William Muriithi
2008-Nov-23 03:09 UTC
[asterisk-users] Large Asterisk installations (~10, 000 extensions), preferably at universities
Bourvine,> > So, why won't we save the big bucks we pay them, hire two professionals > (who cost less) and support an open source code by ourselves? This way > we depend on ourselves only. > > > > Thanks, __Yehavi:I remember hearing University of Pennsylvania have been using Asterisk for sometime. I am not certain where I came across that information, but google confirmed it as a fact. And you may need to ask for more details from Digium as they worked together, or call the school. I am relatively certain they would share their experience. The deployment was of 15,000 extensions, just about what you have in mind. Below is some articles. http://www.networkworld.com/news/2007/071707-open-source-voip.html http://www.digium.com/en/company/casestudies/viewcasestudies/University-of-Pennsylvania William> > > > > 2008/11/21 Grygoriy Dobrovolskyy <megahohol at gmail.com> > > > > 2008/11/21 Yehavi Bourvine <yehavi.bourvine at gmail.com> > > Hello, > > > > Our university has to upgrade soon its old Nortel PBX's which > holds around 10,000 extensions tied to 5 PBXes. Up to now we thought > about commercial solutions but now there is a window openning for open > source solution. However, I need examples to convince that this solution > is feasible, and preferably at other universities. > > > > Are there any pointers for such installations? > > > > Thanks! __Yehavi: > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com <http://www.api-digital.com/> -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > Hello very interesting project you have, however asterisk is not > a registry server, i suggest that you use opensips/opense/kamalio for > your registrar, from where you dispatch to you asterisk servers, inside > a good environment with a controlled network and nice tagged voip flow > you could acheve a good results. > > > _______________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com <http://www.api-digital.com/> -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ----------------------------------------- > Disclaimer: > > This e-mail communication and any attachments may contain > confidential and privileged information and is for use by the > designated addressee(s) named above only. If you are not the > intended addressee, you are hereby notified that you have received > this communication in error and that any use or reproduction of > this email or its contents is strictly prohibited and may be > unlawful. If you have received this communication in error, please > notify us immediately by replying to this message and deleting it > from your computer. Thank you. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/9a8636b6/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Fri, 21 Nov 2008 09:46:13 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926C9B5.8080902 at evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Jason Aarons (US) wrote: > >> Just switching from Nortel to something else may not eliminate >> hardware/software failures, or prevent those without experience from >> pushing the enter key at the wrong time. > > One also has to keep in mind - Asterisk, like any large open-source > project, gets a lot more QA, patches and bug fixes than any commercial > product sold in the intra-industrial channel (i.e. excluding consumer > mass-market stuff) ever will! It has a massive installed base, many > users reporting bugs through an open and easy to understand process, and > a large community either directly or derivatively involved in > contributing fixes and testing code. > > How much installed base from which to harness that kind of large-scale > technical feedback does Nortel have? Avaya? Cisco? > > Asterisk has by far the best QA mechanism. In terms of potential bugs > that impact "mission-critical" availability, I would feel better using > it than any of these black-box, proprietary vendor solutions any day. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 10 > Date: Fri, 21 Nov 2008 15:46:59 +0100 > From: Philipp Kempgen <philipp.kempgen at amooma.de> > Subject: Re: [asterisk-users] A way to run extenrnotify when IMAP > events take place... > To: Asterisk Users <asterisk-users at lists.digium.com> > Message-ID: <4926C9E3.4070707 at amooma.de> > Content-Type: text/plain; charset=ISO-8859-1 > > Danny Nicholas schrieb: >> Here is a "Dirty" solution - create a PERL or other script to "listen" for >> changes to voicemail DB/Dir. When VM is deleted, launch script to turn off >> Cisco MWI (should be simple since you are turning on with script). Not >> "Best" solution, just workable one. > > Yeah. If all else should fail there are various dirty solutions > such as listening to events on the manager interface, INotify, > implementing a SMDI interface yourself ... > > Philipp Kempgen > > -- > http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com > Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > -- > > > > ------------------------------ > > Message: 11 > Date: Fri, 21 Nov 2008 09:47:57 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installations (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926CA1D.4070902 at evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Alex Balashov wrote: >> Jason Aarons (US) wrote: >> >>> Just switching from Nortel to something else may not eliminate >>> hardware/software failures, or prevent those without experience from >>> pushing the enter key at the wrong time. >> >> One also has to keep in mind - Asterisk, like any large open-source >> project, gets a lot more QA, patches and bug fixes than any commercial >> product sold in the intra-industrial channel (i.e. excluding consumer >> mass-market stuff) ever will! It has a massive installed base, many >> users reporting bugs through an open and easy to understand process, and >> a large community either directly or derivatively involved in >> contributing fixes and testing code. >> >> How much installed base from which to harness that kind of large-scale >> technical feedback does Nortel have? Avaya? Cisco? >> >> Asterisk has by far the best QA mechanism. In terms of potential bugs >> that impact "mission-critical" availability, I would feel better using >> it than any of these black-box, proprietary vendor solutions any day. >> > > Also, if there is a show-stopping bug, it can be addressed in a > relatively expedient manner, especially if you are paying Digium for > support. > > With the other guys, you're going to have to wait for Service Pack 8 > Patchlevel 4 Release 2 Build 3789 in 12-24 months. It might have a fix. > Maybe. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 12 > Date: Fri, 21 Nov 2008 09:53:36 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926CB70.3040701 at evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Al Baker wrote: > >> Remember - You are going from a CARRIER GRADE purpose built piece of >> hardware with Software built under a rigid CMM with extensive >> "soak-testing" to software that has been developed under , shall we say, >> a somewhat less rigid and stringent methodology. >> You will be moving from an environment supported by hundreds of highly >> trained people, some with decades of TELCO experience >> to one where you support comes from a somewhat less seasoned group of >> individuals. > > But in choosing "carrier grade" (everyone calls their stuff that) > vendors you are also going to a much smaller installed base and much > lower total reporting and QA pool. I would take the sheer number and > dynamism of the Asterisk installed base over their comparatively limited > deployments, even if we grant the unsubstantiated premise that the > latter is developed under a less rigid and stringent methodology. > > Let me put it this way: if I wrote a piece of software and sold it to > 10 customers, it won't matter for overall product quality that I fix the > problems they report and maintain it for them under the guidance of a > "rigid" and "stringent" methodology. That's nice. Hope it fixes their > problems. It is really of comparatively minor benefit to prospective > future adopters. It's not nearly as valuable as simply doing the best I > can with bug reports and test cases from hundreds of users. > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 13 > Date: Fri, 21 Nov 2008 11:59:29 -0300 > From: "Sebastian Milioto" <smilioto at gmail.com> > Subject: [asterisk-users] Ping > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <e6e7910f0811210659m7dc9d8b7t4c171a9093b59c95 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Ping > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/8392150e/attachment-0001.htm > > ------------------------------ > > Message: 14 > Date: Fri, 21 Nov 2008 10:04:57 -0500 > From: "Noah Miller" <noahisaacmiller at gmail.com> > Subject: Re: [asterisk-users] Limit the number of users in a meetme > conference? > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <8699dcab0811210704w2ec131eepdf7fc0ae18c10e42 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Dan - > >>> I found the "maxusers" defined in meetme.c, but I'm >>> not sure how this value is set. Does anybody know >>> if one can limit the number of users permitted in a >>> meetme conference? I know there's MeetmeCount(), but >>> I'd rather avoid the dialplan logic and just set >>> maxusers instead. >> >> That feature is primarily used with RealTime conferences. >> The maxusers value is read from a database and enforced >> on RealTime enable conferences. This presumes you are >> looking at 1.6.X or Trunk code... > > Ah. No realtime for me, so I guess I'll just stick with using > MeetmeCount() in the dialplan. Thanks for the info! > > > - Noah > > > > ------------------------------ > > Message: 15 > Date: Fri, 21 Nov 2008 10:29:02 -0500 > From: "Noah Miller" <noahisaacmiller at gmail.