Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))
On Mon, 3 Nov 2003, Steven Critchfield wrote: > So you bought that line of Marketecture didn't you. I think there are > several large open source projects that prove that C is maintainable. > Maintainability is really a function of organization. If you can't be > organized, you will not produce very maintainable C code. s/C code/code/g Java might force organization somewhat, but if you're not organized, you still can make the code look like a complete mess. -- JustThe.net Internet & New Media Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
On Wed, 15 Oct 2003, Uriel Carrasquilla wrote: > Paul: > in your opinion, which hardware SIP phone is the best price/performance > device after taking into account support costs? > Regards, > Uriel Wow... topic drift occurred *real* quick on this thread. :) I'd like to hear more people address the issue with the X-Lite. Believe it or not, my original post *was* about the X-Lite and not the Grandstream. In spite of the fact that I don't like it, it does work for just placing simple SIP->SIP and SIP->PSTN calls, and X-Lite, right now, doesn't. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec
On Wed, 15 Oct 2003, rnc Info Lists wrote: > Do you have a 100 or 101? You have indicated different models in your > postings. Were you able to get Call Transfer and Call Waiting working > with your Asterisk system and other phones? Which version of the > Grandstream firmware do you use? There most recent on their website this > weekend was at least 2 version numbers higher than what came on my phone > in August. Think that they are making improvements rather frequently. I have a 100. I probably said 101 without looking at the phone - the phone says 100. :) I should TFTP the latest firmware. But I do want to try to get X-Lite working. I'd rather use a softphone (limited desk space, four-port hub). -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
On Wed, 15 Oct 2003, WipeOut wrote: > Call transfer and call waiting do work, although the call waiting is a > little loud and anoyoing.. :) Yerright, call waiting works rather well actually; I meant three-way/ conferencing. Been a long week. Sorry :) -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
On Wed, 15 Oct 2003, WipeOut wrote: > Steven J. Sobol wrote: > > >X-Lite build 1079 consistently chokes no matter which codec I use - > >after five seconds I suddenly have no sound coming in and possibly no > >sound going out too. Putting the line I'm on on hold and then switching > >back to it gives me another five seconds of sound, then it dies, etc. > > > If you are not using a headset then X-Lite cuts the sound after a few > seconds.. Its probably the echo cancelation kicking in.. Connect a > headset and all should be fine.. Aside from the fact that that behavior is broken... :) I just purchased an Altec-Lansing combo headset/boom mic. Nice unit, but the same results from X-Lite. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
On Wed, 15 Oct 2003, Jon Pounder wrote: > >The Grandstream 101 I'm using is a piece of junk but I don't have the same > >problem with it. > > What don't you like about the grandstream ? (I am not looking to flame you, > but was considering buying and if there are problems would rather find out > beforehand) Nothing works. Call transfer and call waiting, in particular. (Well, almost nothing; vm notification does work) There is no place to plug in a headset, and since I do a fair amount of tech support and longish conference calls, that's a big deal for me. However, keep in mind that I have an old, no-longer-manufacturered model (the Budgetone 100). Don't take my frustration with my outdated phone as a sign that you should dismiss Grandstream out of hand - I just don't like my 100. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec
X-Lite build 1079 consistently chokes no matter which codec I use - after five seconds I suddenly have no sound coming in and possibly no sound going out too. Putting the line I'm on on hold and then switching back to it gives me another five seconds of sound, then it dies, etc. The Grandstream 101 I'm using is a piece of junk but I don't have the same problem with it. Not sure whether the issue is X-Lite or the Asterisk server. Has anyone else had this problem? -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI: getting the return code from an exec()'d application?
On Fri, 26 Sep 2003, Chee Foong wrote: > Hello Steven, > > I am planing to do the same thing: make dial return correct dial status and > use agi to detect it. > > Is it possible for you to share the modified dial source Oh, sure, getting ready to go to bed but can do it tomorrow. A diff would probably be the best thing to do, unless people have a problem with me posting the source as an attachment? > If i am not mistaken, result return by exec is like: > 200 Result= Yeah, but I'm using James's AGI perl module here. He tells me that the exec() return code SHOULD be the return code of the application that gets executed. That, unfortunately, isn't happening. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI: getting the return code from an exec()'d application?
