difficult to read, but the upside is that users could define whatever names they want for things like extensions and mailboxes without any naming clashes. ___ --Bandwidth 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] Scalable IVR with asterisk
I'm looking at a project that is basically just an IVR, but will potentially be handling 20K or so calls per day, maybe even more. Any reason why asterisk would not work for this? I'm thinking in terms of being able to distribute calls across an asterisk server farm from some type of central termination/switching/proxy hardware, whatever that might be (Max TNT, etc..) I can't think of any inherent reason why this won't work. Anyone else? Chris ___ --Bandwidth 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 Manager and Ruby
On 11/1/06, Rajkumar S [EMAIL PROTECTED] wrote: Hi, Any one using Rubi asterisk manager interface http://rubyforge.org/projects/rami/ ? How stable/usable it is? It probably hasn't seen much use. I created that back when I was just learning ruby, so it could probably use some refactoring as well. And If anything has changed in the asterisk manager protocol that would be an issue also. I created it against the beta version at the time, can't remember what that was. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IVR Capacity
On 12/20/05, Serge Schumacher [EMAIL PROTECTED] wrote: With 2 quad E1 I can handle 240 channels at the same time, so those 240 channels have to playback voicemenus on the same time. The survey will have about 30 questions and the complete survey will take about 20-30 minutes Why not use a much simpler solution, such as finding a way to schedule different employees/offices/regions what have you to call in at different times? Or better yet don't use an IVR at all. Not that it's any of my business, but something just doesn't feel right about having employees slog through a 30 minute IVR. I can think of much better ways to get the information you want, at less cost, and probably less hassle for the employees. IMO the only good thing about this approach is that it might irritate enough people into giving you more honest answers about what they really think. If you are really brave, at the end of the survey ask them what they think about having to take the survey. Chris Chris ___ --Bandwidth 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] A2billing Trunk
On 12/16/05, MapsAir [EMAIL PROTECTED] wrote: Dear helpers and supporters! I have been playing with the A2billing for a week, but I still get stuck on creating a working trunk to terminate the call. Is any one show me how to setup a trunk in A2billing or pointing me to some where that I can find out the information. As Kevin has already said before, this is off topic and should be taken elsewhere. I think it would be a good idea to create a set of rules for the list and post them once a week or so. It's not really fair to expect a new user to have to search through old messages to find a posting from a month ago about what is or isn't acceptable, even if it is common sense. If on signup the rules were part of the signup email, and then reposted to the list every week or so, then there really wouldn't be any excuse for these off topic posts. I would bet that would cut down a lot on the number of off topic posts. Chris ___ --Bandwidth 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] sharing a line w/multiple extensions
On 12/15/05, Robert La Ferla [EMAIL PROTECTED] wrote: Is there a way for another extension to join a call in progress? i.e. If I can't share the line with all extensions, it would be nice to have a single button (dial sequence) that allows any extension to join the call. How can this be configured? Probably through some creative use of meetme. But IMO once you start using hacks like that, you should probably either look at alternatives or adapt to a different way of doing things. It's also possible that there is a solution that would work well for you, but as always it depends on the details. You might post your exact setup, hardware, number of phones, what type of environment they are used in, etc.. For example as someone else mentioned in a home environment you can do something similiar to what you want using the sipura SPA-3000. It's possible, depending on the details, that a similar solution might exist in your case. I kind of doubt it, but it can't hurt to try. Chris ___ --Bandwidth 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] mpg123 on x86_64 (Opteron MP)
I am trying to install AMP for Asterisk, which requires mpg123. Apparently, they say, mpg321 does not work with their setup. I would think that you could just edit musiconhold.conf after AMP is installed, and have it use something else like madplay. Madplay has worked very well for me so far. Chris ___ --Bandwidth 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] Synthesized Voice for Asterisk
On 12/9/05, Dakota [EMAIL PROTECTED] wrote: Are there any cool free software I can use to create automated voice message greetings for my PBX? I want to customized some of my messages, however prefer to use a standard voice. There is nothing out there that sounds good and is free. And if you want something you can integrate with an application like asterisk that runs on unix, it's even more expensive. Several hundred dollars minimum from what I have found. The festival tts engine is free, but the the quality leaves a lot to be desired. Definitly not something you would use in a business. Chris ___ --Bandwidth 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] audio lost on incoming calls
I've had this nagging problem ever since I started using asterisk and have yet to find the source of the problem or a solution. When making calls to DID's that I have from various providers that terminate into my asterisk box, The first 1-2 seconds of audio is lost. If it was a delay I could understand, but the first seconds of audio is just never heard at all. The only common item I can identify is that all of these calls go over the pstn at some point. When making pure voip calls the audio is just fine. For example calling in using a direct sip or iax2 url works fine, as does using my FWD accounts. It doesn't seem to matter how good the connection is to my provider. It happens equally whether asterisk is running at our data center at internap or from my dsl at home or at work. Whether nat is in the loop somewhere makes no difference either. Any ideas on what could be causing this, or how to go about debugging it? Chris ___ --Bandwidth 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] Best Switch for VOIP Applications
On 12/5/05, calvis [EMAIL PROTECTED] wrote: I need to replace my switch. Does anyone have any recommendations for a switch that is VoIP friendly? I want it to be a managed gigabyte switch. There are lots of brands out there, but would prefer some recommendations from the list. We use the Fastiron workgroup swiches and really like them. Very solid but a tad expensive. Chris ___ --Bandwidth 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] voicexml vendors
Any suggestions/comments on companies that provide hosted voicexml solutions that work with SIP? Seems like a new enough market that the pricing is pretty high and the number of vendors that will work with smaller volumes is low. So far Voxeo is at the top of my list. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can I get to a menu system while in a queue??
On 12/2/05, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, Is it possible to get to a menu system while in a call queue. Yes, set the context in queues.conf. Then put the extensions you want available to callers in the queue in that context. Chris ___ --Bandwidth 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] Fax Service
On 12/2/05, Rich Adamson [EMAIL PROTECTED] wrote: I know this isn't an * question, but... Does anyone know of a Fax over IP Provider? Client looking to dump 3 regular fax lines, and bring broadcast faxes in house (around 700 per day) in house. Their broadcast fax service is currently outsourced. Can anyone point me in the correct direction? Also, if anyone has some tips and pointers, that would be great as well. I've been using www.trustfax.com for over six months and they are very reliable. If you're talking about low volume, dig around their site and you'll find it is $9.95/year (800 number included) plus $.10/page. Lots of other higher volume plans as well. I second trustfax. Been using them for about six months myself and never had a problem. Chris ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two Phones - Same extension?
