[asterisk-users] prohibit CallerID presentation
On ISDN lines it's possible to prohibit the presentation of caller id, what if I have a SIP gateway, something like an Audiocodes Mediant 1000. How do I prohibit the caller id presentation on that one? Regards, Kristian -- Kristian Larsson KLL-RIPE Network EngineerNet At Once [AS35706] email: [EMAIL PROTECTED] irc: [EMAIL PROTECTED] phone: +46 470 592717cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Users Mailing List Traffic
On Sat, Mar 18, 2006 at 08:23:03PM -0600, Rich Adamson wrote: This same issue has been discussed many times over the last two years. Not likely its going to change now. I just love this attitude. Could someone managing these lists outline the requirements to change the lists? Do we need a vote or is it something which can be simply adopted once the right people recognize the problem (and if so, who is the right people)? Aaron Daniel wrote: Splitting the list by type of request may be a good idea, but splitting based on skill level is just a bad idea... I'm pretty sure that regardless of a newbie's status, they'll still just go to the other lists as the newbie list likely won't do much good. In short, I agree with different lists for hardware and configuration questions... Aaron On Mar 18, 2006, at 4:05 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I was also thinking a list for newbies... PaulH - Original Message - From: Robert La Ferla [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, March 18, 2006 2:33 PM Subject: [Asterisk-Users] Asterisk Users Mailing List Traffic The volume/traffic on this list has been getting pretty heavy. I find it hard to follow certain discussions and there are some that I am not interested in. Perhaps, we could split the list into two: One for discussing hardware (client phones and cards) and one for the software (configuration, problems, etc...) Or some other better scheme that someone can propose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
On Mon, Mar 13, 2006 at 08:12:40AM -0700, Douglas Garstang wrote: Thanks Kristian. It isn't clear how this means a registration on one Asterisk system magically appear on the other though... I'm not quite certain as I build my call routing on scripts instead of Asterisk built in commands, but I beleive Dundi should be able to help you out in situations like this. Kristian -Original Message- From: Kristian Larsson [mailto:[EMAIL PROTECTED] Sent: Monday, March 13, 2006 12:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Clustering On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us Pretend we have peer 123456, then put exten = 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten = 123456,1,NoOp exten = 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian. What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman
Re: [Asterisk-Users] Clustering
On Fri, Mar 10, 2006 at 08:20:27PM -0700, Douglas Garstang wrote: Kevin, From the voip wiki at http://www.voip-info.org/wiki-Asterisk+sip+regcontext: If regcontext is specified, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us Pretend we have peer 123456, then put exten = 123456,2,Dial(SIP/123456) in your extensions.conf When phone 123456 becomes available and registers to the Asterisk, the dialplan will look like: exten = 123456,1,NoOp exten = 123456,2,Dial(SIP/123456) and as you know the dialplan always begin on priority 1 so if the phone is not registered you don't automatically move to priority 2. What I'm curious to know is whether there is a way to use this with SIP RealTime... there doesn't seem to exist a setting for both regexten and regcontext. Any pointers? Kristian. What does this mean exactly? How is it used? I've read the same piece of information dozens of times over the last few months and it makes as much sense to me today, as it did back then, which is about zero. Wow... IAX can be used to share registration info? I've never seen that mentioned anywhere. After reading the patchy docs on DUNDi, I kind of got the impression that it _might_ be able to do that sort of thing, but the docs where so bad they where useless. And while we're on the discussion topic, why doesn't Digium release some docs on DUNDi? It's their baby after all. It seems to be that almost no one uses it, simply because there's no docs that explain how to do it. Alternatively, if you don't have time, can you point me to anywhere where instructions on how to use regcontent is succinctly and clearly documented and explained? Doug. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Fri 3/10/2006 8:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering Douglas Garstang wrote: I'd just die to see an example of that. I've never seen an example that actually works. I quite distinctly remember reading somewhere (sorry, forget where) that this command was broken. It's not broken. If you find some official documentation that says so, then it needs to be fixed. If you read it somewhere else, then that source is not something you should trust. regexten in sip.conf works just fine; it can easily be used to make an extension 'appear' and 'disappear' from the desired context based on the status of the peer's registration. If that context is then shared among the Asterisk servers (via DUNDi, IAX2 switches or some other technique), then calls to that extension will be handled by the server it registered to automatically. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clustering