So I hacked up the Dial app to return a numeric return code instead of changing contexts based on a number being busy or unanswered. The purpose for this modified dial app, which I call AGIDial, is to help me concoct a "follow-me" type of application. The app returns -1 for a completed call, 0 for unanswered, or 1 for busy. Well, I hooked the thing up to an AGI script that uses perl and AGI.pm, and ran some tests. The AGIDial app is definitely returning the right status codes and is able to differentiate between the three types of call termination. But the AGI script always reports a status code of 0. And I figured out why. $AGI->exec() seems to grab the return code of a Perl print() command which outputs the command to the server - but the return code of the print() is not what I want - the return code of the application is what I want. How do I exec an app through AGI and get *its* return code? -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does SIP work?
On Wed, 24 Sep 2003, Roger Schreiter wrote: > we have 2 snom phones running with sip. (Asterisk-0.5.0). > The sip part seems to be very stable. I've used * successfully with both the GS 100 and X-Lite build 1059 SIP phones. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
On Sat, 6 Sep 2003, Tom Forbes wrote: > I'm more curious to know what exactly it is about AGI scripting that > would make PHP an inappropriate choice. I love PHP, but I think Perl's superior string handling makes it a much better choice for AGI scripts. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Status Codes
Can anyone give me a pointer to descriptions of the status codes my Grandstream phone displays? I've looked on Google but can't find a definitive listing of SIP codes. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie IVR question
On Sun, 31 Aug 2003, Josh Edwards wrote: > > Are there any examples for ther psql or agi scriptscan I use php > with > agi You most certainly can, but I recommend something more efficient like c++ or perl, at least for any backend functions. That said, if you insist on using PHP for the backend or if you would like to try to use PHP to create some web-based tools for configuring *, I may be able to help. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More questions. Call Waiting and Threeway
I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking something (plus, I'm still not up to speed on the hardware/telco end of the setup - all of the work I'm doing is with software). Is there any way to control whether three-way and caller ID are enabled per-call or per-SIP-phone? What I'd like to do is, for example, be able to dial *70 from my SIP phone to turn off call waiting, or be able to enable three-way on a per-phone basis. I don't know what's on the other side of the Zap channels (i.e. PRI, CT-1, whatever), but if it makes a difference, I can find out. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any way to distinguish between...
a call on which caller ID is unavailable, and a call that's supposed to be private? As a side note, I have a phone on which I have caller ID blocked, but the Asterisk server still ends up getting caller ID from that line anyway. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Qwest CallerID question re:[Asterisk-Users] Caller ID problem
On Sat, 23 Aug 2003, Ryan Tucker wrote: > To check with a buttset, hook it up to the pair and monitor as you call in > from another phone. If it's DTMF, you'll hear DTMF. If it's FSK, you'll > hear a squawk much like a packet radio transmission or the data bursts you > hear on weather radio when a warning is issued immediately after the first > ring. -rt On my SBC phone, I used to hear a high-pitched chirp before the Call Waiting beep (much like the first chrip of a V.90 modem negotiation tone) when someone called in and I was on the line. Does this mean SBC was using FSK to transmit caller ID on my line? -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minnesota PUC: Phone rules apply to VoIP
On Thu, 21 Aug 2003 [EMAIL PROTECTED] wrote: > Interesting development in Minnesota PUC regulating VoIP: Not surprising. Ohio and several other states are mulling over similar rules. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP dialtone?
