[email protected] wrote:
>Send asterisk-users mailing list submissions to > [email protected] > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > [email protected] > >You can reach the person managing the list at > [email protected] > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of asterisk-users digest..." > > >Today's Topics: > > 1. How to stop ringing when incoming PSTN call is answered > externally? ([email protected]) > 2. Is Asterisk Support RFC-5168 (DHAVAL INDRODIYA) > 3. Re: Dynamic hint from db? (Roland) > 4. Asterisk and chat (Matteo Calorio) > 5. Re: Rate sheet "normalization" (A E [Gmail]) > 6. Re: Rate sheet "normalization" (Alex Balashov) > 7. Re: Rate sheet "normalization" (C. Savinovich) > 8. Re: Rate sheet "normalization" (Leandro Dardini) > 9. Re: Rate sheet "normalization" (C. Savinovich) > 10. Re: Rate sheet "normalization" (Don Kelly) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Tue, 27 Mar 2012 22:27:40 -0400 >From: "[email protected]" <[email protected]> >Subject: [asterisk-users] How to stop ringing when incoming PSTN call > is answered externally? >To: [email protected] >Message-ID: <[email protected]> >Content-Type: text/plain; charset=ISO-8859-1; format=flowed > >This is a hard one to explain. My home PSTN line is connected via an >Openvox A400P card to my Asterisk 1.6.2.23 box which then routes >incoming calls to my 2 SCCP extensions. > >The calls are routed just fine, but when a call is answered at one of >the extensions or externally (by a home telephone) the asterisk >extensions continue to ring one more time. Is there a way to have >Asterisk drop an incoming PSTN call as soon as it's answered? > >CLI output when receiving a PSTN call: > Starting simple switch on 'DAHDI/3-1' > -- Executing [s@from-pstn-3:1] Wait("DAHDI/3-1", "1") in new stack > -- Executing [s@from-pstn-3:2] Verbose("DAHDI/3-1", "CALLERID is >XXXXXXXXXX") in new stack >CALLERID is XXXXXXXXXX > -- Executing [s@from-pstn-3:3] Verbose("DAHDI/3-1", "Time is >20120327-204307") in new stack >Time is 20120327-204307 > -- Executing [s@from-pstn-3:4] Dial("DAHDI/3-1", >"SCCP/1000&SCCP/1100,30") in new stack > -- Called 1000 > -- Called 1100 > -- SCCP/1000-00000038 is ringing > -- SCCP/1100-00000039 is ringing > == Spawn extension (from-pstn-3, s, 4) exited non-zero on 'DAHDI/3-1' > -- Hungup 'DAHDI/3-1' > > >[from-pstn-3] >exten => s,1,Wait(1) >exten => s,n,Verbose(CALLERID is ${CALLERID(num)}) >exten => s,n,Verbose(Time is ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) >;exten => s,n,Answer >exten => s,n,Dial(SCCP/1000&SCCP/1100,30) >exten => s,n,Hangup > > > >------------------------------ > >Message: 2 >Date: Wed, 28 Mar 2012 10:43:20 +0530 >From: DHAVAL INDRODIYA <[email protected]> >Subject: [asterisk-users] Is Asterisk Support RFC-5168 >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> >Message-ID: > <cae8gfe8lsah-xtvct6+to0rmikozy_rvp-dnjzlbo3akk2g...@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >Hi All, > >i am working on video setup within asterisk my simple question is asterisk >support RFC-5168. > >if yes then in which version ? > >thanks >Dhaval >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/1ec0c899/attachment-0001.htm> > >------------------------------ > >Message: 3 >Date: Wed, 28 Mar 2012 10:19:12 +0200 >From: Roland <[email protected]> >Subject: Re: [asterisk-users] Dynamic hint from db? >To: [email protected] >Message-ID: > <cacmtg6+tjktgs1epf_1dvaa10m8svhtjdnighmm4rpu_8ug...@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >I'll answer my own question for the archives... although my question maybe >was just too obvious ;-) > >The problem was, that I had put this piece of Dialplan in my >extensions.conf like this: > >[StumpelLocal] >exten => _ZXX!,hint,${SIP_BYEXT(${EXTEN},${CONTEXT})} >exten => _ZXX!,1,Verbose(3, Search extension ${EXTEN} in context >${CONTEXT}) > same => n,Set(SIP=${SIP_BYEXT(${EXTEN},${CONTEXT})}) > same => n,GotoIf(${SIP}?:notFound) > same => n,SIPAddHeader(Alert-Info: internal) > same => n,Dial(${SIP}) > same => n(notFound),Playback(you-dialed-wrong-number) > same => n,Hangup > > >[StumpelZwaag] >include => StumpelLocal > >I registered my SIP accounts in the database with context StumpelZwaag. But >of course the hints aren't being executed from StumpelZwaag, but probably >from StumpelLocal (if they have a contect at all??), the lookup failed. So >I changed the dynamic hint to: > >exten => _ZXX!,hint,${SIP_BYEXT(${EXTEN},StumpelZwaag)} >exten => _ZXX!,1,Verbose(3, Search extension ${EXTEN} in context >${CONTEXT}) > same => n,Set(SIP=${SIP_BYEXT(${EXTEN},${CONTEXT})}) > same => n,GotoIf(${SIP}?:notFound) > same => n,SIPAddHeader(Alert-Info: internal) > same => n,Dial(${SIP}) > same => n(notFound),Playback(you-dialed-wrong-number) > same => n,Hangup > >I hardcoded the context, so now it works. > >Also I found that "core show hints" on the CLI, also show the hints that >were creating through this dynamic hint. So you will see the actual list of >hints. At first I thought the dynamic hint would only be shown as > _ZXX!