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Jerk McJerkface posted:EDIT: I just checked with him, can you send me your email/contact info so I can pass it to him and he can contact you with a quote? I PM'd you my info last week, but haven't heard anything yet. Let me know if you don't get PMs and want the information another way?
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| # ? Mar 22, 2010 16:04 |
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| # ? May 19, 2013 07:27 |
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markus876 posted:I PM'd you my info last week, but haven't heard anything yet. Let me know if you don't get PMs and want the information another way? I passed it to him, let me ask him, he's like thirty feet away. EDIT: He said he'll call you today. Jerk McJerkface fucked around with this message at Mar 22, 2010 around 16:11 |
| # ? Mar 22, 2010 16:09 |
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Is Trixbox considered a more mature setup than AsteriskNow ? I'm happy learning everything bit by bit (lots of reading the O'Reilly book already), but having an existing system that works to mess about with and alter is really useful. I found the ANow installation I used didn't have working MoH by default (haven't figured out what's stopping it working), IAX dropped out after a while and hasn't come back up, and a few other minor niggles. Loving the system, the language used in the .confs is pretty straightforward for the majority and shows you can do some really cool stuff. Was shocked how quickly we were able to route a few calls between some softphones and set up VM etc, intrigued about setting up a fake radio station setup with call screening, queues etc just to see what kinda work goes into that. edit: Just to add extra confusion, I've grabbed the Druid latest build as well. I take it its similar again, but uses it's own GUI? Anything particularly good or bad to differ to the competition? The pHo fucked around with this message at Mar 22, 2010 around 17:35 |
| # ? Mar 22, 2010 17:15 |
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ab0z posted:Ok I ran that command from the asterisk cli, how do I retrieve the log? Anyone?
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| # ? Mar 23, 2010 20:11 |
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ab0z posted:Ok I ran that command from the asterisk cli, how do I retrieve the log? Sorry, I've been busy. Do this: edit /etc/asterisk/logger.conf find a line that looks like this, won't be exactly, but it'll get you close. messages=dtmf, console, error, verbose, and add "debug" to the end of it. Then do asterisk -r logger reload Then with debugging on, generate some problem calls. Then go to /var/log/asterisk/messages (messages is the log file) you can get the data there. All this assumes you run Centos, I'm not sure where Ubuntu logs to, but you should be able to find that pretty easy.
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| # ? Mar 23, 2010 20:40 |
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Jerk McJerkface posted:Sorry, I've been busy. Thanks! I'll spend some time on this tomorrow at work. Please don't take this as me being pushy, I just didn't want it to get lost on the last page. I'm super grateful for your help!
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| # ? Mar 24, 2010 04:43 |
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Jerk McJerkface posted:Then go to /var/log/asterisk/messages (messages is the log file) you can get the data there. Should be the same directory. It's the same in Debian.
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| # ? Mar 24, 2010 07:45 |
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ab0z posted:Thanks! I'll spend some time on this tomorrow at work. No worries, I just get swamped and forget about my thread.
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| # ? Mar 24, 2010 11:52 |
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Jerk McJerkface posted:No worries, I just get swamped and forget about my thread. I think everyone who has been helped by you in this thread agrees with me when I say thank you for still keeping current with this after nearly two years and 700 posts.
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| # ? Mar 24, 2010 14:42 |
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ShizCakes posted:I think everyone who has been helped by you in this thread agrees with me when I say thank you for still keeping current with this after nearly two years and 700 posts. Thanks. I'm mulling around opening a new thread and making a better OP.
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| # ? Mar 24, 2010 14:53 |
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markus876 posted:I PM'd you my info last week, but haven't heard anything yet. Let me know if you don't get PMs and want the information another way? You spoke to our guy right?
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| # ? Mar 24, 2010 15:37 |
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Nevermind, I've got what I need. If anyone else struggles with Flite for a month, make sure you're not using asterisk 1.6.
Briantist fucked around with this message at Mar 24, 2010 around 21:40 |
| # ? Mar 24, 2010 19:51 |
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Hey, I've got a quick question. Any recommendations on decent power over ethernet equipment for a small deployment that won't break the bank? I've never worked with PoE stuff before so I have zero first-hand knowledge of what's good, and I always trust experience over random reviews on the Internet. Just a small 16-port switch, unmanaged is fine.
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| # ? Mar 25, 2010 09:40 |
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I run on Cisco where they'll pay for it, Linksys where they won't, and Netgear when a tiny cheap one is needed. Netgear's 50/50 switches (half the ports are PoE, half aren't) are dirt cheap for unmanaged and very quiet, but the largest they sell those in is 16 port (8 PoE). I'd either go with two of the Netgear 16/8s attached together or a Linksys 24 port. I think Linksys also makes an 8 port PoE with two gig ports, so you could link two of those together and not have the 100mbit bottleneck the dual Netgear solution has.
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| # ? Mar 25, 2010 18:17 |
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I'd just like to note that those Netgear 16 port switches run very warm. My coworker has 2 and foolishly stacked one on top of the other and the one on the bottom just burned out the PoE part of the thing about a week ago (the switch itself still works fine, just no more PoE). Mind you we pretty much run them fully loaded power-wise so that probably works them fairly hard. For the record the model that we use is Netgear FS116P.
