Re: [asterisk-users] 2000+ user Asterisk PBX
On Sat, Aug 2, 2008 at 5:45 PM, Femi <[EMAIL PROTECTED]> wrote: >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On Behalf Of Chris Rowson >> Sent: 02 August 2008 19:42 >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] 2000+ user Asterisk PBX >> >> >> Any 2000+ user Asterisk PBX installs out there? >> >> >> >> Please hit me off-list, I need some support on a 2000+ user Asterisk >> PBX >> >> with high availability and over 10E1s to PTOs >> >> >> >> >> >> >> >> Femi >> > >> > I would be interested in some of the replies if you wanted to continue >> the >> > topic on-list... Your problem might help someone else down the line. >> >> Me too, >> >> Any reason you want this off the list particularly? >> >> Chris > > Sorry if I appear selfish by asking for it to be off list > Please by all means post non-commercial replies to my request on-list > however any responses of a commercial nature (and that is primarily what I'm > looking for) would naturally be off-list. > > Femi > Yes there are. I am putting one together currently and have done 1,000 in the past, it is just a matter of putting the right things in the right places. Obviously, a decent sized budget is required, some TLC, and a few racks if you want good redundancy. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] lookup for '_sip._udp.sip.stanaphone.com'
I am having problems with my sip service with stanaphone. I think it is related to my firewall which had a glitch yesterday. Can anyone tell me what this means below? -- ast_get_srv: SRV lookup for '_sip._udp.sip.stanaphone.com' mapped to host sip.stanaphone.com, port 5060 Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I issue a Flash to Zap (PSTN) from SIP?
This is what I do: /etc/asterisk/features.conf [applicationmap] inflash => *4,caller,Flash,() outflash => *3,callee,Flash,() in /etc/asterisk/extensions.conf before accepting a call: exten => s,n,Set(DYNAMIC_FEATURES=inflash) on an outgoing call: exten => _1XX,1,Set(DYNAMIC_FEATURES=outflash) in incoming calls the user has to press *4 on outgoing calls the user has to press *3 On Sat, Aug 2, 2008 at 4:59 PM, Jim Duda <[EMAIL PROTECTED]> wrote: > I've seen a few posts on this issue, however, no definitive answer. > > My PSTN is connected to Zap/4. I have simple Call Waiting service on > the PSTN line. > > All the other phones are SIP clients. > > When I'm on an Zap/SIP connection and another call comes in, I can hear > the Call Waiting Tone on the SIP line. > > How can I issue a Flash/Hook to the Zap line in order to accept the > other call? > > Also, is there any means to get Caller ID for the other call? > > I've seen posts that I can use *0 or *3 to send the Flash/Hook, however, > that doesn't work for me. > > I realize there is a Flash( ) Dialplan function. How can I use this > Function in the Dialplan with a call which is currently in progress? > > Any advice is most appreciated. > > Jim > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Door phone monitor to Asterisk box?
I would look at AIPHONE for product that are built to do this and work perfectly. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Rowson > Sent: 02 August 2008 19:42 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] 2000+ user Asterisk PBX > > >> Any 2000+ user Asterisk PBX installs out there? > >> > >> Please hit me off-list, I need some support on a 2000+ user Asterisk > PBX > >> with high availability and over 10E1s to PTOs > >> > >> > >> > >> Femi > > > > I would be interested in some of the replies if you wanted to continue > the > > topic on-list... Your problem might help someone else down the line. > > Me too, > > Any reason you want this off the list particularly? > > Chris Sorry if I appear selfish by asking for it to be off list Please by all means post non-commercial replies to my request on-list however any responses of a commercial nature (and that is primarily what I'm looking for) would naturally be off-list. Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Now - rPath virtual appliance
Anyone either playing with it for testlab purposes or deplying in the field? http://www.rpath.org/rbuilder/project/asterisk/release?id=5725 Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I issue a Flash to Zap (PSTN) from SIP?