On Sun, Mar 12, 2006 at 01:29:01PM -0700, Ron McCarthy wrote: Regarding OSPF, so your saying you have multiple * boxes setup with same exact config and then just have OSPF fail everthing over to the new server if it cant get to it? That makes sense, just never of even thought of doing it that way. Heck, if you want to get real complex just run BGP and you could then setup priorties for each server and all kinds of cool stuff. BGP is not better suited for this then OSPF is. By default it is slower and you simply don't need everything that's in BGP. OSPF with fast hellos is a much cleaner approach, and you can set costs for interfaces. Kristian. Are you then using regexten on all servers so when a * tries to make a call it can find where to go, or are you using something else? Thanks! Ron On 3/12/06, Douglas Garstang [EMAIL PROTECTED] wrote: It doesn't. It's transparent to the user agent. -Original Message- From: Wai Wu [mailto:[EMAIL PROTECTED] Sent: Sunday, March 12, 2006 9:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering How does OSPF tell the remote end (assuming he does not know your setup) start sending RTP packets to the other interface? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Douglas Garstang Sent: Sunday, March 12, 2006 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Clustering No, only if a network interface in the server fails. We have two network interfaces per system (actually we have four, but two are on a private network with a MySQL server). If one of the network interfaces fails, OSPF will switch the default route over to the other interface pretty quick smart. There's probably a little luck involved here too. -Original Message- From: Gabriel Afana [mailto:[EMAIL PROTECTED] Sent: Sat 3/11/2006 10:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Clustering So you are actually able to maintain a call in progress even if the server its connected to fails (by routing to another)? - Gabe - Original Message - From: David Coulson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, March 11, 2006 7:15 PM Subject: Re: [Asterisk-Users] Clustering From what I can find online, OSPF seems to be a technology or method, not necessarily a program. What are you using to perform OSPF? OSPF is a routing protocol. Quagga (quagga.net) is a good open source implementation of OSPF for Unix. David ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk at large
. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Tue, Jan 31, 2006 at 06:29:07PM -0700, Damon Estep wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, January 31, 2006 4:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question I don't know how much 1+1 by you is, but lets recalculate this for a moment: First the bandwidth per channel: http://www.airewaves.com/aire/support/bandwidth_explain.php 1.5mbps (mega *BITS* not BYTES per second) to a full T1, which equals 1536 Kbits, each channel then takes 64kbps. 64*5,000=320,000kbps. 32,000/1,024=312.5 Mbps (round off to Mbps), no where close to a Gb. Every single PC made in the last 4 years I came across, can handle this type of bandwidth. BTW, this all amounts to just over 39 MBYTES per second. 312.5/8=39.0625 Not that I disagree with your point, the bandwidth is not huge, but the math is a little fuzzy; First of all, a g.711u stream over UDP is closer 80k than 64k, the payload is 64k + udp overhead + IP overhead. Now consider that the call is originated as SIP (llok back a few days in the thread), and lets assume the call goes to an external hard or softphone, and lets also assume that there is a reason to keep the RTP stream running through asterisk (monitoring, recording, transferring, dtmf, ability to re-enter IVR, etc). I make all the assumptions safely since the thread was started by someone looking to set up a large call center and I have followed thread out of curiosity. So a 80k full duplex RTP stream originates on media gateway somewhere, hits the asterisk box, is internally bridged, and is sent back out to a phone somewhere. My math says this puts a 160kbps full duplex load in the NIC. Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? No no no. First you come to the conclusion that you have 800Mbps of traffic, but this is bi directional thus 400Mbps in each direction. Then you're comparing you're 800Mbps to 1Gbps. If you compare bi directional you need to count the card as 2Gbps. So you are nowhere near 80% but closer to 40%. To get to a comfortable load you would need 2x GigE NICs (for ~40% utilization), of course now we are adding additional overhead for the bonded NIC trunking protocol. Is still contend this is not practical without multiple very high end servers and round robin call origination from the upstream provider delivered over something like GigE or OCx. Maybe someone will step up and post some real-world application limits based on experience... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Wed, Feb 01, 2006 at 03:38:21PM +0800, Dinesh Nair wrote: On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? additionally, 5000 simultaneous SIP calls at 20ms intervals will send, 5,000 * 50 * 2 = 500,000 packets per second (full duplex). not too many boxes can handle such packet load, in spite of the relatively small packet sizes. Indeed, a FreeBSD machine doing just routing lookups can handle somewhere around 600Kpps. FreeBSD is using a Radix tree for routing lookups, by using Linux you may choose something better performing such as LC-trie where you're able to push quite a lot more. But this is pure routing done in the kernel, with asterisk you have to bring the packets to userspace and back limiting the performance by quite a lot. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Sun, Jan 29, 2006 at 05:24:12PM +1000, Rob Thomas wrote: The question is somewhat ludicrous, and I'm slightly surprised that no-one has sat down and done the maths about bandwidth utilization. So I did. To handle 5000 calls coming in over a PRI, you'd need 210 or so T1s or 170 E1's. Yes. All of those would generate 320Mega BYTES of data per second (eg, 32Gigabit/sec) No. Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds Using g729 it's a lot less. And even if it was 320Megabytes/s that equals a little over 3Gbps not 32. There is no way possible that you're going to pump that amount of data through a PC. Don't care about codecs and dialplans, PC's just don't have that sort of internal bandwidth from peripherals. Well, with your above miscalculations, no. But 3.2Gbps is possible with a few boxes. And there are PCs with at least four different PCI-X buses, that's 4GB/s so it might even be possible with one machine. Anyway, it's still a lot of calls. If you do, honestly, need to handle 5k calls, you'd probably have to have a bank of Cisco 5850s doing the termination - With a max of 5 DS3 (1 DS3 = 28 T1's) into each one, you'll need 8, or probably 9 as you'd want to have one as a hot spare. Each of those DS3's would go into some beefy switching fabric (note, that each one is producting 225mbit) and you'd have some sort of asterisk box with huge internal bandwidth handling each one. Cross connect all 9 asterisk boxes via 10Gbit networks (note, you'll need PCI-16x 10g cards) and have a pair of voicemail servers. I'd suggest a pair of big Sun boxes. Then, of course, you have the issue of getting the calls _out_ of the asterisk machines. You've just doubled your bandwidth requirements, so you'll need to double up on the asterisk machines, and split the network up further. I'd take a guess that you could do it under USD$1million (just for hardware) but I wouldn't be surprised if it was USD$10million. I'm happy to sell you any of this 8-) --Rob -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vic Sent: Sunday, 29 January 2006 1:16 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 5,000 concurrent calls system rollout question Hi, Zoa, yes, these calls are from SIP to SIP. We will have more than 3000 (more like 5000)concurrent calls come into system and we will need to handle them. We will also need an IVR function as well. I am not up to speed on Asterisk yet, so, I am a little bit confused by all the different ways of doing it. Someone is talking about IAX: I think it can only be used between Asterisk servers, right? In this particula rscenario we are getting calls as SIP directly from carrier, so we will not need to do any conversion (I think). We just route the calls to the destination, that's it. Any suggestions on how to proceed? Can Asterisk do it? I read somewhere that it takes about 30 MHz per one voice channel, so if we want to have 5,000 calls, we will need 150,000 MHz? Thats like 50 3 GHz machines... Not going to fly with our people. Or do 30 MHz are only necessary for transcoding? In other words, if it comes in as SIP and we keep it that way, can we make it a bt more feasible number? Zoa [EMAIL PROTECTED] wrote: It can be done, are those 3000 calls sip to sip ? If so it could easily be done, if they are not sip to sip you will need a bunch of servers. Zoa. Vic wrote: Hi, we are currently considering different options for rolling out a large scale IP PBX to handle around 3,000 + concurrent calls. Can this be done with Asterisk? Has it been done before? I really would like an input on this. Thanks! --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717
Re: [Asterisk-Users] How many digium cards per server ?