On 12/1/05, Mike McMullen [EMAIL PROTECTED] wrote: Hi All, I have an employee who works mostly in our office but maybe once or twice a week has to work from home to help care for her special needs child. As background we have AAH 2.0 running with 8 analog lines connected to two digium t400P cards. We have 10 sipura-841s as handsets in the office. I would like the employee to be able to make and take calls from her house when the she has to work from home. I'm leaning towards just installing s/w on her laptop with a headset for that setup. My question is how to handle setting her up so that she only has one extension shared between the office phone and her laptop. For this to work, do I need to unplug her phone from power/network in the office when she is at home or, hopefully, is there some other magic that can happen? What about just using call forwarding? Chris ___ --Bandwidth 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] PBX Manager beta1 release
The first beta of the PBX Manager can be downloaded from http://asterisk.paymentonline.net. Major new stuff includes expanded support for cdr reports and queues, as well as improvements in the dialplan scripts and several bug fixes. Windows is also supported in this release. PBX Manager has been tested on Debian linux, Freebsd, and Windows XP. The main features that we think distinguish PBX Manager are as follows: - Seamless multi tenant support - PBX Manager and Asterisk can run on different servers - Built in Asterisk manager library and proxy - Cross platform - Easy installation - Won't interfere with your existing configuration files. - Ability to create customized distributions for clients/customers. Cons: - It's new, doesn't have all the features that we would like yet, and it's not thoroughly tested. - It's not for someone that wants a complete asterisk installation, as it requires that you already have asterisk installed and running. - The templating and provisioning system isn't documented, and it needs to be if you want to have any hope of actually using that part of PBX Manager. - No support for Zap interfaces at the present. It's on it's way however. - No official support. This isn't our primary business, which is one of the reasons we decided to release this under a BSD license to begin with. PS. I've had a number of people asking about how to contribute. All the code is in subversion at the present. In the next few days I'll try to find a way to make the repository available so people can checkout the current release and make patches against it. Cheers, Chris Ochs Payment Online Inc http://www.paymentonline.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] Asterisk doesn't start
On 11/25/05, harry gaillac [EMAIL PROTECTED] wrote: Try to post your problem to asterisk-dev Hmm that seems to be your solution for just about everything doesn't it Harry? :) I think the problem is that asterisk-addons got built out of order or didn't get rebuilt at all, but I can't remember for sure. asterisk-addons has to be built after asterisk. I have 1.2 running on freebsd 5.4-STABLE and 4.11 without any issues. Chris ___ --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] global numbering plans
Thanks Bret. How difficult was it to compile this list? I'm assuming it's a compilation of pubically available data? I'd be interested in how much work it would be to keep this up to date? Chris ___ --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] Command line
I was wondering if you can use Asterisk from the command line to make it make an outgoing call and issue other commands whilst it's in the call? Sort of like when you use Minicom with a modem connected to a serial port and send it AT commands. I would suggest call files or the manager api, depending on what you are trying to do. Call files are probably easier to work with for quick and dirty jobs. Using the manager api lets you make the call and monitor it also from a separate program otuside of asterisk. Chris ___ --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] manager interface behavior
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote: I'm working on a manager client that I designed to hold open TCP connection to asterisk while it is running for varoius purposes. After being puzzled by unexpected behavior, I realized that the server closes the connection after it completes an originate action - or at least it does in the case of my test transactions. I solicit opinions: is this a feature or a bug? I've never seen that behavior and I've written several clients for the manager api. I guess it's possible that a particular combination of variables in the request could trigger an error that makes asterisk do that. I would try issuing the same originate by telneting in manually and see what happens. That way you can positively rule out your client being the one that's disconnecting. Chris ___ --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] Agent Logoff
On 11/22/05, Marcus Deluigi (intern) [EMAIL PROTECTED] wrote: That helped a little. Thanks a lot! Is there any chance to determine the agent id (defined in agents.conf) of a caller? If I'm understanding you correctly, you seem to be under the impression that you can only use RemoveQueueMember/AddQueueMember on agents that are defined in agents.conf? If so, there is no such relationship. The agent id you pass to these commands doesn't have to be defined anywhere.. Chris ___ --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] Re: manager interface behavior
On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote: snacktime wrote: On 11/23/05, Bill Michaelson [EMAIL PROTECTED] wrote: I'm working on a manager client that I designed to hold open TCP connection to asterisk while it is running for varoius purposes. After being puzzled by unexpected behavior, I realized that the server closes the connection after it completes an originate action - or at least it does in the case of my test transactions. I solicit opinions: is this a feature or a bug? I've never seen that behavior and I've written several clients for the manager api. I guess it's possible that a particular combination of variables in the request could trigger an error that makes asterisk do that. I would try issuing the same originate by telneting in manually and see what happens. That way you can positively rule out your client being the one that's disconnecting. to which I reply: That's the first thing I did, and it confirmed the behavior (see below). To be precise, the disconnect occurs after the Newchannel report. So I infer that you think it is inappropriate. I've recoded the client so that it immediately reconnects. Anybody actually tried this? I can imagine that the developer might have assumed that such a request would likely come from a transient client, and that it would be helpful to terminate the connection. But if so, I don't think it's the right decision. Maybe it's just an oversight. Any other opinions? I'm too lazy to read the server side code. Anything to see in the debug logs when you did the originate? I'd probably file this as a bug. I've never had the originate command make the server drop the connection like that, and I've never heard anywhere that it would be normal behavior. Chris ___ --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] equivalent to SetvarIf ?
On 11/21/05, Wilson Pickett [EMAIL PROTECTED] wrote:
Is there a syntax I can use to set a variable based on the evaluation
of an expression? I need something that will work in 1.0.9 and 1.2.
Isn't this what you're looking for:
set(VARIABLE=$[NULL${something}=NULL]})
I'm not quite sure I understand that. However using a regex works.
But I'm getting an error that I halfway understand and don't know how
to fix.
Set(something=800111)
This works:
Set(var2=$[${something} : ([1-9])])
This doesn't, giving me an 'invalid repetition count(s)' error:
Set(var2=$[${something} : ([1-9]{2,10})])
Anyone know what's wrong with my syntax?
Chris
___
--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] Bad Lines - What can the phone company do?
On 11/22/05, Justin Selleck [EMAIL PROTECTED] wrote: We suffer with some bad CO lines in the Seattle Redmond area. To compensate our gains have been tuned 10 rx and 2 tx. We have also had to add a 3 second wait to outgoing calls because many times the front of the number gets missed by the telco. Is there anything we can request from the phone company? They have checked our lines (probably just for tone) and say there is nothing wrong. Did they physically come out and run tests at the demarc? Verizon is usually pretty good about doing that if you ask. Verizon's front line support is clueless, but their onsite techs are pretty good. I had them come out to my place and they gave me all the stats on the line when they were there. Chris ___ --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] Bad Lines - What can the phone company do?
Also.. All that is required of the phone company is a minimum line quality, anything else is at their pleasure. And if you want to push them a little call up and enter the option to cancel your service. That is the *fastest* way to get to people who can actually do something for you as I found out. Those are the people who instantly scheduled a tech to come out and gave me some discounts also. They put their best people in the customer retention department. Of course that assumes there is a viable option in your area that you can hold over their head. Over in Kirkland Comcast offers full phone service, and for the couple of hundred a month I pay to Verizon they were more than willing to work with me to keep my business. Chris ___ --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
[Asterisk-Users] equivalent to SetvarIf ?
Is there a syntax I can use to set a variable based on the evaluation of an expression? I need something that will work in 1.0.9 and 1.2. Chris ___ --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
[Asterisk-Users] priority jumping
I'm pulling my hair out trying to figure out a way to write a dialplan that uses commands such as DBget that can jump,and make sure it will work with asterisk 1.0.9 and 1.2. And in the latter case also work regardless of what the priority jumping global option has been set to. The only command that I saw documented as supporting the 'j' option was Dial. Do the rest now support it also? If so that should give me what I need. If not any pointers? Chris ___ --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] priority jumping
On 11/21/05, BJ Weschke [EMAIL PROTECTED] wrote: On 11/21/05, snacktime [EMAIL PROTECTED] wrote: I'm pulling my hair out trying to figure out a way to write a dialplan that uses commands such as DBget that can jump,and make sure it will work with asterisk 1.0.9 and 1.2. And in the latter case also work regardless of what the priority jumping global option has been set to. The only command that I saw documented as supporting the 'j' option was Dial. Do the rest now support it also? If so that should give me what I need. If not any pointers? Yes. I'm happy to report that the behavior is now consistent across all apps for the 1.2 release. :) Great,thanks BJ! Chris ___ --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
[Asterisk-Users] standard extension with forwarding
I threw together this standard extension and would like some feedback
if there is a better way. I didn't want to use priority jumping, and
I needed it to handle calling outside numbers also without opening up
my whole outside dialing context to incoming callers. It's based on
the default standard extension macro in asterisk.