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <8699dcab0811210729i29e38cbcjd4c6542a02fc983e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > >> Due diligence is required on anything 10,000 people are going to be >> pounding on. Undersizing is common, > > I think due diligence is THE key with any open source solution, > including asterisk. I'll admit that I pretty badly screwed up one > asterisk installation because I didn't adequately prepare it (shipped > it to the customer and had their IT staff install - bad plan). And > while I've never done a system anywhere near 10K extensions, I've had > good experiences with some large-ish installations because I budgeted > in the time for research and testing. > > I know that in the past there have been people on this list who have > done very large scale asterisk deployments. Not sure if any of them > are still around to comment. > > With that many extensions, I'll second using a SIP registrar like > Freeswitch or OpenSer. Just use asterisk to provide the services. > > >> and is only one of the roads that >> leads to Hell (I prefer Patterson Lake Road myself since I drive in from >> the North East). > > Hmm. You must live near Ann Arbor. > > > - Noah > > > > ------------------------------ > > Message: 16 > Date: Fri, 21 Nov 2008 10:32:51 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926D4A3.7000306 at evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Noah Miller wrote: > >> With that many extensions, I'll second using a SIP registrar like >> Freeswitch or OpenSer. Just use asterisk to provide the services. > > Third. > > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 17 > Date: Fri, 21 Nov 2008 07:36:27 -0800 > From: "Roderick A. Anderson" <raanders at acm.org> > Subject: [asterisk-users] [SOLVED] TDM400 (?) zap hangup > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926D57B.5050501 at acm.org> > Content-Type: text/plain; charset=UTF-8; format=flowed > > Roderick A. Anderson wrote: >> And if that ain't confusing I don't know what would be. >> >> I bought a TDM400 with two modules (FXO, FXS) a couple or so years ago >> and ended up never using it. Passed it along to a friend who is having >> some problems with it. (He isn't on this list.) >> >> We've both tried searches using Google but haven't been able to find >> anything that helps. So this is more a question of >> what-the-heck-should-we-be-searching-for. :-) >> >> The TDM400 works taking inbound calls and gives a dial tone when the >> phone is picked up but as soon as a key is pressed the line (Asterisk >> says) hangs up. Asterisk is configured based on a working system but >> that one only has one module inbound (FXO?) The outbound settings are >> based on docs from voip-info.org. >> >> Does this ring a bell for anyone? No pun intended. >> >> Unfortunately the system is 35 miles away and I haven't got the logs >> handy so I can't provide more right now. Just hoping for a clue on >> search terms. > > Thanks to Tzafrir Cohen and Jared Smith we've solved the problem. > > It was a "A Series of Unfortunate Events". The main one was, there was > no (and then an incorrect) context= for the ZAP channel. The incorrect > one came about because of a miss-communication while testing. We were > able to dial-out but the logic in the dialplan to select a context for > local calls, toll-free, etc. was missing. Once we got the channel set > to the correct context all was well. > > > Again thanks, > Rod > -- >> TIA, >> Rod > > > > > ------------------------------ > > Message: 18 > Date: Fri, 21 Nov 2008 10:42:12 -0500 > From: "Matt Florell" <astmattf at gmail.com> > Subject: Re: [asterisk-users] How long will Asterisk 1.4.x > supported/maintained > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <61575c810811210742j6080a6d8q8018aa202d02d687 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On 11/20/08, Steve Totaro <stotaro at totarotechnologies.com> wrote: >> On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> > On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote: >> >> 2008/11/17 Philipp Kempgen <philipp.kempgen at amooma.de> >> >> >> >> > Tilghman Lesher schrieb: >> >> > > On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote: >> >> > >> Is there somewhere a