On Thu, 21 Aug 2003, John Todd wrote: > > 1) 911 service. Yes, that is one of three reasons to keep your PSTN > line. The other two reasons are: Inbound calls from local callers > still should work on a POTS line, for now. You can't find VOIP > providers in most area codes Packet8 looks like it has locals in 47 of the "lower 48" with local access coming to Hawaii at some point in the near future. I checked - in some states like Ohio they only hit the big cities - on the other hand, out here in California I can even get a number local to my house because they have Victorville exchanges (and while there is a relatively large population here, we're still mostly rural and we are in the middle of the Mojave Desert :) > trivially: there _must_ be a database of address-to-PSAP mappings. > Any PBX administrator (or SIP phone owner, for that matter) should be > able to figure out their address. Methods for associating the PSAP > number with the phone are numerous, and trivially implemented Yeahbut what if I am connected to my VOIP account from my house, and then I go on vacation and take my phone with me? I don't think the 911 obstacle is impossible to overcome, but there are some issues that still need to be addressed. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unregister SIP connection?
Is there a way to make * forget that SIP phone [EMAIL PROTECTED] is registered? I ask because I have a few different PSTN numbers that I use for various reasons, and I can reprogram my Grandstream, but unless I also restart *, calls to the originally-registered number still ring through, and calls to the number that I am trying to switch to do not work. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] So now I'm playing around with Queues....
On Wed, 6 Aug 2003, Steven J. Sobol wrote: > and I found a reference to an AgentLogin.rtf. Looks great, except I can't > get it to work. Mysteriously enough, now music on hold AND queues work. Well, MOH works. Queueing works, almost. I can dial in from outside and be placed in queue. If I log in to my Grandstream, my call from outside comes off hold, and I can talk through the outside line and hear it on the Grandstream - but sound doesn't seem to work the other way. IOW, no voice traffic is making it from the Grandstream to my cell phone (which is the phone I generally use to make the call in from outside). -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] So now I'm playing around with Queues....
and I found a reference to an AgentLogin.rtf. Looks great, except I can't get it to work. queues.conf: [sjs-testq] music = default timeout = 1 retry = 1 maxlen = 0 member => Agent/10001 agents.conf: agent => 10001,1234,Steve Sobol extensions.conf: (I have a phone line set up on which the main menu tells you to press 1 to be added to queue. Pressing 1 lands you here) exten => 1,1,playback(auth-thankyou) exten => 1,2,queue,sjs-testq <<--- I don't think this is right! exten => 1,3,goto(1,1) (From the agent's SIP phone) [sjs-agentlogin] exten => *46,1,wait,1 exten => *46,2,AgentLogin(10001) Dialing *46 gets me the password prompt. I enter the password, but the call is disconnected a few seconds later. Dialing 1 from outside seems to put me into the queue, but without any type of hold music... ok, auth-thankyou isn't music, it's a voice clip, but I was figuring it would just play over and over until a hangup or until the queue was picked up. I set this up based on the stuff I found in the agentlogin.rtf file, but there's not a whole lot of information in that file, nor could I find much on the web... -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
On Tue, 5 Aug 2003, McAughan, Matt wrote: > Does anyone keep a known telemarketer caller id database? Here it is: CALLER UNKNOWN PRIVATE :) Most CLID's come up Unknown. -- JustThe.net Internet & Multimedia Services 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?
On 1 Aug 2003, Steven Critchfield wrote: > Something that may need to be thought out, we seem to run into this > database interface problem regularly enough. Is it time that the > database access get moved to a resource and then the extensions > app_sql_postgres.c could be made into just a sql app that works with any > backend database. 1. Define an API providing whatever functions you might need. There probably aren't many - open DB, query, close DB, report last error, what else? Figure out which functions you need, and write prototypes for them. 2. For each type of database, write and compile the functions defined in step 1 into a shared library. 3. Allow the user to choose which library they need, and dl() it at runtime. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PHP API for Manager - Plaintext auth needed?
Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager
At 12:13 PM 7/31/2003 -0500, you wrote: Does anyone see any potential problem with eventually linking in ssl in such a way as to allow connections to be made securely? I do not know the amount of work this would entail, but this would potentially allow for our IAX traffic to be encrypted too. If we're using stunnel to tunnel unencrypted connections through an SSL pipe, I'm not sure any rewriting would have to be done on Asterisk's end. Clients might have to be re-coded, though but for IAX/IAX and Manager connections, I don't think a lot of work would be required. Again, I have to play around with stunnel and asterisk before I can confirm any such assumptions. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 **NEW ADDRESS** Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager
On Thu, 31 Jul 2003, Dan wrote: > Hi Roy, > > It is not much safer to use SSH to connect to the computer and then > 'asterisk -r' to the asterisk console? I personally would think so. I believe the Manager interface is supposed to be an interface through which a remote control-panel type program or web page could connect to an Asterisk server. It's a straight, unencrypted TCP connection. However, does anyone see any reason we couldn't use stunnel on manager-port connections? I've been thinking about trying this out, but have not had the time to actually do it. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager.pm port
On 31 Jul 2003, Steven Critchfield wrote: > If you are running the manager from the webpage, then I can remotely > understand php manager interface. But if you plan on making a command > line manager app, then please do yourself a favor and just help with the > perl stuff. Remember php is perl -1 or more revisions. I do say this as > a php programmer for my work. We don't let php infect our backend > console apps. PHP is cool, but I agree with you that it makes absolutely no sense for commandline applications unless you need some functionality you can't get with another scripting language -- and that's not likely if you have Perl installed. No, I'm doing it strictly for use on web pages. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager.pm port
For anyone that cares... I am porting James Golovich's Manager.pm over to PHP. I plan on also doing some documentation which will cover both the Perl and PHP APIs, which will be almost identical (at least, to whatever extent is practical). Will let y'all know when I have some usable code to show you. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk
On Wed, 30 Jul 2003, Brian West wrote: > Same here. Same build. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
On Wed, 30 Jul 2003, Steven J. Sobol wrote: > I've successfully used the FreeTDS libraries on a Linux box to connect to > a MySQL server s/MySQL/MS SQL/g -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Microsoft SQL
On Wed, 30 Jul 2003, Florian Overkamp wrote: > As suggested by another poster: MS-SQL is mostly based on Sybase, so any > Sybase driver (there is one for PHP for instance) can probably be used, > from AGI or otherwise... I've successfully used the FreeTDS libraries on a Linux box to connect to a MySQL server. The original MS SQL was based on Sybase SQL Server 4.2, but for SQL 7 or SQL 2000, you'll want to compile the FreeTDS library with version 7.0 support (you can either do 7.0 or 4.2). -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk user guide ..
At 01:58 PM 7/28/2003 -0400, you wrote: Jeremy, While I see your point, I don't think it's reasonable to ask an end user (as opposed to a system admin) to hang out on IRC to learn how to use his/her phone while dealing with live calls and trying to do their job (sales, marketing, support, accounting, business admin, etc). I've spoken with Jeremy and he's knowledgeable and usually pretty helpful, but I agree with Troy's point here - sometimes it's not practical to hang out on IRC. To address the other issue - why is there not an end-user guide? Documentation would depend on the way you personally (as the system admin) set up the Asterisk box. You may have people dialing 9 for an outside line, while another person may have people dialing 8, etc. You made this point later in your e-mail. That is the problem. You can't guarantee that two people will set their PBXen up exactly the same way. There are some components that basically work the same regardless of how they're configured. Voicemail, for example. But the general PBX functions depend entirely on your setup. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 **NEW ADDRESS** Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best software SIP client
At 10:31 AM 7/26/2003 +0200, you wrote: I had the same kind of problem until I upgraded my asterisk-0.4.0 to latest CVS. Then X-Lite kind of worked : I could hear and the announcements and let a message in my voicemail. But DTMF doesn't seem to work : I tried to log in my voicemail to hear my messages, "dialed" the good "password", but asterisk never seemed to receive what I typed, specifically, 0 doesn't seem to work well. DTMF works for me, but from softphones it seems to take a second or two for the voicemail system to respond to the keypress. This doesn't seem to happen when calling, for example, into the voicemail from my cell phone. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 **NEW ADDRESS** Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Best software SIP client
On Fri, 25 Jul 2003, Dave Packham wrote: > Fixed it I have audio now... uninstall everything xten makes and > manually clear out all the xten/xlite stuff from the registry.. search > for XtenNetwork and kill the keys. reinstall Xpro and it works... go > figure For what it's worth, I was having lots of problems until I did just this... but I think it's because I screwed up configuration of X-Lite. But an uninstall/registry clean followed by a reinstall and a proper configuration of all of the relevant settings worked. I'm now doing inbound and outbound calls through our Asterisk using X-Lite (the most recent build, which I believe is 1050). -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI.pm?