; Standard extension macro:
; ${ARG1} - Device(s) to ring
; ${ARG2} - Voicemail Box
exten = s,1,DBget(CFU=CF/${MACRO_EXTEN}/CFU)
exten = s,2,DBget(CFB=CF/${MACRO_EXTEN}/CFB)
exten = s,3,DBget(CFNA=CF/${MACRO_EXTEN}/CFNA)
exten = s,4,GotoIf($[${CFU} != ]?s-CFU|1)
exten = s,1,Set(DIALNUM=${ARG1})
exten = s,2,GotoIf($[${LEN(${DIALNUM})} = 7]?s-DIAL|3:s-DIAL|1)
exten = s-NOANSWER,1,GotoIf($[${CFNA} != ${NOANSWER} != 1]?s-CFNA|1)
exten = s-NOANSWER,2,Set(NOANSWER=1)
exten = s-NOANSWER,3,Voicemail(u${ARG2})
exten = s-NOANSWER,4,Hangup
exten = s-BUSY,1,GotoIf($[${CFB} != ${BUSY} != 1]?s-CFB|1)
exten = s-BUSY,2,Set(BUSY=1)
exten = s-BUSY,3,Voicemail(b${ARG2})
exten = s-BUSY,4,Hangup
exten = s-CFU,1,Set(DIALNUM=${CFU})
exten = s-CFU,2,GotoIf($[${LEN(${DIALNUM})} = 7]?s-DIAL|3:s-DIAL|1)
exten = s-CFB,1,Set(DIALNUM=${CFB})
exten = s-CFB,2,GotoIf($[${LEN(${DIALNUM})} = 7]?s-DIAL|3:s-DIAL|1)
exten = s-CFNA,1,Set(DIALNUM=${CFNA})
exten = s-CFNA,2,GotoIf($[${LEN(${DIALNUM})} = 7]?s-DIAL|3:s-DIAL|1)
exten = s-DIAL,1,Dial(Local/[EMAIL PROTECTED]/n,20)
exten = s-DIAL,2,Goto(s-${DIALSTATUS},1)
exten = s-DIAL,3,Dial(Local/[EMAIL PROTECTED]/n,20)
exten = s-DIAL,4,Goto(s-${DIALSTATUS},1)
exten = _s-.,1,Goto(s-NOANSWER,1)
___
--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] New asterisk management tool
On 11/18/05, Leif Neland [EMAIL PROTECTED] wrote: I need a hint:From pbxmanager/doc/INSTALL2.Install a database adaptor via rubygems.Postgresql, Mysql, and Sqlite3are all supported and tested to work.Eh... How to install? The syntax is 'gem install [adaptor]'. For Mysql the name is 'mysql', I think for Postgresql it's 'postgres', and for Sqlite3 it's 'ruby-sqlite3'. I'm working on a pdf installation guide which should be ready early next week. Until then just let me know if you run into any issues. Chris Leif___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
[Asterisk-Users] New asterisk management tool
Although I posted a demo a couple of weeks back, we have a new release of our management gui that has a lot more user friendly features and has gone through a bunch of testing. Still no name for it as it's mostly an internal project, but we will come up with something asap. Right now I believe it's ready for more input from the community. Before being read for beta testing we want to get some documentation out. For the brave there is a download at http://asterisk.paymentonline.net. Althought the web gui is fairly straight forward, underneath are some features we hope will be useful. Here is a basic feature rundown. * Transparent multi tenant support. * Template/Scripting system that allows a lot of different ways of laying out the dialplan and configuration menus. It doesn't lock you into using any particular layout, and it won't interfere with your existing configuration. Distributions can be easily created that have different layouts or menus. * Does not need to run on the same server as asterisk. You can run a simple distributed ruby proxy on asterisk which will proxy all requests to read/write files. * Uses ruby on rails which provides a good MVC structure for the code. Should be easier to modify then a typical php application. * Can use postgresql, mysql, or sqlite3. * Easy installation. * Built in asterisk manager client and proxy. * Separate web gui for voicemail users. * Runs on linux and bsd. It should also run on windows we just haven't had the time to test that yet. All of that said there are still a few things that need to be done. Queues, conferences, and call parking can't be managed yet, but that's simply because we left them to last. Should be another week before those get in. Zaptel configuration will probably come last, as we have no need for it but will add it anyways since others will probably want it. A large part of the system is the templating and scripting engine, which is not documented as of yet. We also need to add some more default scripts. The demo is pretty light but enough to give an idea of what you can do. You can view the demo at http://asterisk.paymentonline.net:3000. The user login is 'demo', password 'changeme'. The admin login is 'admin', password 'changeme'. Although it's easy enough to reset everything, deleting things in the admin interface will make the demo a lot less usable, so please be kind:) If you do want to install the distribution and play with it, it won't touch any of your asterisk configuration files so don't worry about that. You might need to email me about where to put the appropriate #include statements though, as I don't think that is covered in docs/INSTALL. And everything except for the Payment Online name and logo is BSD licensed. Chris ___ --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-Dev] Re: [Asterisk-Users] New asterisk management tool
On 11/17/05, Jan Saell [EMAIL PROTECTED] wrote: looks very nice but you get an error trying to loginto the voicemail system There seems to be a bug where if you were logged in as a user it gives that error once then clears up. We just put that up today so people would see that the voicemail login can be used outside of the management interface, but it hasn't been tested very well. Also the voicemail login is for asterisk voicemail users, not the 'demo' user. Chris ___ --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-Dev] Re: [Asterisk-Users] New asterisk management tool
Sorry but I had to disable the admin interface as people keep deleting the demo user. All the good stuff is in the user interface anyways, the admin interface is just to modify the scripts (which arent' documented yet) and for deleting users (which seems to be the 'in' thing to do). Chris ___ --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] New asterisk management tool
On 11/17/05, Henning Kilset Pedersen [EMAIL PROTECTED] wrote: tor, 17,.11.2005 kl. 00.24 -0800, skrev snacktime: Although I posted a demo a couple of weeks back, we have a new release of our management gui that has a lot more user friendly features and has gone through a bunch of testing.Still no name for it as it's mostly an internal project, but we will come up with something asap. Right now I believe it's ready for more input from the community. Before being read for beta testing we want to get some documentation out.For the brave there is a download at http://asterisk.paymentonline.net.This looks very, very sweet. We do all our inhouse development on Rails, and are rapidly falling in love with the technology. Looking forward toripping your code apart and making wise-ass comments about it ;p This is our second major project with rails. It's amazing how fast development goes with rails. So far this whole project is probably under 80 hours total. And the code will probably get some refactoring here and there also as we get more proficient with rails. Chris ___ --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] New asterisk management tool
On 11/17/05, Claudio Canseco [EMAIL PROTECTED] wrote: Hi, Congrats for the gui!, I was checking the voicemail butI get a crash erroron Internet explorerin the voicemail menu, when pressing mailboxes button after the displaying the contents for this option it crashes and then IE starts a crash error report and closes. I been able to reproduce it every time I tried the same. When doing the same on Firefox it does work right. I you want i cant post the dump file IE reports to MS. Well looks like that bug isn't entirely fixed. We had the same problem with the beta version of scriptaculous and downgrading to the stable version fixed it. Would you mind sending me the version of windows and MSIE you are using? Chris ___ --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] stop asterisk when Idle
But I found some situations that, after several millions of calls seconds,need to reboot the box and not only restart asterisk. That's really not necessary,and it's almost painful to watch people do this... If you posted some detailed information about your system and the problem you are having maybe someone could help you fix the actual problem. Chris ___ --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
[Asterisk-Users] multi tenant with queues
I'd like some feedback on my solution so far for using queues in a multi tenant configuration. For most of the configuration files I've been able to use a naming scheme for the context names, which works nicely for making multi tenant fairly transparent. However that won't work for everything and queues is one of them. In queues.conf the naming scheme will work for defining a queue. It won't work for the agents though as they all have to have unique names. My thought is to create a pool of available agent numbers, and the web gui for the tenants will let the tenant pick the agent numbers they want to assign out of the pool. As numbers are used they are taken out of the pool, and as they become available they go back into the pool. The downside to this is that a tenant won't get to pick the exact numbers they want, but that doesn't seem like too much of a compromise for a multi tenant system. Anyone have any better ideas? Chris ___ --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] multi tenant with queues
On 11/17/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I had the exact same dilemma and switched to using AddQueueMember/RemoveQueueMember instead of using agents. This solved my problem. Thanks!! That looks like a better solution all the way around. Chris ___ --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
[Asterisk-Users] SPA 3000 and MWI
My spa3000 is acting funny and I don't think it's an asterisk issue but thought someone might know what's going on. I reset the unit to factory configs yesterday and that's when it started. Whenever I have message waiting set to yes (in the user 1 config area of the sipura) I get the stuttered dialtone when the receiver is picked up even when no messages are waiting. Asterisk is sending the following to the spa300: Event: message-summary Content-Type: application/simple-message-summary Content-Length: 80 Messages-Waiting: no Message-Account: sip:asterisk@ Voice-Message: 0/0 (0/0) Any ideas what could cause this? Chris ___ --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] Asterisk realtime extensions context inclusion
On 11/13/05, Daniel Clark [EMAIL PROTECTED] wrote: Thanks for the reply, it's an approach I didn't think of to simply include the information from the other contexts into where I would be including from. In most cases that would work, but not in my case. Each user of my system will be able to place outgoing calls using their own sip connection (as in one they create with sipgate or vonage etc). To ensure that each user can dial out with their own sip connection and nobody else's they are each getting their own context and that context is the only place in the dialplan to dial that particular external sip connection. For a small amount of users it's possible to include all the information in each context, however I'm dealing with 15,000 users and would like a database small enough to fit on the hard disk!Even if your contexts are fairly good size, 15,000 of them is nothing as far as the space they will take up in the database or how it will effect query performance. Would it not be possible to do something with the Goto app? In each persons dialplan I can have an extension to catch internal numbers and then forward to another context using exten = 1,1,Goto(context2) or something like that? I don't think you can jump to a context itself, only an extension within a context. You might take a look at using agi or fastagi for outbound call routing. My gut tells me that a fastagi app connected to your own database schema would be a lot more efficient then using realtime. ___ --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] open asterisk?
