Re: IAX2 Java library (was Re: [Asterisk-Users] New IAX software phone (for WIndows platform))

2003-11-04 Thread Steven J. Sobol
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

Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec

2003-10-15 Thread Steven J. Sobol
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

Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec

2003-10-15 Thread Steven J. Sobol
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

Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec

2003-10-15 Thread Steven J. Sobol
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

RE: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't:no sound after 5sec

2003-10-15 Thread Steven J. Sobol
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

[Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec

2003-10-14 Thread Steven J. Sobol
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

Re: [Asterisk-Users] My Grandstream works, but my X-Lite doesn't: no sound after 5sec

2003-10-14 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Does SIP work?

2003-09-25 Thread Steven J. Sobol
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

[Asterisk-Users] AGI: getting the return code from an exec()'d application?

2003-09-25 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Newbie IVR question

2003-09-10 Thread Steven J. Sobol
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

[Asterisk-Users] SIP Status Codes

2003-09-08 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Newbie IVR question

2003-09-01 Thread Steven J. Sobol
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

[Asterisk-Users] More questions. Call Waiting and Threeway

2003-08-26 Thread Steven J. Sobol
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

[Asterisk-Users] Any way to distinguish between...

2003-08-24 Thread Steven J. Sobol
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

Re: [Asterisk-Users] VoIP dialtone?

2003-08-21 Thread Steven J. Sobol
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

[Asterisk-Users] Unregister SIP connection?

2003-08-12 Thread Steven J. Sobol
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

Re: [Asterisk-Users] So now I'm playing around with Queues....

2003-08-08 Thread Steven J. Sobol
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

[Asterisk-Users] So now I'm playing around with Queues....

2003-08-06 Thread Steven J. Sobol
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

Re: [Asterisk-Users] (no subject)

2003-08-05 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Asterisk community input: FreeTDS (cdr_tds.c)or unixODBC (cdr_unixodbc.c) ?

2003-08-01 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Manager.pm port

2003-07-31 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Manager

2003-07-31 Thread Steven J. Sobol
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

[Asterisk-Users] PHP API for Manager - Plaintext auth needed?

2003-07-31 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Microsoft SQL

2003-07-30 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Microsoft SQL

2003-07-30 Thread Steven J. Sobol
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

Re: [Asterisk-Users] X-Lite and Call transfer using Asterisk

2003-07-30 Thread Steven J. Sobol
On Wed, 30 Jul 2003, Brian West wrote: Same here. Same build. AOL -- 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] Manager.pm port

2003-07-30 Thread Steven J. Sobol
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

RE: [Asterisk-Users] Asterisk user guide ..

2003-07-29 Thread Steven J. Sobol
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,

RE: [Asterisk-Users] Best software SIP client

2003-07-25 Thread Steven J. Sobol
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

[Asterisk-Users] AGI.pm?

2003-07-23 Thread Steven J. Sobol
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

[Asterisk-Users] Dynamically setting up/tearing down extensions

2003-07-21 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Dynamically setting up/tearing down extensions

2003-07-21 Thread Steven J. Sobol
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

Re: [Asterisk-Users] Dynamically setting up/tearing down extensions

2003-07-21 Thread Steven J. Sobol
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