statement from Digium how long they will support >> >> > >> Asterisk 1.4? >> >> > > >> >> 0>> > > There is no statement, because we haven't even discussed when >> >> the EOL for >> >> > > 1.4 will be reached. Certainly that means it won't happen for at least >> >> > the >> >> > > next 60 days, but beyond that, I really don't know. >> >> > >> >> > For the average non-techie user who does not want to compile >> >> > themselves that may sound funny (if not scary). >> >> > >> >> > When Debian Lenny (featuring Asterisk 1.4) is finally going to be >> >> > released that version might not even be supported any more. >> >> >> >> >> >> I think to a large extend, Asterisk is not to be considered as binary >> >> distributed at all, as many hardware it supports is not directly managed by >> >> kernel team. >> > >> > Interesting consideration. Debian Etch and RHEL5 are based on kernel >> > 2.6.18, but support quite a few hardware devices not included in that >> > kernel. >> > >> > If this issue bothers you, please help test the alternative timing >> > mechanism support now included in trunk. >> > >> > -- >> > Tzafrir Cohen >> > icq#16849755 jabber:tzafrir.cohen at xorcom.com >> > +972-50-7952406 mailto:tzafrir.cohen at xorcom.com >> > http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir >> > >> >> >> I still compile and install 1.2 for the most part, for call centers >> and large systems. >> >> The few 1.4 installs that I have done have been for "medium" sized >> PBXs, say 50-70 phones/users and they have been trouble free for the >> most part. Safe_asterisk may make some troubles transparent. >> >> I am not really sure what 1.4 has over 1.2 for the average PBX installation. >> >> Then you have the OpenPBX guys who forked 1.2, I know they have added >> functionality to 1.2, but the following puts me off. Perhaps >> vaporware, perhaps not, it all relies on the devs. You also have >> people like Matt Florell who have continued to add functionality to >> 1.2 but since Digium won't take them, or the dev doesn't want to sign >> over their first born, they are hard to come by but certainly out >> there. >> >> 1.4 may follow the same path, being forked. >> >> 1.6 is not on my radar. >> >> >> -- >> Thanks, >> Steve Totaro >> +18887771888 (Toll Free) >> +12409381212 (Cell) >> +12024369784 (Skype) > > Hello, > > We really just maintain a set of patches for 1.2 (just updated > waitforsilence a couple weeks ago in fact) and we regularly install > 1.2.30.2 in call center setups. It is rock solid and extremely proven > in high-call-volume situations. > > We have started installing 1.4.21.2 on some systems that are not high > load as well (1.4.22 has some strange issues with it we have noticed) > because we do have clients requesting to use 1.4 for some of the nicer > PBX functionality that it has as well as better SIP support. > > We test 1.6 periodically and we are very much looking forward to some > of the great new features of it, but it crashes very quickly when > trying to use it in call center situations. just keep in mind that in > my opinion the 1.4 tree did not become usable until 1.4.18 when most > of the major bugs were finally fixed. > > MATT--- > > > > ------------------------------ > > Message: 19 > Date: Fri, 21 Nov 2008 17:42:17 +0200 > From: "Atis Lezdins" <atis at iq-labs.net> > Subject: Re: [asterisk-users] Ping > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <670f60170811210742v4d9baf35pc58f7f5db5cd3d09 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Fri, Nov 21, 2008 at 4:59 PM, Sebastian Milioto <smilioto at gmail.com> wrote: >> Ping >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > Pong > > GMail's preview looks fun - "Ping -- Bandwidth and Colocation Provided > by http://www.api-digital.com" > > Regards, > Atis > > > -- > Atis Lezdins, > VoIP Project Manager / Developer, > IQ Labs Inc, > atis at iq-labs.net > Skype: atis.lezdins > Cell Phone: +371 28806004 > Cell Phone: +1 800 7300689 > Work phone: +1 800 7502835 > > > > ------------------------------ > > Message: 20 > Date: Fri, 21 Nov 2008 13:46:00 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <dcc007e10811210746s60e8d957i649106883a40ed3b at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <noahisaacmiller at gmail.com>wrote: > >> [..snip..] > > With that many extensions, I'll second using a SIP registrar like >> Freeswitch or OpenSer. Just use asterisk to provide the services. >> > > Is Asterisk even needed? > > - Gonzalo > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/e72dd610/attachment-0001.htm > > ------------------------------ > > Message: 21 > Date: Fri, 21 Nov 2008 09:46:27 -0600 > From: "Danny Nicholas" <danny at debsinc.com> > Subject: Re: [asterisk-users] Limit