This was just wrong, and an insult to Allison Smith IMO. What did you accomplish by all of this? Did you really think people would insinuate the same things you did and flock to your cause? If you have an ounce of decency you should apologize to those involved. Chris ___ --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] Voicepulse Open Access problems
On 11/13/05, Paul [EMAIL PROTECTED] wrote: I have 2 voicepulse open access numbers coming in over SIP. I use themfor some testing and at other times I just comment out the registerlines and let them go to the voicepulse mailboxes.I went to use them yesterday and they are not working. Calls go to the voicemail and if I enable their unavailability forwarding that works.Anyone else having this problem? I didn't change anything on my end. Itjust stopped working. Since then I have tried a few things but nothing helped so I reverted to the config that worked once upon a time. One of their gateways took a dive about a week or so ago. Look at the gateways they have listed and use the second one, that one works. Nice of them to send us out a notice though after being down for that long. I've talked to them on the phone and they were easy to get ahold of, but they don't seem to pay much attention to their website or to notifying customers of things we should know. Quality has always been pretty good though. You know one provider that has always been really proactive with this kind of stuff is Teliax. They consistantly send me email messages about any changes, and it's a nice way of letting customers know that someone is actually there. Just the other day I got a notice about an old gateway they were phasing out. It reminded me to check all my setups and sure enough I had one with the old gateway still in my system. Chris ___ --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] Advice on Asterisk-based home voicemail+fax+data system
Although I love asterisk, I've found that for a home setup I really like the SPA-3000. The dialplan in the 3000 sends out all local calls via the pstn and routes long distance to asterisk where I can use 2-3 different providers in case one goes down. The 3000 can also send an incoming pstn call to an asterisk extension after so many rings for your voicemail. Asterisk just handles voicemail and outgoing voip calls. I use to have a hylafax server, but switched to using trustfax about 6 months ago. For $10 a year and $.10 a fax it doesn't make any sense to be running my own fax server, especially since it comes with an 800 number, fax to email, pdf conversion, etc.. Chris ___ --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] Asterisk realtime extensions context inclusion
On 11/13/05, Daniel Clark [EMAIL PROTECTED] wrote: Hi I'm using asterisk realtime to control all of my extensions. As part of this I need to be able to dynamically create new contexts and extensions. The new contexts I create will also include existing contexts. Does anybody know the how to specify context inclusion for asterisk realtime as the database only has colums for id, context, exten, priority, app and appdata. You can't. Since those other contexts are in the database, why not just select them and then insert them into the newly created context? Or better yet dump realtime and generate extensions.conf from your own database schema. You could even use the realtime schema with just a couple of extra fields for things like include files, that way you dont' have to throw away the work you have already done. Asterisk doesn't handle database failures very well. Maybe it's been fixed now, but for instance a dialplan reload used to wipe out your whole dialplan if the database was down instead of just skipping the reload. I spent quite a bit of time writing an application for ARA at one point, only to toss it all out after seeing how it actually worked. I still think it's a good idea, and I don't mean to disparage those who put all the work into it, but it's implementation leaves something to be desired. Chris ___ --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] Asterisk Crashing (high load issues)
Have you tried it with a straight linux kernel from kernel.org ?? Whatversions?Have you tried it with a non-SMP kernel from kernel.org and/or yourdistro?Have you tried a nice, simple, distro like debian? IMHO, I found redhat,etc make too many customisations even to simple things like the kernel, so even when I used to use redhat, I always used my own kernel withoutany of their patches etc. One thing I always did was to not compileanything into the kernel unless it was needed for the system, andusually I'd disable module loading completely (though you can't do this with asterisk unfortunately). I second all of that. Much better to go with something like debian and start with a minimal install, then only add what you need. IMO Fedora on a production box is asking for trouble. Any distro that does a major release every 4-6 months is going to have problems now and then, and since you dont' need all the new stuff that goes into a distro like fedora, why run it? Stick with something that doesn't change a whole lot and is fairly conservative. Chris ___ --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
[Asterisk-Users] New asterisk web gui for small/medium sized businesses
I posted last week that I would get out a release asap, so here it is. Before I start in on putting up an actual website for it I thought I would put out a beta release to get things going. At this point there isn't a name for this project yet, as it's primarily an internal piece of software that we have been developing. For now I'll call it Asterisk web gui (AWG). AWG has a particular focus, which is to provide an easy to use interface for managing and monitoring asterisk, as well as a nice web interface for voicemail users as well. We are trying to make it as easy to install as possible. It plays nice with existing asterisk installations, and it won't overwrite any of your existing asterisk configuration. If you already have ruby installed on your system the setup time should be around 15 minutes. Once you have done one or two installations and know what the steps are, installation on a new system should average 5-10 minutes at the most. We have tested it on Freebsd and Debian. It should work on windows also. AWG is not intended to help you install asterisk and do your basic configuration. There are other software packages that do everything from start to finish such as AMP. AWG is dictatorial software. We will not include features by consensus unless they also fit our vision of what a tool like this should include. That said we want all the feedback we can get, particularly from businesses who are looking at it as a solution they might deploy for their clients. Just realize that it has a particular focus, and that's not going to change. There are also a few features not currently present that are on our todo list to get done asap. A basic interface for viewing CDR records, zap channel configuration, and a page to monitor real time information such as channels, peers, queues, etc.. There is a basic online demo at http://69.25.136.214:3000. The administrative login is user admin, password 'changeme'. The user login is username demo, password 'changeme'. At the moment there are not a lot of script templates installed, but the ones that are there will give you an idea of what you can do with the provisioning features. At the moment the demo is running in development mode where the errors are verbose and the code is reloaded for every page, so it's not as fast as you would see in a production environment. The setup guide and distribution file are at http://catalog1.paymentonline.com/~chris You can send any feedback to my directly at [EMAIL PROTECTED] or on the list. The whole thing is licensed under the BSD license. ___ --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] web management interface
Thanks everyone for the feedback on this. I'd say early next week for an alpha release. We have decided to release it under the BSD license, and it will go up on rubyforge after the first release. Chris ___ --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
[Asterisk-Users] web management interface
I'm finishing up a first version of a web interface for end users. It's focus is specific for our own uses, but I plan on releasing it under an open source license and would appreciate any feedback while I wrap up the first version. The interface is designed for end users without any real technical knowledge of asterisk except for some basic concepts of how things relate to each other. Such as contexts in a dialplan and how they relate to the context assigned to a sip/iax user, etc.. The interface is for day to day management of areas such as the dialplan and configuring new providers and phones in sip.conf and iax.conf. Things that an end user would want to change on their own. It also includes a nice voicemail interface for voicemail users, and some ability to manage/monitor asterisk via the manager api. One of the main features is the ability to write canned scripts that have associated configuration pages. A script is a text file with the script, and a YAML definition file. In the text file you can put variable placeholders, and in the YAML file you define the variables. The web interface then builds an html form based on the text file and the YAML definition. This way it's easy to add configurable sections in extensions.conf without having to change any of the base code. For instance providing canned scripts for extensions, call routing, voice menu's, etc.. If you have a script that needs a more custom web interface you can do that also by just creating the html form by hand. The same template approach is also used for configuring phones. Since we will be using this for local and remote installations, we also needed multi tenant capability. A basic multi tenant feature set is built in, so multiple businesses can be maintained on one copy of asterisk. Another requirement we had is to be able to coexist with an existing asterisk installation, instead of requring that the management interface take over all the asterisk config files. All you have to do with asterisk is add one include line in each .conf file you want to manage. And last but not least, another reason we couldn't use any of the existing interfaces is that almost without exception all of them were too difficult to install. Or more correctly unnecessarily difficult. We need to have something we can hand our clients and know they will be able to install the thing and run it with little difficulty. Since this interface uses ruby on rails, it includes a built in webserver, and the installation is a matter of untarring the distribution into a directory, changing the ownership of the directory to something asterisk can read, and running the start script to bring up the webserver. If we can work out a bug in tar2rubyscript that makes it fail on freebsd, then the distribution will be just one single executable that you can run as is. I would be very interested in hearing about what features people would like in a tool like this. Keeping in mind that it's not a complete asterisk system and is designed to work with existing installations. I will post a live demo in the next week or so once we get the first release ready. Chris ___ --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] web management interface
On 10/26/05, Dan Littlejohn [EMAIL PROTECTED] wrote: Chris:ARI has been recently expanded into this space for end userconfiguration.Don't know if you looked at it.http://www.littlejohnconsulting.com/?q=node/11 Thanks for posting this. I hadn't seen it up until now. Chris ___ --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] merchant account