I've seen a few posts on this issue, however, no definitive answer. My PSTN is connected to Zap/4. I have simple Call Waiting service on the PSTN line. All the other phones are SIP clients. When I'm on an Zap/SIP connection and another call comes in, I can hear the Call Waiting Tone on the SIP line. How can I issue a Flash/Hook to the Zap line in order to accept the other call? Also, is there any means to get Caller ID for the other call? I've seen posts that I can use *0 or *3 to send the Flash/Hook, however, that doesn't work for me. I realize there is a Flash( ) Dialplan function. How can I use this Function in the Dialplan with a call which is currently in progress? Any advice is most appreciated. Jim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxeo
Proprietary ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voxeo
Is anyone on this list using Voxeo hosted vxml application services? Are they really doing $24 million dollars worth of business in hosted services per annum? I thought with that kind of volume I would have heard them being mentioned more often. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2000+ user Asterisk PBX
>> Any 2000+ user Asterisk PBX installs out there? >> >> Please hit me off-list, I need some support on a 2000+ user Asterisk PBX >> with high availability and over 10E1s to PTOs >> >> >> >> Femi > > I would be interested in some of the replies if you wanted to continue the > topic on-list... Your problem might help someone else down the line. Me too, Any reason you want this off the list particularly? Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HI ~ good friend,
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Saturday, 2 August 2008 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HI ~ good friend, On Fri, 2008-08-01 at 16:08 -0400, Dean Collins wrote: > Yep I totally agree with you that documentation is an area digium is > dropping the ball. > >Mayhaps, but as one of the digium guys, I might add that Asterisk >content is not solely supplied by Digium. Users out there, dissatisfied >with documentation on various parts of Asterisk are more than welcome >to help fill in the gaps! Join #asterisk-doc, get on the mailing list, >put a sheet of paper in your typewriter (er, word processor), and >have at it! Gasp no really I would never have seen that answer coming :P lol you guys are so predictable sometimes. Cheers, Dean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] auto provisioning phones
Thridlane PBX Manager (www.thirdlane.com) has a comprehensive phone provisioning system with templates. If your phone isn't supported already you can create your own templates easily. Jim Michael Graves wrote: > Which Asterisk systems provide automatic provisioning of phones? > > Switchvox? ABE? The AA series appliances? Trixbox? > > I know that the VDEX-40 (Voiceroute) and Jazinga do this. > > Michael > -- > Michael Graves > mgravesmstvp.com > http://blog.mgraves.org > o713-861-4005 > c713-201-1262 > sip:[EMAIL PROTECTED] > skype mjgraves > [EMAIL PROTECTED] > > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HI ~ good friend,
On Fri, 2008-08-01 at 16:08 -0400, Dean Collins wrote: > Yep I totally agree with you that documentation is an area digium is > dropping the ball. > Mayhaps, but as one of the digium guys, I might add that Asterisk content is not solely supplied by Digium. Users out there, dissatisfied with documentation on various parts of Asterisk are more than welcome to help fill in the gaps! Join #asterisk-doc, get on the mailing list, put a sheet of paper in your typewriter (er, word processor), and have at it! As to dimensioning, I might add, that to do justice, and create metrics, is a tremendous task. Factors that would affect the calls/sec and concurrent calls numbers would be: 1. The drivers involved 2. the codecs in use 3. the hardware used (digium vs Sangoma, etc) 4. the CPU speed 5. the memory speed 6. the memory amount 7. the size of CPU caches 8. bus speeds 9. disk speeds 10. network effective bandwidth 11. Call logging (to console & CDR backends) 12. Asterisk software version 13. Hardware (dahdi, etc) driver versions ... and so on.. I cannot even begin to enumerate all the factors. (And, on top of the above, I'm willing to bet that for each factor, the speed will probably NOT be a simple linear relationship) Now, the Cisco guys and others can chop the list down because they can set the hardware the software runs on. They can run a few tests and come up with sets of metrics that give you an idea about how fast the darn thing is. But we just do the software. Digium sells cards, but they run