the number of users in a > meetmeconference? > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> > Message-ID: <3D33EB687590414696C75D9209ACF1F1 at db0005> > Content-Type: text/plain; charset="us-ascii" > > Armed with a little more information, here is a more realistic reply. > In the 1.6.0.1 code, app_meetme.c defines maxusers in line 369 and sets the > max value in line 870 to 0x7fffffff. > Therefore changing line 870 would allow you to limit the maxusers. > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Noah Miller > Sent: Friday, November 21, 2008 9:05 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Limit the number of users in a > meetmeconference? > > Hi Dan - > >>> I found the "maxusers" defined in meetme.c, but I'm >>> not sure how this value is set. Does anybody know >>> if one can limit the number of users permitted in a >>> meetme conference? I know there's MeetmeCount(), but >>> I'd rather avoid the dialplan logic and just set >>> maxusers instead. >> >> That feature is primarily used with RealTime conferences. >> The maxusers value is read from a database and enforced >> on RealTime enable conferences. This presumes you are >> looking at 1.6.X or Trunk code... > > Ah. No realtime for me, so I guess I'll just stick with using > MeetmeCount() in the dialplan. Thanks for the info! > > > - Noah > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > ------------------------------ > > Message: 22 > Date: Fri, 21 Nov 2008 10:48:57 -0500 > From: Alex Balashov <abalashov at evaristesys.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926D869.2080305 at evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Gonzalo Servat wrote: >> On Fri, Nov 21, 2008 at 1:29 PM, Noah Miller <noahisaacmiller at gmail.com >> <mailto:noahisaacmiller at gmail.com>> wrote: >> >> [..snip..] >> >> With that many extensions, I'll second using a SIP registrar like >> Freeswitch or OpenSer. Just use asterisk to provide the services. >> >> >> Is Asterisk even needed? > > Potentially, no. But if you intend to provide subscriber/PBX features, > it is needed as a UA feature box(s). > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > Message: 23 > Date: Fri, 21 Nov 2008 11:14:57 -0500 > From: RE Kushner List Account <lists at darl.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Message-ID: <4926DE81.50206 at darl.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Noah Miller wrote: >> >>> and is only one of the roads that >>> leads to Hell (I prefer Patterson Lake Road myself since I drive in from >>> the North East). >>> >> >> Hmm. You must live near Ann Arbor. >> >> > > No, northern suburbs of Detroit. M-59 to US-23 S to M-36 W..To S. > Howell St..Patterson Lake Rd..To Hell.... > > Ann Arbor is quite a bit South of Hell. Actually it's been some time > since I've been to Hell but I'm sure it's frozen over today ;-) > > -Ron > > > > > ------------------------------ > > Message: 24 > Date: Fri, 21 Nov 2008 11:28:18 -0500 > From: Jerry Geis <geisj at pagestation.com> > Subject: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel > audio > To: asterisk-users at lists.digium.com > Message-ID: <4926E1A2.1000001 at pagestation.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi all, > > I upgraded from asterisk 1.2.23 and zaptel 1.2.19 > to asterisk 1.4.18 and zaptel 1.4.12.1 > I use polycom 501 phones internally. > > Everything seems fine. I can pick up the phone and call out, > calls coming in work just fine. > > The issue I see is when the system first calls me, > then calls someone else. This works if its polycom to polycom. I hear > audio full channel. > If I do polycom to external line like a cell I only get HALF channel audio. > At this time they can hear me but I cannot hear them. > > What might this be??? > > Jerry > > > > ------------------------------ > > Message: 25 > Date: Fri, 21 Nov 2008 17:32:22 +0100 > From: Olivier <oza-4h07 at myamail.com> > Subject: [asterisk-users] OT - SIP message encoding to enhance text > display > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <442fbb120811210832h13e5b054ncf57a66c8a5dcb47 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > > I've read RFC3428 which presents SIP MESSAGE. > Is there any extension or encoding scheme working with SIP MESSAGE that > would enhance text display with blinking or underlining attributes ? > This could be useful to notify SIP hardphone users with some important > events such being in Do Not Disturb mode. > > Regards > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081121/472872a0/attachment.htm > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 52, Issue 57 > ********************************************** >
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