@StumpelLocal : ${SIP_BYEXT(${EXTEN} >State:Unavailable Watchers 0 > >This is not the case, so you can test if your dynamic hint is working >correctly by checking this table. > >Also my phone seems to be rather slow in processing the hint changes... so >it may take a few minutes before the changes take effect. > > > > >On Tue, Mar 27, 2012 at 1:25 PM, Roland <[email protected]> wrote: > >> I would like to fetch my extensions from the database. I created a dynamic >> hint, but doesn't seem to work. The BLF on my phone doesn't change when the >> state of the extension changed. >> >> This is in my dialplan: >> >> exten => _ZXX!,hint,${SIP_BYEXT(${EXTEN},${CONTEXT})} >> exten => _ZXX!,1,Verbose(3, Search extension ${EXTEN} in context >> ${CONTEXT}) >> same => n,Set(SIP=${SIP_BYEXT(${EXTEN},${CONTEXT})}) >> same => n,GotoIf(${SIP}?:notFound) >> same => n,SIPAddHeader(Alert-Info: internal) >> same => n,Dial(${SIP}) >> same => n(notFound),Playback(you-dialed-wrong-number) >> same => n,Hangup >> >> Is something like this possible? >> >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/eb605d1f/attachment-0001.htm> > >------------------------------ > >Message: 4 >Date: Wed, 28 Mar 2012 11:49:01 +0200 >From: Matteo Calorio <[email protected]> >Subject: [asterisk-users] Asterisk and chat >To: [email protected] >Message-ID: <[email protected]> >Content-Type: Text/Plain; charset="utf-8" > >Hello, > > >I have a working Asterisk installation, but I would like to add chat between >users; I have also a working Ejabberd installation and with Asterisk's >jabber.conf file I can make my two systems communicate. > >What I can't do is to link together the two accounts, the asterisk extension >and the ejabberd account for every user. > >The final effect I would like is that a user, simply putting "myuser@pbx" and >"mypass" in his softphone (Jitsi in my case) would get both voice and text >messages enabled with (apparently) a single account. > >It's a little difficult to explain for me, but I subscribed an account on >ippi.fr and so did some friends of mine: well, without doing anything else we >have both voice and chat. > > >Bye, > Matteo > > > >------------------------------ > >Message: 5 >Date: Wed, 28 Mar 2012 09:41:57 -0400 >From: "A E [Gmail]" <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> >Message-ID: > <cadzy+jgzhqsd7nquzp41+f5vtxgq9a83tax3sawqpgbfahw...@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >On Mon, Mar 12, 2012 at 6:52 PM, Markus <[email protected]> wrote: > >> Hi, >> >> this question is not Asterisk specific, but since there are so many >> experts present on this list, maybe its OK to ask anyways. >> >> I'm having a hard time "normalizing" rate sheets from different providers. >> What I mean with this: the goal is to always get the cheapest rate for a >> given destination. What I would like to do is throw like 10 rate sheets >> from different providers together and as output get a single rate sheet >> with only the cheapest rates. However, some providers are listing a >> country, lets say Germany, as code "49" with a specific rate, and another >> provider will list each city individually, and each code separately, e.g. >> Berlin "4930", Hamburg "4940" etc., and probably different cities have >> different rates as well. Now, if the "49" route of the first provider is >> cheaper, my system (a2billing) will still use the more expensive "4930" >> code because it is more specific. >> >> I'm looking for some awesome, smart tool that will automatically >> "normalize" all these code differences and output a clean ratesheet with >> only the cheapest rates. >> >> Does such a thing exist? I wonder how everyone else is "normalizing" their >> different rate sheets. With a homebrewn script? >> >> Thanks! >> >> >Markus, > >you're not the first person and certainly not the last person who's ever >asked about this. I had tried this on several mailing lists a little while >ago. A tool that could handle 10 or maybe even 5 provider rate-sheets all >of which can potentially completely differ in formats from each other. Even >worse are the rate update sheets from each provider which are many a times >different from the initial rate sheets that the provider may have given you >and then again they will differ from the rate updates from the remaining 4 >providers you've just painstakingly inserted into your DB. > >Given the popularity of Asterisk