On Mon, Jan 30, 2006 at 10:39:11AM +0100, [EMAIL PROTECTED] wrote: hi I know people in here use 4 boards, but I believe the only real limitation is the number of PCI slots in your computer. Is this 4 E1 boards? Obviously, with 4x4xE1/T1 boards (480 B channels) you need a pretty fast monster to drag it around due to the way the zaptel/asterisk works. So the actual limit is probably mostly restricted to how fast your PC is. I know other telco engines easily drag 16 E1's, but I am not sure Asterisk can do that even without echo cancel? You have to test... I'd say one of the things you will run into is the number of interrupts coming at you with so many cards. Personally I recommend one card per box. The card goes for $1800(I buy Sangoma with echo cancel) while the computer goes for $800 (HP Proliant 145). So the computer is really the small expense here and with one card per box I instead have several boxes and thus better redundancy. One card per box also guarantess I won't have any interrupt problems and it's capable of transcoding. Kristian. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 5,000 concurrent calls system rollout question
On Mon, Jan 30, 2006 at 12:49:15PM +0100, [EMAIL PROTECTED] wrote: Using G711A (ie, worst case bandwidth wise): it's 64kbit/s not 64Kbyte/s so it's 320Megabits per seconds That will only do if you talk a lot with your mother in law! ;-) For the rest of the conversation (those with both speaking): 5000 * 64k * 2 = 640M Indeed you are correct, I'll defend myself with stating that I presumed we were talkin full duplex ;) It should in theory work with a 1Gbits Ethernet, but you would be counting on ca 65% utilization. I would normally plan with 30-40 % utilization and you need 2 for redundancy anyway. Though now you're wrong ;) 65% isn't correct. If you're counting both in and out traffic you'll have to assume that the Gigg card is capable of 1Gbps in each direction thus 2Gbps in total and 640M of 2000G is about 30% or just as much as 320M is of 1G. I don't know the average packet size of a voice RTP packet but I guess it's quite small. Being a network guy I've dealt quite a lot with software routers and a normal Linux machine can forward about 500kpps, and this is mere forwarding if you run this via Asterisk you should probably split that by ten. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set caller id on Swedish PRI (euroisdn)
On Mon, Jan 30, 2006 at 03:43:18PM +0100, [EMAIL PROTECTED] wrote:
Hi,
I have a problem with setting outgoing caller id to nothing (secret)
on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID
seems to work fine when connecting the same line to a Ericsson PBX - so
something must be wrong in my settings, but I don't know what.
I've tried:
exten = _*70X.,1,Set(CALLERID(name)=) exten =
_*70X.,2,Set(CALLERID(num)=) exten =
_*70X.,3,Dial(Zap/g0/${EXTEN:3}|60|T)
Try setting the Callerpresentation to something else:
http://www.voip-info.org/wiki/page_history.php?page_id=1682preview=2
But the result is always that the caller id is our main number
(A-number).
