Re: [Asterisk-Dev] UK Caller ID patch and new CVS
On Friday 23 July 2004 10:29, Conroy, Lawrence (SMTP) wrote: Frankly though, I don't think ANYONE should buy these cards, clone or no -- Get the TDMxx series. Why? Does progress tone/call supervision/CLI work in the UK for the TDM4xx? (at least now CLI works with the X100P :). The TDM card is a true Digium product, not an OEM card resold by Digium. IIRC the FXO module is far higher quality too. In my experience, ISDN works very well with Junghanns's excellent chan_capi (and the AVM passive cards are cheaper) so if a couple of lines is all you need, ... The X101P doesn't support ISDN. I thought we were talking PSTN here not ISDN? -A. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] voicemail message number limits
On Friday 23 July 2004 06:45, mattf wrote: All you have to do is edit the /asterisk/apps/app_voicemail.c file and change this line: #define MAXMSG 100 change it to whatever you want, we usually change it to 999, then just recompile and install and you're good. If you reexamine the code, you'll find that a MAXMSG of 999 actually limits you one less than necessary -- owing to the fact that the first message is message 0, not message 1. Just go with a MAXMSG of 1000. -- Tilghman ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] UK Caller ID patch and new CVS
Hi again, Andrew, folks, Yup - the X100P (or clone) is a PSTN card for use with single lines. Fine device, and WAY cheaper than a TDM4xx with a single populated FXO module (TDM01B), at least in the UK. I assumed that you weren't seriously suggesting a TDM01B for a single analogue line (if they weren't back-ordered). Hence I guessed that the use case is a few lines. AFAICT, the TDM4xx modules don't do busy detection or call supervision or CLI or have settings for correct impedance right now for use in the UK - all we have is the *potential* to get these working sometime in the *future*. If one does need a couple of lines, then one may have a choice - PSTN access or ISDN access. In civilised countries (i.e. not here) it's cheaper to get a BRI than it is to get two analogue lines. Thus, if one needs more than one analogue line, it may be cheaper to go for a BRI from your Comms Provider. If you DO go for a BRI (or a pair of BRIs) the card(s) WILL be cheaper (and, most important, it works, out of the box). No grief with whether or not CLI works - it does now. Likewise, no grief about progress/call supervision - it works over here right now. To sum up, if the use case is more than one line and you don't live in the U.S of A., then don't forget ISDN - it has several advantages (not least of which is that we don't do SPIDs over here). If the use case is a single analogue line, then the price of an under-populated TDM4xx puts it outside the budget for most people - it's just too expensive at current UK reseller prices. all the best, Lawrence On 23 Jul 2004, at 15:50, Andrew Kohlsmith wrote: On Friday 23 July 2004 10:29, Conroy, Lawrence (SMTP) wrote: Frankly though, I don't think ANYONE should buy these cards, clone or no -- Get the TDMxx series. Why? Does progress tone/call supervision/CLI work in the UK for the TDM4xx? (at least now CLI works with the X100P :). The TDM card is a true Digium product, not an OEM card resold by Digium. IIRC the FXO module is far higher quality too. In my experience, ISDN works very well with Junghanns's excellent chan_capi (and the AVM passive cards are cheaper) so if a couple of lines is all you need, ... The X101P doesn't support ISDN. I thought we were talking PSTN here not ISDN? -A. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Visit our website at www.roke.co.uk Roke Manor Research Ltd, Roke Manor, Romsey, Hampshire SO51 0ZN, UK. The information contained in this e-mail and any attachments is confidential to Roke Manor Research Ltd and must not be passed to any third party without permission. This communication is for information only and shall not create or change any contractual relationship. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
RE: [Asterisk-Dev] UK Caller ID patch and new CVS
Perhaps it's not a question of elegant code but rather quality and maintainability. Once code goes in, those are real issues that take time and effort. I've seen technical integrity mistaken for dictatorship many times, and in the end the users suffer if it is replaced by political solutions. Believe me, you would be far worse off if such decisions were in the hands of a populist rather than an software architect. As an open source project one has every opportunity to change the code, but in submitting code, one must be able to take responsibility for long term issues. Please guys, focus on improving patches rather than asking the maintainers to lower standards. Regards, Rich -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, July 22, 2004 4:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Dev] UK Caller ID patch and new CVS Yes I have. The main problem I have is that if a patch can make things working in the short terms then why not add it? If a more elegant solution is found weeks later that patch can simply be taken out, either way the problem is solved,and people are happy since the software 'just works' while digium gains a larger customer base since the hardware now works as expected and more people can then buy the hardware