and other popular OSS based telephony >platforms with several successful businesses running 100s of millions of >minutes, you'd think at least a few have sorted this problem out. But I >believe those who have, never respond to these emails as it took them quite >a bit of effort to create such a tool and aren't willing to just give it >away. > >Just what I have observed (and was even blatantly told by someone on some >mailing list, can't remember exactly) > >You may have to advertise in the commercial / business list or offer a >bounty. There are several commercial solutions available but I think they >all come as a "feature" of a larger billing/rating/routing platform >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/40f20bf8/attachment-0001.htm> > >------------------------------ > >Message: 6 >Date: Wed, 28 Mar 2012 10:00:59 -0400 >From: Alex Balashov <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> >Message-ID: <[email protected]> >Content-Type: text/plain; charset="utf-8" > >We solve this problem for our customers all the time, in various >situationally-specific ways. But yes, we are not really in a position to >genericise it and give it away. It's not because we are greedy. The time and >resources just aren't there. > >-- >Alex Balashov - Principal >Evariste Systems LLC >235 E Ponce de Leon Ave >Suite 106 >Atlanta, GA 30030 >Tel: +1-678-954-0670 >Fax: +1-404-961-1892 >Web: http://www.evaristesys.com/, http://www.alexbalashov.com > >"A E [Gmail]" <[email protected]> wrote: > >>On Mon, Mar 12, 2012 at 6:52 PM, Markus <[email protected]> wrote: >> >>> Hi, >>> >>> this question is not Asterisk specific, but since there are so many >>> experts present on this list, maybe its OK to ask anyways. >>> >>> I'm having a hard time "normalizing" rate sheets from different providers. >>> What I mean with this: the goal is to always get the cheapest rate for a >>> given destination. What I would like to do is throw like 10 rate sheets >>> from different providers together and as output get a single rate sheet >>> with only the cheapest rates. However, some providers are listing a >>> country, lets say Germany, as code "49" with a specific rate, and another >>> provider will list each city individually, and each code separately, e.g. >>> Berlin "4930", Hamburg "4940" etc., and probably different cities have >>> different rates as well. Now, if the "49" route of the first provider is >>> cheaper, my system (a2billing) will still use the more expensive "4930" >>> code because it is more specific. >>> >>> I'm looking for some awesome, smart tool that will automatically >>> "normalize" all these code differences and output a clean ratesheet with >>> only the cheapest rates. >>> >>> Does such a thing exist? I wonder how everyone else is "normalizing" their >>> different rate sheets. With a homebrewn script? >>> >>> Thanks! >>> >>> >>Markus, >> >>you're not the first person and certainly not the last person who's ever >>asked about this. I had tried this on several mailing lists a little while >>ago. A tool that could handle 10 or maybe even 5 provider rate-sheets all >>of which can potentially completely differ in formats from each other. Even >>worse are the rate update sheets from each provider which are many a times >>different from the initial rate sheets that the provider may have given you >>and then again they will differ from the rate updates from the remaining 4 >>providers you've just painstakingly inserted into your DB. >> >>Given the popularity of Asterisk and other popular OSS based telephony >>platforms with several successful businesses running 100s of millions of >>minutes, you'd think at least a few have sorted this problem out. But I >>believe those who have, never respond to these emails as it took them quite >>a bit of effort to create such a tool and aren't willing to just give it >>away. >> >>Just what I have observed (and was even blatantly told by someone on some >>mailing list, can't remember exactly) >> >>You may have to advertise in the commercial / business list or offer a >>bounty. There are several commercial solutions available but I think they >>all come as a "feature" of a larger billing/rating/routing platform >> >>-- >>_____________________________________________________________________ >>-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >>asterisk-users mailing list >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/a1e16a75/attachment-0001.htm> > >------------------------------ > >Message: 7 >Date: Wed, 28 Mar 2012 07:27:03 -0700 >From: "C. Savinovich" <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> >Message-ID: > > <20120328072703.c58d8914d3535b8829d666a3618b837e.16e5b70218....