Here is an from zapata.conf:
[channels]
language=se
context=from-pstn
switchtype=euroisdn
pridialplan=unknown
prilocaldialplan=unknown
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=1.0
txgain=-4.0
group=0
callgroup=1
pickupgroup=1
immediate=no
overlapdial=no
channel = 1-15,17-31,63-77,79-93
group=1
channel = 94-108,110-124
group=2
context=from-internal
signalling=pri_net
channel = 32-46,48-62
Regards,
Jan
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Kristian Larsson, Net At Once AB
Email: [EMAIL PROTECTED]
Phone: +46 470 592717
Cell: +46 704 910401
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[Asterisk-Users] Caller Presentation
Could someone please outline the differences between: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Thank you -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP Router
On Thu, Jan 26, 2006 at 09:42:36AM +0200, Mohamed Farid wrote: Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? The cisco x8xx series are excellent. I have a 2811, if you're routing needs are basic a 2801 should suit you just find you can cram a few VIC2-2FXS in it and get the voice ports you need. It's capable of 3DES and comes in a nice package too. An excellent router. Oh, and it's rather cheap too :) Kristian. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi Examples
On Fri, Jan 20, 2006 at 09:20:43PM -0500, Michael Miller wrote: I have over 50 Asterisk servers geographically distributed in pairs all connected via DUNDi. Contact me off list and I will be happy to describe my experience. I'm also interested in knowing more of this. Why not write to the list so that more people may know about it? Regards, Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No congestion
Hey! I'm having a small problem. I'm using Realtime to store SIP account information. Dialing works just fine, but when dialing a person already on the phone I don't get a busy tone. Eg, Phone 100 calls 200 and they chat with each other phone 150 calls 100, and gets a regular ringing tone what I would is for phone 150 to receive a busy tone since phone 100 is already speking with someone else, how would I go about doing this? Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Network Wire Brand
On Wed, Jan 18, 2006 at 03:44:03PM -0800, calvis wrote: Sorry about the OT thread, but I am sure that someone could give me some advice. Nothing is more frustrated than doing a long cable run and then finding your cable is defective. OK, I have had it with the General Cable brand of network cable that we currently use for 5e cable runs. I am looking for something that is 100 percent reliable for doing cable runs. Does anyone have any recommendations? Panduit, it's kinda like the Rolls Royce of cables. The best one can get but it comes at a price. Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Using *RT for HA purposes was: [Asterisk-Users] Realtime MultipleAsterisk boxes, iaxusers
On Wed, Jan 04, 2006 at 02:46:36PM +, Alistair Cunningham wrote: Peter Bowyer wrote: I was thinking along the same lines, but for a dynamic setup it should be possible to have SER/OpenSER load balance REGISTER requests according to some strategy/metrics, and then forward INVITEs and other call-related traffic to the 'right' back-end server. Probably lots of reasons why this is too complicated, though One being that it must be the device that NAT phones register with that delivers calls to them. Otherwise, the NAT device sees a packet coming from an unknown IP address and drops it (for common types of NAT such as restricted cone). Since SER needs to deliver calls, it really needs to be SER that accepts REGISTERs and holds the registration information. The Asterisks then send calls from phones to the SER heartbeat address for delivery. This is what we do in our ITSP in a box product. It gives us full redundancy and failover with the registration capacity of SER and the features of Asterisk. Could you perhaps be as kind as to give us a few example configurations and some more detailed documentation on how you've done this. I'm very interested in building something similar, right now I'm running one Asterisk but with estimated growth I'll need two and using a SER in front to load balance would be a really nice solution. Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetCallerPres
I'm trying to set caller presentation to prohibited and I'm having slight problems doing it. Using a machine that has a Sangoma facing my Telco works but when using an asterisk that talks to the first machine using SIP it does not work. I suspect that SetCallerPres is not transitive, ie it's not communicated between SIP peers but need to be set at the actual machine having the Sangoma card, correct? Anyone have a workaround for this? How should I set callerpres to prohib when doing SIP to SIP calls? Or when calling via SIP and then out on the PRI, how can I set callerpres on the machine originating the call? Thank you Regards, Kristian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime Multiple Asterisk boxes and rtcachefriends MWI
I'd be very interested in hearing more about this as I am in need of a similar installation. Anyone have a hint? Kristian On Thu, Dec 29, 2005 at 02:14:29PM -0500, Asterisk wrote: I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B. With rtcachfriends=no this can be done, However I then loss MWI and sip show peers plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached. With rtcachfriends=yes I get MWI Sip show peers, However I cannot call phones that register to a different switch. My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus= chanunavaliable I then route the call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more.. I have tried using AGI and dialing the full contact found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air.So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired. Could realtime be altered to add a status filed to the SIP realtime table? Or is there a asterisk configuration option that I missed? This is my first post so please forgive me if I posted this in the wrong list. Many thanks! Doug Gillespie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PRI problems.
I have an Avaya IP Office PBX connected to an Asterisk system via a Sangoma ISDN PRI card. Dialing from the as terisk system into the avaya works just fine but when trying to call from a phone connected to the avaya syste m something goes wrong. After punching the first four digits the Avaya calls out, shouldn't it wait for all di gits and then dial out? If I try to dial a three digit number it waits for a while then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Is there some way to circumvent this using hacks on the asterisk side? -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PRI problems.