I've seen references to this module in the mailing list archives, but it isn't in the 0.4.0 tarball, nor is it in CVS. I can roll my own and was planning to do so anyhow, but that doesn't seem to make a lot of sense if it already exists. Am I not looking somewhere I should be looking? Most of the Google hits just point to the mailing list. -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
On Tue, 22 Jul 2003, Jeremy McNamara wrote: > Karl (klasstek) and myself (mainly Karl) has spent a few clock cycles > figuring out how to make dynamic extensions happen, but we had no real > motivation to finish the task. Well, I'd certainly be willing to pick up the project from you. I think it should be done in the core, rather than in a module, but that's just my observation, and I've only taken a very cursory look at certain parts of the Asterisk source. > Find either one of us on IRC or search the mailing list archives. > Yes, sir. -- Steven J. Sobol, Geek In Charge, JustThe.net "Microsoft must think they're a navy, they open so many ports." --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamically setting up/tearing down extensions
Hey! On 21 Jul 2003, Armand A. Verstappen wrote: > I've been lurking on the list for a few months now. > > > I'm looking at DynExtenDB (and have played with it). I love that it reads > > the dialplans out of a MySQL database - that is a critical issue for me. > > But it has some issues. > > I haven't found this DynExtenDB however. Could you provide me with some > pointers to it? http://andreasotto.net/asterisk/ > PS: We never finished the Aegir Addon stuff. Maybe we can do that over > iaxtel sometime? I haven't forgotten about it, I just haven't had time to do it. I moved about 2,500 miles across the US from Ohio to California at the end of last month. I also have to do a reinstall of my Aegir/Midgard setup since I managed to break it. I'll start on this this week. -- Steven J. Sobol, Geek In Charge, JustThe.net "Microsoft must think they're a navy, they open so many ports." --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dynamically setting up/tearing down extensions
Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be web-based. I'm looking at DynExtenDB (and have played with it). I love that it reads the dialplans out of a MySQL database - that is a critical issue for me. But it has some issues. I have a test dialplan with one call to Playback() - just plays a wav file then exits. When DynExtenDB() is called, it adds one extra step that calls DynExtenDB_Free()... --If I let the wav file play to the end, DynExtenDB_Free() is called properly. If I hang up prematurely, it isn't, and it also isn't called if I set the dialplan to dial out (for example, to forward the call to my cell phone). --If DynExtenDB_Free() *is* called properly, and I then make another call, DynExtenDB() doesn't seem to be called again. --I'm not sure that setting up a dialplan for extension 'h' is a good idea. What if I call, and then someone else calls and I hang up in the middle of the call? I am ready and willing to make changes to the source to DynExtenDB. In fact, I'd like to get it to a point where it could be used in a production environment. But I have a lot of questions before I can do that. BTW, I have looked in the archives, and it's been suggested that maybe AGI is a better way to handle this sort of thing - but wouldn't the same issues still exist?? Thanks SJS -- Steven J. Sobol, Geek In Charge, JustThe.net "Microsoft must think they're a navy, they open so many ports." --Ben Scott on the ISP-TECH mailing list, 18 June 2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