On 10/22/05, Crystal Stream, Incorporated [EMAIL PROTECTED] wrote: You could have your customers call in and enter all ofthat -- then give them a confirmation number and theycould fill out the rest online. Couple of notes on this topic. First off, trixter's experience with the name being required is a special case. US processing networks don't even ask for the name, cant' do anything with it (I have most of the specs right here in front of me). If there is a name check it's done before being sent to the processing network. Internet payment gateways usually require a name, but it can be anything, no checking is done unless it's an extra feature you pay for, in which case don't use it:) Secondly, IMO the only real practical use for pay by phone is with an existing customer. If it's a new customer you usually want their name, address, email, etc.. But an existing customer could input their account number via DTMF which can then be used to pull up their information that is already in your system, and let you assign the new transaction to that customer record. Works well for paying bills or adding credit to prepaid accounts. Chris ___ --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] merchant account
On 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Thu, 2005-10-20 at 22:52 -0700, snacktime wrote: When you say software for the gateways, you mean you integrated with third party software that connected to one of the processing networks? Do you remember the name of the network that was used?I guess there could be a network that checks the name, but I'd have to see it to believe it. Vital, Firstdata, Global Payments, Paymentech, Nova.. None of those are capable of checking the cardholder name.And together they probably make up over 95% of all card processing in the US.I worked for trintech in 2000-2001, they write software for gateways andbanks.They directly connected to visa, mastercard, amex, etc.They wrote software to provide a variety of services, including softwaredirectly for card backers (note visa and mastercard dont actually issueany cards, its underlying banks that do, but they were some of ourcustomers). I do know that the majority of checks on address that are done are thenumeric portion of the street address and zip code only.So 123 mainstreet postal code 12345 matches 123 spring street in the same postal code.But the requirements that were passed to us from the major cardswere to do name checks as well. Ok that makes sense. There was an extra layer of checking done before the authorization was sent to the processing network. The bank mandated the extra checking which trintech had to perform. For the average internet gateway though you won't see that, unless you pay extra for it. Chris ___ --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] merchant account
On 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: I am interested in hearing some user experiences of anyone using amerchant account.The constraints are that everything entered must beDTMF-able.Card number, CCV, exp, numeric portion of the streetaddress, zipcode are all easy. name however is not so easy. The name is not used for card verification. Never has been. You might be using software or an internet gateway that requires it, but that information is not sent in the authorization request to the bank, and it's not required. I've certified to just about every bank network there is in the US, and I've never seen a specification yet that requires the cardholders name. And that being the case, it doesn't matter what you send in the name field. Chris ___ --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] merchant account
On 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Thu, 2005-10-20 at 22:29 -0400, Omar A. Sabek wrote: The CC merchant machines I've encountered require entry of the account number, exp date, total charge, etc. before dialing and transmitting the data. Even though we are able to pass DTMF successfully through the gateway, we still make the recommendation that any application that requires a negotiation phase (ie, fax machines, CC merchant machines, dial-up modem) remain on a traditional POTS line. And just like you mention, alternative methods are available including web access. Hopefully, the technological disconnect between voip and dial-up data transmission will come to pass sooner than later. I should have added one more requirement.Internet authentication likeso many do.Although all that I have worked with require the name to beentered as text, something that isnt trivial to do (and asking people to tap out their name via dtmf would be too bothersome).I would like a merchant solution that would work 100% off someonecalling into the system and entering their data, authentication willthen occur via the internet. Easy. Get with most any payment gateway and use their api to submit the transactions. The only information that you need from the customer you can get via dtmf. Everything else just put fill in the blanks with whatever you want. Card number, expiration, amount, cvv, and avs data can all be obtained via dtmf. The rest doesnt' matter. Just fill in whatever you want for the name, email, etc.. Now whether this is a good idea depends, but it's dead simple to implement. I did it a few months back in about an hour. Chris ___ --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] merchant account
On 10/20/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Thu, 2005-10-20 at 21:44 -0700, snacktime wrote: The name is not used for card verification.Never has been.You might be using software or an internet gateway that requires it, but that information is not sent in the authorization request to the bank, and it's not required.I've certified to just about every bank network there is in the US, and I've never seen a specification yet that requires the cardholders name. And that being the case,it doesn't matter what you send in the name field.Hmm..when I wrote code to directly talk to the banks we had to send itand the name would cause problems if it didnt match.This was not to agateway this was software for the gateways.And now with fraud concerns being higher I would assume its more critical.Keep in mind this is notfor transactions where the card is present, on those the physical aspectof the card is used (however secure that is supposed to be). When you say software for the gateways, you mean you integrated with third party software that connected to one of the processing networks? Do you remember the name of the network that was used? I guess there could be a network that checks the name, but I'd have to see it to believe it. Vital, Firstdata, Global Payments, Paymentech, Nova.. None of those are capable of checking the cardholder name. And together they probably make up over 95% of all card processing in the US. So even if there is some obscure processor that can check the name, it's not the norm. Now if you are connecting to someone below the major processing networks that adds on their own checks, that's a different story. However most internet gateways dont' check the name. Verisign doesn't, authorizenet doesn't, we don't. Chris ___ --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] more dids added to goiax.com
I like the web of trust idea. Chris ___ --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
[Asterisk-Users] central voicemail storage
I see you can use odbc for storing voicemail messages, but I'm curious about what others might have used for a central voicemail storage. This would be for 5-10 asterisk servers with a fair amount of voicemail. One thought is to just use scripts to pull or push voicemail via ftp to the central server, or to use a separate daemon on the asterisk servers to stick the voicemail into a central database. If possible I want to avoid asterisk using a database directly. I'd also like to avoid NFS. Basically I want a central voicemail solution that doesn't introduce any new points of failure into the system as a whole. Or is there a way to have one asterisk server handle all the voicemail for accounts on other asterisk servers? Chris ___ --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] Asterisk Manager API
On 10/18/05, Ezequiel A. Sculli [EMAIL PROTECTED] wrote: Hi group:I would like contact somebody who has experiences connecting anAsterisk-PBX with Manager API. Thanks. EzequielThat would be quite a few people on this list. Why don't you just ask your question here? Chris ___ --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: RE:[Asterisk-Users] free dids on goiax.com
On 10/18/05, Sergey Okhapkin [EMAIL PROTECTED] wrote: I completely agree. No reason to provide unlimited free service, put some reasonable restrictions like no more 10 different numbers could be called a day or no more than 20 calls a day. Be able to configure up to 5-10 numbers that you can call and no more than 10 or so calls per day total. That's more than the average person calls in a day anyways. And like you said it's free. And if someone complains show them the door. Chris ___ --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
[Asterisk-Users] Ruby module for the Asterisk Manager Interface
I have just released the first version of Rami, a ruby module for the Asterisk Manager Interface. It includes a client library and proxy server for sending multiple simultaneous requests with just one open connection to asterisk. One of the unique features is that the proxy server stores internal events into queues which can be retrieved or searched by value. For example with the Originate command, if you use it with Async, it will return immediately and the proxy server will store the associated events in the queue which can be queried at a later time. WIthout Async the Originate command will block until it is finished, returning all the events at once. Rami is distributed as a Ruby Gem. You can download it and view the documentation at http://rubyforge.org/projects/rami/. Chris ___ --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] res_perl - Compiling error
/usr/bin/ld: cannot find -lndbm Looks like you don't have the ndbm library installed. ___ --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] asterisk certification - thread hijack
The original poster's statement about not even receiving any proof that he was certified is kind of amazing. That's not a certification by any definition I know of. I would push Digium on that because they really don't have a leg to stand on if that is true. If they sold it as a certification then they owe you a certificate of some shape or form, and also something that say's what the certification covered. I wouldn't be too upset about it either because it is probably an honest mistake, but I would be firm on demanding that you get what you paid for. ___ --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] Re: www.openpbx.org
On 10/9/05, Florian Overkamp [EMAIL PROTECTED] wrote: snacktime wrote: permit to be used for their contributions..They won't be happy unless everyone else does things their way.They wouldn't be happy if asterisk was BSD or MIT licensed either. No that's not true. I myself would be perfectly happy with an MPL.However, because Asterisk is available under a GPL formed license, anyfork will need to be GPL too, until such a time that any and all GPLcode has been replaced by something the prospective owners are willing to relicense under something else.FLorian A fork can be anything you want if you own the copyright. They could fork asterisk into a BSD license tommorrow if they wanted to. Or actually it would be a combination of BSD and public domain, with all new code going under BSD. The people who don't like ABE wouldn't be happy with BSD, because what they dont' like is that Digium can take their contributions and release them as part of a closed source product. A BSD license would allow that just like owning the copyright does. Personally I look at it like this. Until the point comes that Digium is contributing less to asterisk then the rest of the community has, then the community is gaining more than they have given. If I contribute code to asterisk which Digium sells for a profit via ABE, that's great. I still have access to asterisk which is worth a lot more than the small part I have contributed. I win, Digium wins, I dont' see the problem. The code I contributed I probably needed anways. And without asterisk I wouldn't have anything at all. And in addition, the more money Digium makes the better asterisk will be which also benefits me in the long run. Chris ___ --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] Re: www.openpbx.org