and use it. I'm not even in the UK so that patch will not even help me at all, but I still think it a good idea; That has to say something. On 22 Jul 2004 at 11:51, Richard Lyman wrote: it's obvious you never attempted to get a patch in the linux kernel source! someone HAS to 'govern' the code, else it's just a pile of spagetti. [EMAIL PROTECTED] wrote: The true nature of open source is defined as the source being available and open. Limiting the included code for the central offering based on someone's will because he thinks it will mean he can sell more hardware if he does not do so is not open source, its dictatorship. I find it very sad that based on my understanding asterisk will not include code that will help many people just because one person feels that to do so would hurt his companies profit margins, when the code is no doubt already available somewhere else or is needed by someone. In the time I have watched this list even before I started posting I have seen much of this; Keep up the dictatorship of the central code repository and I guarantee you a branch of the source code will form within the next 3-6 months. Not by me since I do not have the requisite understanding, but I believe it important to say here that if the open source community does not like the way digium or 'Mark' is doing things it will simply make them unnecessary for the project to go forward by cutting them out of the loop. And that perfectly acceptable from a legal standpoint since asterisk is after all GPL. I don't mean to be cruel or annoying, I'm stating facts as I see them. If I am wrong or ignorant by all means tell me, but if it looks like this to me, how do you think it looks to the thousands of other people ghosting around this project and watching in the shadows as I once did? Just a though. On 22 Jul 2004 at 14:43, Chris Stenton wrote: Mark does not like the history buffer method used. I think code will be included for the the fxo module at some point but not for the X100P. Chris. ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] UK Caller ID patch and new CVS
As one of the core developers of the OpenH323 project, I agree with this view. Sometimes, forcibly integrating user contributed patches at any cost creates more problems than it's solves. It's not always a matter of increasing functionality at any cost - in the long term, an Open Source project is more about mantainability and ease of understanding than simply increasing functionality. Sometimes, it is better to wait until someone who *really* understands the code is able to create a patch that is fully integrated with the existing code, rather than just hacking in a change to solve a problem. I know that OpenH323 is a dirty word as far as some of the Asterisk developers are concerned, but this problem is common to both projects. Good patches are hard to find :) Craig On Fri, 23 Jul 2004 10:57:26 -0500 Dr. Rich Murphey [EMAIL PROTECTED] wrote: Perhaps it's not a question of elegant code but rather quality and maintainability. Once code goes in, those are real issues that take time and effort. I've seen technical integrity mistaken for dictatorship many times, and in the end the users suffer if it is replaced by political solutions. Believe me, you would be far worse off if such decisions were in the hands of a populist rather than an software architect. As an open source project one has every opportunity to change the code, but in submitting code, one must be able to take responsibility for long term issues. Please guys, focus on improving patches rather than asking the maintainers to lower standards. Regards, Rich --- Craig Southeren [EMAIL PROTECTED] / [EMAIL PROTECTED] Phone: +61 243654666 ICQ: #86852844 Fax:+61 243673140 MSN: [EMAIL PROTECTED] Mobile: +61 417231046 Jabber: [EMAIL PROTECTED] Post Increment - Consulting Serviceshttp://www.postincrement.com Vox Gratia - The Open Source VoIP portal http://www.voxgratia.org Raving Of A Strange Mind - the VoIP blog http://www.southeren.com/blog ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
[Asterisk-Dev] chan_alsa seems to be broken.
Hi, chan_alsa seems to be broken. There is no audio-input possible Cu ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] voicemail message number limits
Without looking at the code, a limit of 99 doesn't make sense. Even if they are indexed by only a single byte that is a total of at least 255 unless the code is really broken. Perhaps the wiki needs updated? On 23 Jul 2004 at 7:07, Frank wrote: I had read in the wiki that the voicemail was limited to 99 messages. So I started to go through the code so I could change this. but it appears that this is not the case in the code. Looks like it support much more. Am I missing something in the code that really limits the message total to 99 ? ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [Asterisk-Dev] voicemail message number limits
On Fri, 2004-07-23 at 12:56 -0700, wrote: Without looking at the code, a limit of 99 doesn't make sense. Even if they are indexed by only a single byte that is a total of at least 255 unless the code is really broken. It's an artificially imposed limit to limit the memory usage of the voicemail app, as there are a couple places that use an array of size MAXMSG. -- Joshua M. Thompson [EMAIL PROTECTED] ___ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