On Mon, Jan 02, 2006 at 03:36:57PM +0200, [EMAIL PROTECTED] wrote: On Mon, 2 Jan 2006, Kristian Larsson wrote: I have an Avaya IP Office PBX connected to an Asterisk system via a Sangoma ISDN PRI card. Dialing from the as terisk system into the avaya works just fine but when trying to call from a phone connected to the avaya syste m something goes wrong. After punching the first four digits the Avaya calls out, shouldn't it wait for all di gits and then dial out? If I try to dial a three digit number it waits for a while then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Is there some way to circumvent this using hacks on the asterisk side? Looks like you need to enable overlapdial=yes on the Asterisk side. It will then wait for additional digits sent from the Avaya after the initial ones sent with the SETUP. I did try enabling overlapdial=yes but I saw no real change. Is there any other variable to go with it that I might need to tune? I am quite new to the whole PRI thing. What does it do when setting up a call? First a SETUP and after that it dials? Regards, Kristian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PRI problems.
On Mon, Jan 02, 2006 at 12:09:33PM -0700, Alyed Tzompa wrote: You are not gonna be able to modify this behaviour from the asterisk since in your case asterisk is only receiving the digits from someone else (an Avaya in your case but could be PSTN for instance) Just asked an Avaya support guy and told me you should take a look at the ARS Digit Analysis Table, and modify the Min and Max values to suit your needs. I just found something called digit delay time and digit delay count, it just happens so that digit delay count is set to four, ie the number of digits before the avaya dials out. I can not try it out right now since the system is being used - has to wait for the maintenance window tomorrown evening. But what I find a little strange is that the Avaya works when connected to my telco, shouldn't it be possibly to simulate a telco fully with Asterisk? Thanks Alyed! Kristian. Hope this helps ww6 Return-Path: [EMAIL PROTECTED] Mon Jan 02 05:56:31 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by mail11.webcontrolcenter.com with SMTP; Mon, 2 Jan 2006 05:56:31 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id C0C434092; Mon, 2 Jan 2006 05:55:56 -0700 (MST) Received: from psmtp.com (exprod5mx128.postini.com [64.18.0.42]) by lists.digium.com (Postfix) with SMTP id 20B414082 for asterisk-users@lists.digium.com; Mon, 2 Jan 2006 05:55:52 -0700 (MST) Received: from source ([217.10.96.36]) by exprod5mx128.postini.com ([64.18.4.10]) with SMTP; Mon, 02 Jan 2006 06:55:53 CST Received: from localhost (localhost [127.0.0.1]) by mailfront1.citynet.nu (Postfix) with ESMTP id 892034C08A for asterisk-users@lists.digium.com; Mon, 2 Jan 2006 13:55:53 +0100 (CET) Received: from mailfront1.citynet.nu ([127.0.0.1]) by localhost (mailfront1.citynet.nu [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 17711-06 for asterisk-users@lists.digium.com; Mon, 2 Jan 2006 13:55:53 +0100 (CET) Received: from localhost (unknown [10.1.2.65]) by mailfront1.citynet.nu (Postfix) with ESMTP id 236B64C06A for asterisk-users@lists.digium.com; Mon, 2 Jan 2006 13:55:52 +0100 (CET) X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com Date: Mon, 2 Jan 2006 15:52:51 +0100 From: Kristian Larsson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Mime-Version: 1.0 Content-Type: text/plain; charset=iso-8859-15 Content-Disposition: inline User-Agent: Mutt/1.5.8i X-Virus-Scanned: by amavis at citynet.nu X-Spam-Status: No, hits=-3.8 tagged_above=-999.0 required=999.0 tests=ALL_TRUSTED, BAYES_05 X-Spam-Level: X-pstn-levels: (S:65.75150/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [90/4] Subject: [Asterisk-Users] Asterisk PRI problems. X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] X-SmarterMail-Spam: SPF_None X-Rcpt-To: [EMAIL PROTECTED] I have an Avaya IP Office PBX connected to an Asterisk system via a Sangoma ISDN PRI card. Dialing from the as terisk system into the avaya works just fine but when trying to call from a phone connected to the avaya syste m something goes wrong. After punching the first four digits the Avaya calls out, shouldn't it wait for all di gits and then dial out? If I try to dial a three digit number it waits for a while then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Is there some way to circumvent this using hacks on the asterisk side? -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 03:50:27AM -0800, Luigi Rizzo wrote: On Tue, Dec 20, 2005 at 01:41:30PM +0200, Tzafrir Cohen wrote: ... And this is bad for us. With Gizmo we can talk. With google talk we have stand a chance of talking. But we're blocked from Skype. since you cite it, what compatibility is there with google talk ? any pointer to descriptions of the protocols used ? Google talk is jabber based and they intend to support SIP... Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow?