I don't know, after looking at their roadmap I don't get it. It must be the asterisk commit policies that are driving this. They have some good ideas, but they are going about this the wrong way if their goal is to create a successful fork of asterisk. Chris ___ --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] Re: www.openpbx.org
On 10/8/05, Paul [EMAIL PROTECTED] wrote: Mike M wrote:On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote:Also consider that there are situations where 100% open source is neverallowed. Check out visa/mastercard processor certification for a good example. Digium dual licensing availability means I could actually standa chance of using asterisk as the basis for systems used by military andlaw enforcement in applications that require extremely high security. There is a popular vendor of closed source products whose security has beencompromised often. The security of OpenSSH is well established.Reading this list iwe learn that the open source version of Asterisk is currently being used by military personnel.Asterisk offers ways for users to implement eavesdropping applications whichundermines the goal of attaining extremely high security.Open source is for sharing if that's feasible and closed source is not. Dual licensing is for both.My point was not to argue that closed source enhances security. I wasjust pointing out that there are situations where the customer will notaccept open source. Credit card processing would be a good example. You could design *-basedsystems for both the client(merchant) and server(processor) functionsbut last I knew visa/mc would not certify open source solutions. Off topic but wanted to correct this.. Its not the software that has to be certified, it's the merchant (or payment processor). Ya you can pay a security auditor to look at your software and say that it's compliant, but it doesn't really mean anything. If you are a qualifying merchant or payment processor you would still have to go through the complete audit even if you used 'certified' software. Also, as a merchant you either have to go through the full audit yourself, or use a certified payment gateway. You cannot for example use 'certified' software as a merchant and connect directly to the bank networks without going through the full audit yourself at an average cost of around $20,000. Chris ___ --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] Re: www.openpbx.org
On 10/8/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: Tony Hoyle wrote: No, since Asterisk requires that copyright be assigned to Digium for all patches. Submitters to OpenPBX may be unwilling to do this, especially since that's one of the main reasons for its existance... Please stop spreading misinformation. We have addressed this at leastfour times in the last six months on this list.Digium does NOT require copyright assignment for contributions to theAsterisk tree. Digium does require either that the code be public domain (unrestricteduse), or that Digium be granted a license to reuse the code at ourdiscretion (the disclaimer). Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium can only put in their commercial version what they themselves have written, or what others have freely given them to use under the public domain. The only people that would have a problem with this are the one that believe so strongly in the GPL that it's the only license they will permit to be used for their contributions.. They won't be happy unless everyone else does things their way. They wouldn't be happy if asterisk was BSD or MIT licensed either. Chris ___ --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] Re: www.openpbx.org
On 10/8/05, Tony Hoyle [EMAIL PROTECTED] wrote: snacktime wrote: Being that Digium wants to be able to sell a commercial version, I don't see how they could have been more accomodating then this. Digium canThey could just use the GPL as is, since they chose the license in the first place.. they clearly have no issues with it.They already have the rights to use the code granted by the GPL - that'snot what the disclaimer is for.The disclaimer gives them the same rights as the owner so they can relicense the contributed code under a non-GPL license for commercialreasons.Not everyone is happy with that, clearly. I understand, that's why I said 'Being that Digium wants to be able to sell a commercial version'. TBH I'd rather digium had chosen something like BSD to start with andavoided all the GPL politics but the situation we have is the one we have. Agreed. ___ --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
[Asterisk-Users] 911 Notices
With the deadline coming up for sending notices to customers, I found it curious that out of 4-5 different providers I use, to date only one of them has contacted me. The rest don't even have anything on their website that I could find. Junction Networks was the only one that actually sent me a letter and also have everything right on the first page when you login to their system. A week or so ago I remember reading an article where the CEO from one of my vendors was complaining that they wouldn't have enough time to get all of their customers to respond in time. I thought that was pretty funny given that they don't seem to even be contacting anyone yet. There isn't even anything on their website except a statement that they do not plan to support 911 anytime soon. Am I missing something here? Is the FCC going to be extending deadlines and that's why the apparent lack of action on this issue? Just curious. I thought I would have started receiving letters a long time ago. Chris ___ --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: ARTCP Project (was RE: [Asterisk-Users] Features you'd like toseeina GUI?)
Why does the system have to be based on a linux distro? I think that's the wrong way to go. It's one thing to create a linux distro around a popular piece of software, but it's another to create software that can only be used as an entire linux distribution. If I were you I would take an existing application server platform of some type that is already popular, and build a management interface on top of that. Say something like Zope or mod perl/Mason. Preferrably you want a platform that has a good web application server and can also be extended using a good general purpose programming language like Perl or Python. Then after the core product is done add in secondary things like backups and monitoring. And although I know php is hugely popular and would make it easy for many to contribute, I would think twice about it. In real life it can get messy really quick on large projects, and it's not the best general purpose programming language. my favorite would be python, but that's just me. As for databases use an abstraction layer. Even if you aren't familiar with databases other than mysql, someone else will be. Some of these types of decisions will be what decides whether your project goes anywhere or not. And also, be prepared to do most of the work yourself with little help until a first usable version is produced. Lot's of people jump on the bandwagon once you get something going, but few will jump in and help a lot right from the start. If you don't have the time yourself, or can't put together at least a handful of people that do have the time, you are kind of doomed from the start. Everyone (like me) will be more than ready to give you their opinions on how to do things, but you won't see many of them when it comes time to actually do any work:) Good luck ___ 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] Features you'd like to see in a GUI?
On 8/3/05, brent clements [EMAIL PROTECTED] wrote: There are so many asterisk management guis out there,some good, some not. I would suggest doing something different. I think there is a big need for a opensource virtual hosted pbx interface. I think it would help out alot of the smaller ITSP's who are trying to get into the virtual hosted pbx market but don't have the money or resources to develop or purchase a commercial product. We paid somebody to do ours though, if there was an opensource version when we were getting into this, we'd probably use it and also contribute. That's just my two cents. You might take a look at http://asterisk.ochsnet.com. It's a project I started a couple of months back. It can manage virtual hosting under a single instance of asterisk by virtualizing all the configuration files, as well as managing multiple instances of asterisk. It hasn't seen a lot of testing outside of our own environment, but I'm happy to assist if there are any issues. It's BSD licensed. Chris ___ 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] Dialplan configuration with Realtime
On 7/6/05, Gundemarie Scholz [EMAIL PROTECTED] wrote: Hello! Following the instructions on voip-ip.org I have implemented Realtime with MySQL for my Asterisk server. The individual extension configuration is managed in a table called extensions. Still I have to keep some data in the extensions.conf, namely the switch and the include statements. Is there a way to minimize that or completely get rid of them? No, but you can put extensions.conf into mysql via realtime static while using realtime extensions at the same time. If your goal is to keep everything in the database that will work. Chris ___ 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] Voicemail = SMS
On 6/30/05, Mark Charlton [EMAIL PROTECTED] wrote: Hi I have been trying for a while to find a way to get an SMS send when I receive a voicemail into my asterisk system. I don't want to send an SMS if the caller doesn't leave a message. I have voicemail.conf set up to email and delete. Any and all suggestions will be greatly appreciated. The manager action MailboxCount gives the number of old and new messages in a mailbox. You would have to call the manager via an agi but it would give you the info you want. Chris ___ 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] Passing called number in SIP
On 6/27/05, Andres [EMAIL PROTECTED] wrote:
However, it's not really passing the called number per say. What
it's doing is putting the extension I have in my register statement
into the To field. I'm assuming the To field is actually being
populated with whatever * set the Contact field to when it
registered.This seems to mean that I need a unique username for
every SIP DID I have if I want to be able to route them to different
context's.
Is there a standard way of handling this issue when you have multiple
SIP DID's ?
Chris
Say you have a block of 100 DIDs with Level 3 for example. You can just
configure something like this in your incoming context:
exten = _30355597[0-9][0-9],1,Dial(SIP/${EXTEN},30)
I don't get it:) If I have 100 DID's but only one register statement,
isn't the called number for all 100 going to be the one extension name
I registered as? Or am I missing something? At least with the
provider I am testing with the called number is always the extension
in my register statement, regardless of what the DID really is.