in a very widely different range of machines. Coming up with a formula that you can plug numbers into would be a big, expensive task, and by the time it was produced, it would be wrong because of software fixes, speedups, slowdowns, etc. No, really, the only practical approach is for a user to freeze all the factors he can, like software version and most of the hardware numbers, and run load tests to see how well things work out. In the end, the individual implementor is responsible to know what his implementation can do. If the combo doesn't meet his needs, he can get a faster cpu, more mem, etc. murf > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Al Baker > Sent: Friday, 1 August 2008 3:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] HI ~ good friend, > > I must disagree. > "Dimensioning" of Asterisk is a very sorely lacking area and is one of > the main area CISCO > and such eats its lunch. There simply no a base of solid metric that > allow for true "provisioning" . > Yes, there are INVALUABLE anecdotal reports from people who have been > kind, and sharing of their > experiences and for which are all very very grateful. > BUT > That that just is not the same as as solid, vendor based Metrics. > Can you imagine calling and asking DISCO, "What do I need for 400 calls" > > an their answer is > "Here please go read these mostly outdated anecdotal reports and call > back with your order" > Sorry. I love *, but this area of it is not where it needs to be. > > Dean Collins wrote: > > > > Hi welcome to the asterisk community. > > > > > > > > The answer you want are here; > > http://www.voip-info.org/wiki/view/Asterisk+dimensioning > > > > > > > > The short answer is; Pretty much yes, depending on hardware and > > horizontal scaling with multiple servers sharing the load. > > > > > > > > > > Cheers, > > > > Dean > > > > > > > > > *From:* [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] *On Behalf Of *??? > > *Sent:* Friday, 1 August 2008 9:43 AM > > *To:* asterisk-users@lists.digium.com > > *Subject:* [asterisk-users] HI ~ good friend, > > > > > > > > hi ~ nice to meet you, i just join here, today, > > > > > > > > i am a student, and i am very interesting in asterisk. > > > > > > > > and i have a IP-PBX server, made by me with my friend, > > > > > > > > while when i studying, i have a question, > > > > > > > > is there any limit users for asterisk? > > > > > > > > ex) registed users number is 1000 or 1 or 10 like that, is > > that possible? > > > > > > > > and how about the concurrent calls? 1000 concurrent calls is possible? > > > or 2000 concurrent calls? > > > > > > > > my PBX server's user is just less then 15, almost my friends, > > > > > > > > so, i can't test, over 10 users and 1000 concurrent calls, > > > > > > > > please tell me, it is possible or not? > > > > > > > > thanks your permission to join there, > > > > > > > > > > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > asterisk-users mailing list > > To UNSUBSCRIB
Re: [asterisk-users] 2000+ user Asterisk PBX
> Any 2000+ user Asterisk PBX installs out there? > > Please hit me off-list, I need some support on a 2000+ user Asterisk PBX > with high availability and over 10E1s to PTOs > > > > Femi I would be interested in some of the replies if you wanted to continue the topic on-list... Your problem might help someone else down the line. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2000+ user Asterisk PBX
Any 2000+ user Asterisk PBX installs out there? Please hit me off-list, I need some support on a 2000+ user Asterisk PBX with high availability and over 10E1s to PTOs Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HI ~ good friend,
On Fri, Aug 01, 2008 at 03:51:21PM -0400, Al Baker wrote: > I must disagree. > "Dimensioning" of Asterisk is a very sorely lacking area and is one of > the main area CISCO > and such eats its lunch. There simply no a base of solid metric that > allow for true "provisioning" . > Yes, there are INVALUABLE anecdotal reports from people who have been > kind, and sharing of their > experiences and for which are all very very grateful. > BUT > That that just is not the same as as solid, vendor based Metrics. > Can you imagine calling and asking DISCO, "What do I need for 400 calls" > an their answer is > "Here please go read these mostly outdated anecdotal reports and call > back with your order" > Sorry. I love *, but this area of it is not where it needs to be. I'll byte. Al, What do you need for "400 calls" on Asterisk? Please give me a short and clear answer or I won't buy Asterisk from you. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