@email18.secureserver.net> > >Content-Type: text/plain; charset="us-ascii" > >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/12095784/attachment-0001.htm> > >------------------------------ > >Message: 8 >Date: Wed, 28 Mar 2012 16:38:03 +0200 >From: Leandro Dardini <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> >Message-ID: > <CAOoW2hZcPUpR91OnKKa9n=wnroy+hvamjx9dpozud09cavv...@mail.gmail.com> >Content-Type: text/plain; charset="iso-8859-1" > >Continuing with the top post... > >I believe in open source philosophy. A software or a list of telephone >prefix makes no difference. If you want to make such list open source, >you'll be sure somebody will contribute to maintain it update and all will >benefit from it. > >Leandro > >2012/3/28 C. Savinovich <[email protected]> > >> >> I really don't think it is fair for anyone to give out such work for >> free. Unfortunately, many people are used to asking for free software >> solutions for all their problems. Whatever happened to paying for someone >> else's time and effort? >> >> >> Christian Savinovich >> *VoIP & Telephony Consultant* >> >> >> >> >> -------- Original Message -------- >> Subject: Re: [asterisk-users] Rate sheet "normalization" >> From: Alex Balashov <[email protected]> >> Date: Wed, March 28, 2012 10:00 am >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <[email protected]> >> >> We solve this problem for our customers all the time, in various >> situationally-specific ways. But yes, we are not really in a position to >> genericise it and give it away. It's not because we are greedy. The time >> and resources just aren't there. >> >> -- >> Alex Balashov - Principal >> Evariste Systems LLC >> 235 E Ponce de Leon Ave >> Suite 106 >> Atlanta, GA 30030 >> Tel: +1-678-954-0670 >> Fax: +1-404-961-1892 >> Web: http://www.evaristesys.com/, http://www.alexbalashov.com >> >> "A E [Gmail]" <[email protected]> wrote: >> >> On Mon, Mar 12, 2012 at 6:52 PM, Markus <[email protected]> wrote: >> >>> Hi, >>> >>> this question is not Asterisk specific, but since there are so many >>> experts present on this list, maybe its OK to ask anyways. >>> >>> I'm having a hard time "normalizing" rate sheets from different >>> providers. What I mean with this: the goal is to always get the cheapest >>> rate for a given destination. What I would like to do is throw like 10 rate >>> sheets from different providers together and as output get a single rate >>> sheet with only the cheapest rates. However, some providers are listing a >>> country, lets say Germany, as code "49" with a specific rate, and another >>> provider will list each city individually, and each code separately, e.g. >>> Berlin "4930", Hamburg "4940" etc., and probably different cities have >>> different rates as well. Now, if the "49" route of the first provider is >>> cheaper, my system (a2billing) will still use the more expensive "4930" >>> code because it is more specific. >>> >>> I'm looking for some awesome, smart tool that will automatically >>> "normalize" all these code differences and output a clean ratesheet with >>> only the cheapest rates. >>> >>> Does such a thing exist? I wonder how everyone else is "normalizing" >>> their different rate sheets. With a homebrewn script? >>> >>> Thanks! >>> >>> >> Markus, >> >> you're not the first person and certainly not the last person who's ever >> asked about this. I had tried this on several mailing lists a little while >> ago. A tool that could handle 10 or maybe even 5 provider rate-sheets all >> of which can potentially completely differ in formats from each other. Even >> worse are the rate update sheets from each provider which are many a times >> different from the initial rate sheets that the provider may have given you >> and then again they will differ from the rate updates from the remaining 4 >> providers you've just painstakingly inserted into your DB. >> >> Given the popularity of Asterisk and other popular OSS based telephony >> platforms with several successful businesses running 100s of millions of >> minutes, you'd think at least a few have sorted this problem out. But I >> believe those who have, never respond to these emails as it took them quite >> a bit of effort to create such a tool and aren't willing to just give it >> away. >> >> Just what I have observed (and was even blatantly told by someone on some >> mailing list, can't remember exactly) >> >> You may have to advertise in the commercial / business list or offer a >> bounty. There are several commercial solutions available but I think they >> all come as a "feature" of a larger billing/rating/routing platform >> >> ------------------------------ >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/0ff3e834/attachment-0001.htm> > >------------------------------ > >Message: 9 >Date: Wed, 28 Mar 2012 08:58:29 -0700 >From: "C. Savinovich" <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> >Message-ID: > > <20120328085829.c58d8914d3535b8829d666a3618b837e.98755e6ec0....