For example I have 4 DID's with this one provider. With the following
register statement they will all come in with the sip user/called
number as 111222:
register = user:[EMAIL PROTECTED]/111222
With this register statement they all come in to sip user/called number s:
register = user:[EMAIL PROTECTED]
What happens if I put a register statement for every DID? Wouldn't that work?
Chris
___
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] realtime sip confusion
On 6/27/05, Steve Blair [EMAIL PROTECTED] wrote: snacktime wrote: In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip. You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Thanks for the note but why do you say it makes sense? If the username of the caller is used to identify a peer that seems really bad. If used this way then I'd have to define every number that is likely to call into my Asterisk box. Could you explain? It makes sense because it mirrors how sip.conf works, as opposed to doing something different. To me it looks like a limitation of SIP, whereas IAX was designed to work in a PBX environment. I'm not clear on a lot of this, but with the way SIP works I can't see an easy way to get the callerid and the called number without using some custom and/or little used headers. I would be very interested in hearing about how this is customarily done. Obviously the providers are getting that information from their upstream proxies, otherwise they wouldn't be able to route the calls. Why that information isn't passed downstream I don't know. Maybe it requires customization of * beyond what a provider wants to support? Chris ___ 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] Passing called number in SIP
I thought maybe one of the providers here could answer this question. When using IAX to make calls, it passes the called number. Being designed to work with a PBX this makes sense. SIP works differently though and I'm curious why providers don't have a way to pass the called number on their DID's, or choose not to do so. Providers that themselves use upstream SIP proxies are obviously getting both the callerid and the called number. However all incoming calls I get to my DID's have the caller id as the SIP user being called. I'm guessing that it takes using some additional SIP headers to get both called number and callerid, and that most providers probably don't want to have to support that for their clients. That's my very rough guess. Can anyone shed some light on the real reason? Chris ___ 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] LogWatch for Asterisk
On 6/27/05, Moises Silva [EMAIL PROTECTED] wrote: Im arriving to that step in my development. If nothing comes out earlier, just wait a couple of weeks and i will be glad of sharing the tools i use for monitor my Asterisk Service. Big brother works well. It's also easy to configure it to call you via the manager interface or .call files instead of using pagers. Chris ___ 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] Passing called number in SIP
On 6/27/05, Andres [EMAIL PROTECTED] wrote: snacktime wrote: I thought maybe one of the providers here could answer this question. When using IAX to make calls, it passes the called number. Being designed to work with a PBX this makes sense. SIP works differently though and I'm curious why providers don't have a way to pass the called number on their DID's, or choose not to do so. Providers that themselves use upstream SIP proxies are obviously getting both the callerid and the called number. However all incoming calls I get to my DID's have the caller id as the SIP user being called. I'm guessing that it takes using some additional SIP headers to get both called number and callerid, and that most providers probably don't want to have to support that for their clients. That's my very rough guess. Can anyone shed some light on the real reason? Chris, The SIP Message has both a From Header (caller id), and a To Header (called number). If you are not getting those properly then either you have broken Asterisk implementation (like the ones between 1.0.4 and 1.0.7), or your SIP provider is messing things up. To figure out who is at fault then use Etheral to capture the SIP INVITE directly before it hits your Asterisk and see if is is missing the To Header. (I really doubt it is). Strange now it's working with the called number being passed in the To field. I must have just not seen it before. However, it's not really passing the called number per say. What it's doing is putting the extension I have in my register statement into the To field. I'm assuming the To field is actually being populated with whatever * set the Contact field to when it registered.This seems to mean that I need a unique username for every SIP DID I have if I want to be able to route them to different context's. Is there a standard way of handling this issue when you have multiple SIP DID's ? Chris ___ 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] LiveVoip is Bankrupt
For those that paid by credit card, you can call your bank and get any amount they owe you refunded. You are not a creditor as far as the bankruptcy is concerned, the acquring bank is. Chris ___ 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] LiveVoip is Bankrupt
While some may be able to get credit card refunds (depending on a variety of factors, like how long ago they were charged, any court orders in place right now, etc - most banks wont give you a refund if they know they wont get any money from the merchant, unless you can prove fraud to some degree) there are more than likely more customers that will not. Doesn't work that way. Issuing banks are guaranteed payment by acquiring banks. It's the acquiring bank that has to eat the loss, not the issuing bank. Issuing banks eat losses when a cardholder defaults, but never when a merchant defaults. And in cases where the service is delivered over an extended period of time, the clock for when you can chargeback doesnt' start ticking until that time period is up. That's why acquirers don't like prepaid plans or extended length subscriptions. Someone like livevoip can charge a bunch of people and the acquiring bank can be eating losses over a year out. Chris ___ 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] Missing first second of voice on outgoing SIP/IAX calls
I've been curious about why this happens for the longest time. When I make an outgoing SIP/IAX call the first half second or so of the voice never makes it to me. This is consistant on every provider I have used except for voicepulse, and it always happens. With voicepulse it never happens. It doesn't seem to make any difference whether it's SIP or IAX. I don't really want to mention any names because this isn't really a gripe, it's just got me very curious. On these same providers I do not get the lost voice on incoming calls. Anyone have any ideas? The only thing I can think of is that maybe voicepulse isn't using asterisk? Chris ___ 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] realtime sip confusion
Trying to use realtime sip for the first time, and it's not working as expected. I have one user entry in the sip database. Everthing else is still in sip.conf. When I get an incoming call, this is the database query: SELECT * FROM ast_home_sip_realtime WHERE name = '+18003859169' The 800# is the caller id of the caller, which doesnt' make any sense to me. Is there any documentation about how realtime sip/iax actually work beyond just the schema's that are on the wiki? Chris ___ 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] realtime sip confusion
Chris Seems to me that your UA is sending that number as its SIP Username. You can look in /var/log/asterisk/debug for lots of RealTime info if using res_config_mysql. This was an incoming call via a DID. I can call from any phone and the query is always on the callerid. Part of my problem is I'm completely guessing on how sip realtime works, there is absolutely nothing I can find that say's 'this is what sip realtime does in a user/peer/friend context'. Also there is a bug where if a context has a dash, realtime splits the string on the dash and does two queries. I don't know why it's picking up the context's in the first place since I don't understand the logic. I do know that I have a couple of unique context names such as 'from-teliax' or 'voicepulse-out', and in mysql I see realtime making queries like the following: SELECT * from sip where name = 'from' SELECt * from sip where name = 'teliax' Chris ___ 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] Asterisk Zoom x5v 5565
On 6/24/05, Nazareno Pereyra Lima [EMAIL PROTECTED] wrote: Hi Geoff Manning, Yes my Zoom x5v its running with internet OK, but I would like not to use Global Village service for VOIP, that´s why I'm trying to find a way to use the Asterisk as my own Gatekeeper ( like Global Village service ) that will permit 2 or more Zoom x5v 5565 or Zoom IP Phone use VOIP in my Company. And for 2nd chooise, I don't know if it's possible to use the service of Voip between two Zoom (x5v 5565) without any service ( without Global Village or any gatekeeper ) . Because I can do this with the two CISCO ATA 186 ( don't know if this possible with Zoom). Thanks! ... Be expecting your response.. I never did figure out why, but when I used the x5v as a dsl router and tried to put an aterisk box behind it on a nat, every time I started asterisk the x5v would completely lockup. My guess is that zoom never anticipated another sip device being used in that manner. I contacted them but all their monkey's knew how to say was 'we only support global village'. I took it back and got a refund. Chris ___ 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] realtime sip confusion
In case someone else made the same mistake I did, and because I can't find this information posted anywhere, here is what I found out about realtime sip. You can use it to register UA's that are registering to asterisk, and you can use it for peer context's for outgoing calls, but you cannot use it for incoming calls from gateways you have registered with. I would have thought that when a call came in it would query either for the hostname of the gateway you registered with, or maybe the extension you registered as, but instead it looks up the username of the caller, which for incoming calls will usually be the caller id. It makes sense when you stop and think about it, but it's not exactly intuitive at first. Chris ___ 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] SIP DID routing
How do you get the called number on incoming SIP calls? I've never had multiple DID's via SIP from one provider before and somehow I never realized that with IAX it just works, and SIP is different. If I don't set an extension in the register command the incoming invite has sip:[EMAIL PROTECTED] in the To field. Now if I have multiple DID's that I want routed to different extensions, what's the solution? Is there a SIP header that is normally used to pass the called number in? Hope that makes sense.. Chris ___ 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] Question on bridged calls
Correct. However, you can probably guess that most sip/iax providers also use canreinvite=no anyway. Which is annoying to say the least, but understandable. Is this correct or am I completely missing something? You're also assuming that most itsp's use asterisk, and that is not a valid assumption. Yes I did. Do some just roll their own server using the iax libraries? Or is iax being supported by commercial platforms now? Chris ___ 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] Management: Reload performace Realtime performance ?