On Tue, Dec 20, 2005 at 09:54:50AM +0900, [EMAIL PROTECTED] wrote: I sincerely believe that it's completely non-sense to make a channel for Skype. Skype is a *proprietary* protocol. If they(ebay) don't like the idea of someone messing around their network, they will change the protocol specification, launching a new version, for example, and *all* the work and time spent on this will just going to sink. Probably it is better to loose time with something else. I agree. Perhaps put the time into making new cool features so that Skype folks can look at some SIP client and say 'wow - I want that too, let's switch from Skype' ;) The world would be a better place without Skype, without proprietary standards.. Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Avaya system
Just the other day I tried connecting an Avaya IP403 Office IP PBX to my asterisk. The IP403 is currently used for all the phones at our office and it is connected via it's own PRI to the PSTN. Now I have a Asterisk machine with three PRIs used for our SIP services. To be able to utilize our capacity better I would like to let the Avaya connect to the Asterisk and share the three PRIs that it has. So, I connected the Avaya to my Asterisk, configured the Sangoma card to act as the CO side. PRI came up and I'm all happy. I try: dial [EMAIL PROTECTED] and voila it dials the correct extension on the Avaya. I'm even happier! :) Now I try to dial out, after punching the four first digits the Avaya dials out. The asterisk in turn dials out to the PSTN. No matter which number I try it just dials out after the fourth digit. If I punch something shorter, like a three digit number it waits for a while and then dials. Is this some feature to let the CO know of which area code the calls is going ahead of time? Anyone with Avaya knowledge know how to turn this off? Is there some way to circumvent this using hacks on the asterisk side? Thanks Regards Kristian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RealTime and automatic extension registration.
Hi! I was hoping someone might answer a few questions. For a SIP user it is possible to configure something called regcontext and regexten. My understanding is that when the use registers with asterisk it will automatically add an extension for the user in the context specified by regcontext? How would I go about doing this with Asterisk RealTime. I have quite a few users and it would be really great if there could be some form of automatic extension adding. Regards, Kristian. -- Kristian Larsson, Net At Once AB Email: [EMAIL PROTECTED] Phone: +46 470 592717 Cell: +46 704 910401 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Hardware recomendation
How would one go about to implement such a cluster? How do the different Asterisk boxes know of the extensions on all the other boxes? Is each client bound to it's box or can it connect to any box in the cluster, ie if one fails can the other take over and share the load of the failed on between themselves? I would be very interested in hearing more of such solutions and people experiences with it. Regards, Kristian Larsson On Thu, Dec 08, 2005 at 10:00:01AM +0200, Zoa wrote: Yes, transcoding is not going to work for that density. asterisk doesn't do g723, and even if it would your system would not be able to handle more than 150 simultaneous g711 to g729/g723 transcodings. If you would go for plain g711, you could do 500, but i don't recommend it, especially if you have little asterisk experience. (i'd say go for a cluster). Zoa www.asteriskguru.com Krystian Filiks wrote: I will be using IP Hard and soft phones all the way, so everything will be on Ethernet, for this I want 1Gbit incoming and 1Gigabit outgoing, looking for atleast 500 simultaneous calls, with 2 3.6Ghz processors I think I could squize out more then that. For codec I want to use g711 on the outgoing as it will only be over local lan and just about 2 meter away from the termination point (so almost 0 in loss) as for incoming I think g.729 or 723 maybe GSM. I know that the recoding take the most of the CPU power so perhaps I can do g.7xx codec all the way, that is a mather of test and see. No other cards in the box then LAN cards. On top of that I'll run voicemail, text to speech and music on hold. Any comments? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: den 8 december 2005 02:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Hardware recomendation Krystian - what kind of port density are you aiming for? Will you be running analog or digital? Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Krystian Filiks wrote: Hello asterisk people! I have been running a test * server a P III box for some time now and it's been rock stable. Now I'm looking to build a production system with as big capacity as possible on 2 Xeon 3.6Ghz processors. I'm wondering what you are thinking about Supermicro 6014H-32 SuperServer with Dual 3.6Ghz Xeon processors and 2M casche each, 2 X Gigabit LAN ports, 1Gb of RAM and about 80Gb of SATA HDD. For the OS I was thinking about Debian and the latest stable release of Asterisk. I will be using IP to IP technology without any PRI cards only IP to IP. Clients will be using SIP and Aserisk will terminate on to H.323 or possibly SIP How can I benchmark this thing (Aprox) without having to buy the server? Has any one had any experience of such server? Please comment. --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancellation
I am having problems with echo, first let me explain my setup: I have a Gateway box, which basically is an Asterisk with a PRI card. It's only job is to interface with 2 incoming ISDN PRI connections. Then there are two other asterisk boxes to which my users are registered. Dialing from a phone it hits the first asterisk which forwards it to the gateway box and then on to the PSTN. What are the general causes of echo? When calling from my SIP phone I hear no echo but the other end, the PSTN end, hears a lot of echo. What could cause this? Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Complicated Dialing plan routing
Hey everyone, I'm having a slight problem with my dialplan which I was hoping you could help me with. First let me explain the scenario; I have a few hundred different customers split into a few different area codes. What I want to allow them to do is to call each other normally, ie with area code and if the called party is within their own area code they should be able to call them without first typing in the area code. If the number dialed does not exist in my dialing plan it should go out through a PRI extension to the PSTN. Ie. 0470-112233 should be able to call 0470-445566 by calling either 0470445566 or 445566 and if the extension 0470445566 doesn't exist it should try the PRI. I have tried using the Goto command to jump between different contexts for the different area code, but if I use Goto(0470,445566,1) and 445566 doesn't exist it doesn't go out the PRI which it should. I've also experimented with using include and different contexts which doesn't seem to work either. So far the only way I've made it work is to have two extensions one for 0470445566 and one for 445566 and a lot of different contexts... I would like to avoid this and just have the extension in one place and instead use some smart dialing plan to compensate for area codes and the alike. I presume someone else is already doing what I want to do and perhaps could share their knowledge. Let me also mentioned I've searched the list archive but came up empty handed. Please CC me as I'm not currently on the list and thank you. Regards, Kristian Larsson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + TDM04b trouble
I have a an Asterisk server running asterisk 1.0.3 and a TDM04b card. I'm having a problem with my setup. Incoming and outgoing calls are working to 95%. When the other party hangs up their phone after I've hang up mine it starts ringing in my phone. example: 1. I get an incoming call 2. I answer and talk a bit 3. We say goodbye and I hang up the phone 4. The person at the other end hangs up his phone 5. My phone starts ringing again. If I pickup I only get a dial tone. If the other party hangs up before I do there is no problem. I live in Sweden and so I use what I think is Swedish settings for everything. It would be nice if some could provide me with the regional settings for Sweden. Oh and what does the following mean: Jan 22 00:49:46 NOTICE[10995]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'se' Regards, Kristian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IP Phones
Cisco came up with PoE before the standard was set and so it differs. The polarity is switched, so using a dumb power injector and a crossed cable one could make it work anyway. Quoting Julio Arruda [EMAIL PROTECTED]: Keith Burns wrote: I think you need to look at a few other factors. ... 2. Line power - Cisco uses one standard, other phones use another... but Cisco is the 900# gorilla in the powered switch market... your call... I'm curious about this point.. Most if not all vendors that support PoE are not already support 802.3af standard ? ... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