@email18.secureserver.net> > >Content-Type: text/plain; charset="us-ascii" > >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20120328/1ee747ef/attachment-0001.htm> > >------------------------------ > >Message: 10 >Date: Wed, 28 Mar 2012 11:28:24 -0500 >From: "Don Kelly" <[email protected]> >Subject: Re: [asterisk-users] Rate sheet "normalization" >To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> >Message-ID: <[email protected]> >Content-Type: text/plain; charset="utf-8" > >Adding to the top-posted discussion of doing this for free? > > > >A one-time rate sheet is of no value. This is something that would need >constant updating. > > > >Think about the open source projects that provided something useful and were >improved by the community for a couple years, then became static?continuing to >do what they did, but not receiving any more support from the community. They >may continue to be of value, even though they don?t improve. > > > >If the same thing happened with the rate sheet, it would quickly become not >only valueless, but dangerous to rely on?and there would likely be no free >replacement to enable you to stay in business. > >--Don > >Don Kelly > >PCF Corp >People Come First >651 842-1000 >651 842-1001 fax > >From: [email protected] >[mailto:[email protected]] On Behalf Of C. Savinovich >Sent: Wednesday, March 28, 2012 10:58 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] Rate sheet "normalization" > > > > > >Sure someone will benefit from it. But what about all those others who are >financially affected by it? I certainly do not think that someone necessarily >always contributes. It ultimately affects our economy, because money doesn't >circulate, and many people who are in a perfect position to disburse money >just don't do it. It affects an entire industry (the software industry) that >could flourish and create even better products thanks to a competition that >could exists if there were financial rewards. Ultimately, free software (not >open source) affects the little guy and benefits the big guys. Gone are the >days when a talented programmer could create a program and make a million >dollars from his basement with his talent. If your company name is not Google, >if you don't do a tap and a dance to an investor, then you got no chance, as >opposed to the days when you could just sell your own version of Vicidial and >make money for a year. > > > >But rather than discussing about the pros and cons of open source, which is >here to stay, I would think that some people are doomed to fail if they think >they can run a business entirely on free rides. The goal of Open Source is to >benefit from sharing and share ahead, but unfortunately a segment of the >market has gotten to the point that it only takes but doesn't give back. > > > >Christian Savinovich > >VoIP & Telephony Consultant > >646-982-3572 > > > > > >-------- Original Message -------- >Subject: Re: [asterisk-users] Rate sheet "normalization" >From: Leandro Dardini <[email protected]> >Date: Wed, March 28, 2012 10:38 am >To: Asterisk Users Mailing List - Non-Commercial Discussion ><[email protected]> > >Continuing with the top post... > > > >I believe in open source philosophy. A software or a list of telephone prefix >makes no difference. If you want to make such list open source, you'll be sure >somebody will contribute to maintain it update and all will benefit from it. > > > >Leandro > >2012/3/28 C. Savinovich <[email protected]> > > > >I really don't think it is fair for anyone to give out such work for free. >Unfortunately, many people are used to asking for free software solutions for >all their problems. Whatever happened to paying for someone else's time and >effort? > > > > > >Christian Savinovich > >VoIP & Telephony Consultant > > > > > > > >-------- Original Message -------- >Subject: Re: [asterisk-users] Rate sheet "normalization" >From: Alex Balashov <[email protected]> >Date: Wed, March 28, 2012 10:00 am >To: Asterisk Users Mailing List - Non-Commercial Discussion ><[email protected]> > >We solve this problem for our customers all the time, in various >situationally-specific ways. But yes, we are not really in a position to >genericise it and give it away. It's not because we are greedy. The time and >resources just aren't there. > >-- >Alex Balashov - Principal >Evariste Systems LLC >235 E Ponce de Leon Ave >Suite 106 >Atlanta, GA 30030 >Tel: +1-678-954-0670 <tel:%2B1-678-954-0670> > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