I was reading around in the mailing lists and people say reloading is stable. Now this tool has to manage 1000 clients so the conf files are quite big and reloading needs some time. What happens if a call comes in during that reload time ? How is the performance in general of the process described above (assumed the used hardware is not under- and not overdimensioned), can such a tool easily handle 1000 clients ? Does somebody use a similar tool with many clients ? This really depends on your usage patterns and a lot of other things like what database you are using and how it is configured, etc.. You probably need to list a lot more details about what it is you are trying to do before you will get an answer. I haven't had the time to test what happens when you reload a large static database, but I'm guessing it would load everything from the database first, then when it replaces what's in memory it only takes a second or so. Somewhere in the mailing lists someone said that the realtime uses many database queries. If there are also 1000 clients to manage, this should lead to lots of database queries. That's only for the realtime extensions. Chris ___ 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] Question on bridged calls
If I connect to a provider using iax, and that provider connects to his provider using only sip, the provider I am connecting to isn't going to be able to bridge the call and drop out of the media stream correct? If I'm understanding how bridging works, you lose the ability to have the media stream going directly between the two endpoints of the call with most of the providers out there if you use iax, unless the provider has their own tdm network. Is this correct or am I completely missing something? Chris ___ 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] 403 forbidden on SIP register
I'm getting 403 forbidden errors when attempting to register to a certain provider. I've tried just about every combination of configuration settings I can think of with no luck. Following is what I would think should work (and one of the settings I have tried). Rather then list every combinaton I've tried, what are the common causes of a 403 forbidden on a register attempt? Other providers work fine using the same syntax. And the provider that I am having sip registration problems with works fine with iax. register = username:[EMAIL PROTECTED]/provider-context [provider] type = friend context = provider-context host = host.provider.com username = username secret = password disallow = all allow = all ___ 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] 403 forbidden on SIP register
md5 instead of plaintext? Doesn't asterisk take care of this automatically with SIP? I have other providers that use md5 and they all respond with a 401 challenge and then asterisk generates the md5 and uses the realm given it in the 401. Also, I think I just seen a change in the last day or two that had something to do with 403's, and if I recall correctly, it also addressed upper/lower case something or another. Can you use ethereal or sip debug to determine the exact item that was sent that might be causing the 403? Either one should at least provide a hint. Here is a sip debug. All I get back is an immediate 403 forbidden. This is also what I get back on other providers if I had the wrong password. This is also cvs HEAD from yesterday, although jumping back to a version from a month ago didn't make any difference. Chris --- (10 headers 0 lines)--- Jun 21 15:42:52 NOTICE[34281]: chan_sip.c:4671 sip_reregister:-- Re-registration for [EMAIL PROTECTED] REGISTER 11 headers, 0 lines REGISTER attempt 1 to [EMAIL PROTECTED] Reliably Transmitting (no NAT) to 208.139.204.228:5060: REGISTER sip:voip-co1.teliax.com SIP/2.0 Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK027fe0e3 From: sip:[EMAIL PROTECTED];tag=as737624ba To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 -- SIP read from 208.139.204.228:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 69.25.136.31:5060;branch=z9hG4bK027fe0e3 From: sip:[EMAIL PROTECTED];tag=as737624ba To: sip:[EMAIL PROTECTED];tag=as3201eb44 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 ___ 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] 403 forbidden on SIP register
This should work, taken in account the username/pass are correct and the hostnome is the one they provided you as SIP registry server. What looks odd to me is the last two lines. Why are you first disallowing all codecs and in the next line allowing them all again ? You should either disallow=all and allow some codecs or just go with the allow=all without the disallow line. That was just a leftover of the different settings I had while testing. Normally it's disallow=all, allow=ulaw. Chris ___ 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] Call parking in multi user environment
On 6/16/05, C F [EMAIL PROTECTED] wrote: This should compile against HEAD, this also includes a priority +101 if the current parking spot is already in use. Compiles fine, thanks much!! Chris ___ 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] Help with Cron and Reload
On 6/15/05, Federico Alves [EMAIL PROTECTED] wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. Use the full path when calling asterisk. The cron environment is not like a standard shell in all respects. Chris ___ 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] Call parking in multi user environment
I'm looking for a solution for call parking in an environment where multiple users are hosted on a single instance of asterisk. The main issue being a way to keep user A from picking up calls parked by user B. I downloaded the supervaletparking code from asterlink which would appear to be a solution, but it doesn't compile on the latest CVS HEAD. I sent an email to the author just in case he wasn't aware that it stopped working at some point. I looked around and I couldn't find references to other solutions. Did I miss something or am I correct in assuming that the valetparking addons were the only available options? Chris ___ 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] -HEAD/--STABLE using 100% cpu
On 6/14/05, Kevin Bockman [EMAIL PROTECTED] wrote: Hello, I've been doing some testing lately on Asterisk. I've had some problems with it using 100% cpu at times. One time, it held the 100% cpu usage for 12 seconds. Are you sure it's asterisk using the cpu? Sounds like mpg123 to me. If it is mpg123, use madplayer instead and that will solve your problems. There are instructions on the wiki on how to configure it. Chris ___ 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] -HEAD/--STABLE using 100% cpu
On 6/14/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: and where would I be able to find madplayer? I'd be interested in testing it but cannot find it. Actually it's called madplay not madplayer, my mistake. There should be a package for it on most linux distributions. I know debian has it, and I know freebsd has a port. Chris ___ 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] dialplan appdata separators
In realtime extensions the pipe is the separator. And I see a number of commands that also use the pipe as a separator in regular extensions.conf. Can the pipe be used universally instead of a comma as a separator? Chris ___ 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] Re: Re: Digium Website Update: Asterisk Business Edition
No, but there was some talk about exactly what linking refers to. If you develop a 3rd party .so that asterisk loads, it does fall under the GPL; you can't make a wowie-gee CDR or call routing module and license it any way you please. That really depends. Generally the gpl works the other way around (when it's your code loading gpl libraries). Say I write a cdr module that doesn't use any asterisk code or header files with it's own interface and release it under the bsd license. Then in asterisk I load it and call it's functions. It's not under the gpl. And even if I release a module in a way that a copy does fall under the gpl, that doesn't stop me from releasing other copies under any damn license I want to. I could for example license it to a commercial vendor to use in their own voip software under a commercial license or the bsd license. What I can't do is take away your right to use the gpl copy I put out there. That is the ONLY thing I can't do. Chris ___ 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] Wildly inaccurate CDR records
On 6/11/05, Obelix [EMAIL PROTECTED] wrote: Quoting Obelix [EMAIL PROTECTED]: Is this question too difficult, or is it simply one that only a few users have experienced? I believe the forkcdr command is what you want, although I've never used it. Chris ___ 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] New version 1.013 of Asterisk VConfig
This is mostly a testing/bug fix release. Hopefully by the next version I will have some real documentation up on the site. Since it's primarily a platform rather than an end user system, without documentation it's not nearly as useful as it could be. http://asterisk.ochsnet.com Chris ___ 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] IP PHONE iareaphone x100, tested??
On 6/8/05, Jorge Ortega Perez [EMAIL PROTECTED] wrote: Good day, im gonna buy 2 IP phones to test Asterisk, i don't wanna spend too much $$$ on then, so i was looking at the internet and i read a lot, the cheapest are the Grandstream BudgetTone but some reviews of this list says they are not so good ... so i found iareaphones but i can't find reviews about them, i would like to know if someone has experience with them, at their site the phone seems to be done to work for Asterisk ... but im not gonna buy something without a good review ... I got one a few weeks back. It's cheap. It has a strange annoying dialtone, it doesn't have north american ring patterns that I could get working, and it has what sounds like bad feedback on the speakerphone that you can hear at both ends of the connection. I wouldn't have bought it if I had known. Chris ___ 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] New version of Asterisk VConfig
I repackaged everything into one distribution and cleaned up the installation. Should be a lot simpler to install now. Asterisk VConfig is a platform for virtual hosting of end users on a single instance of Asterisk using the realtime database structure. Right now the functionality of the web interface is limited to a direct configuration interface. As soon as Realtime SIP/IAX is done I will get to work on adding more features into the web interface. Chris ___ 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