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| # ? Mar 25, 2010 18:39 |
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Any of the others here that do this for a living dealt with Packet Island (now Broadsoft PacketSmart)? They're a combination of these little ATA-sized boxes that plug in to the customer LAN and a hosted service. The devices either sit on a mirrored switch port or sit inline anywhere on the voice network and monitor a bunch of details about call quality, or they can also generate calls for preinstall testing. I just got my hands on a pair of their endpoint devices and a subscription to the service, just wondering if anyone's had any experience with them that might be useful ahead of time. I will post a review of some kind in a few weeks once I've gotten used to them.
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| # ? Mar 29, 2010 19:39 |
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Jerk McJerkface posted:edit /etc/asterisk/logger.conf That line is commented out: code:
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| # ? Mar 30, 2010 13:22 |
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ab0z posted:That line is commented out: There may be a file called /var/log/asterisk/full that has a log in it, but I'm not sure how that defaults.
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| # ? Mar 30, 2010 13:51 |
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I'd just like to say I've definitely benefited from reading this thread, lots of good info and knowledgeable people. Thanks. I'm in charge of implementing some form of phone system for a SMB that only needs 10 lines. I'm having a really hard time justifying the need of an actual PBX vs. a hosted pbx for our needs. Can anyone give me a rundown on what we would be missing out on by going with an inexpensive hosted PBX? I was currently thinking about skype for business if I'm honest, the features we need appear to be there and the price is right.
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| # ? Mar 31, 2010 00:52 |
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coinstarpatrick posted:I'd just like to say I've definitely benefited from reading this thread, lots of good info and knowledgeable people. Thanks. Hosted PBX is fine, a lot of companies use it, and we're prepping offering that as a business model, actually. You won't loose any features, because your running the same exact software as if you had it in house. The only problems you'll run into are internet based, so make sure you have a stable internet connection that you route over, like a T1, or P2P, although for the cost of a P2P you can probably buy your own PBX.
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| # ? Mar 31, 2010 10:25 |
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I briefly looked into hosted asterisk boxes and they seem way more expensive compared to other hosted services like web hosting or a VPS. What is a good price on hosted asterisk? Is it a standard practice to charge per-user licensing?
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| # ? Mar 31, 2010 12:39 |
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Cock Democracy posted:I briefly looked into hosted asterisk boxes and they seem way more expensive compared to other hosted services like web hosting or a VPS. You know what, I have no idea on the average price, but it is typical to charge per seat. But, just so you know, on a regular inhouse PBX maintenance charges are really high, probably somewhere around 50 bucks per seat per month.
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| # ? Mar 31, 2010 12:48 |
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Should I be able to use this AT&T 3G PCMCIA Ultra Unlocked GSM Data/Air Card - OEM in my Asterisk server as a voice line with a suitable PC-Card to PCI adapter? If it complicates matters I'm running FreeBSD 8
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| # ? Apr 15, 2010 13:43 |
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roadhead posted:Should I be able to use this AT&T 3G PCMCIA Ultra Unlocked GSM Data/Air Card - OEM in my Asterisk server as a voice line with a suitable PC-Card to PCI adapter? No, the data cards have no voice hardware.
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| # ? Apr 18, 2010 01:34 |
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Anyone have any experience with or comments about the Aastra 6731i phone? I'm looking at that one for a small system, 5 endpoints and a server, with a PoE switch to power the phones.
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| # ? Apr 18, 2010 01:43 |
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gruvmeister posted:Anyone have any experience with or comments about the Aastra 6731i phone? I'm looking at that one for a small system, 5 endpoints and a server, with a PoE switch to power the phones. We sell alot of them, they are very popular. I actually have one on my desk. Just a couple things: 1) They don't come with power adapters (but you have POE so it's a non-issue) 2) The speaker phone is duplex, but for some reason the newest firmwares broke that so the speaker cuts out if you talk or from background noise. It didn't happen in firmware 2.4.X, but in 2.5.X its really bad. We've been working with Aastra to fix it, and I think 2.5.2 was ok, but in the latest ones it's back for some reason. 3) They don't have a headphone jack. Interestingly there's a 2.5mm jack for one, but the jack does nothing. 4) The eight buttons up top, 2 of them can't be changed. They have to be save and delete. We complained to Aastra and I think they unlocked them in the newest firmware, but I haven't had a chance to test one.
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| # ? Apr 18, 2010 03:23 |
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Thanks for the input, sounds like a great match. Headphone jack isn't important, and neither are power adapters for this system. I can buy them separately if needed though, right? Also, if I were to buy a batch from CDW today, do you have any idea what version of the firmware would be on them?
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| # ? Apr 18, 2010 04:09 |
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wolrah posted:No, the data cards have no voice hardware. Ok, good thing it was only $20. Can I tether a GSM Handset to the machine via USB/bluetooth/line-in possibly?
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| # ? Apr 18, 2010 06:56 |
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roadhead posted:Ok, good thing it was only $20. Bluetooth. It might also be possible to combine some 2.5mm-3.5mm analog adapters and either bluetooth or USB dialing control with a console channel, but it will almost certainly be more work than straight Bluetooth.
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| # ? Apr 18, 2010 17:14 |
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gruvmeister posted:Thanks for the input, sounds like a great match. Headphone jack isn't important, and neither are power adapters for this system. I can buy them separately if needed though, right? Also, if I were to buy a batch from CDW today, do you have any idea what version of the firmware would be on them? You can buy the power adapters, yes. I've been getting them with 2.4.0 firmware (I think) but you can get all the firmwares off their website.
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| # ? Apr 21, 2010 02:08 |
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My trunk is not registered, that is why incoming calls don't work. "sip show registry" reports 0 SIP registrations. My "Register String" field for the trunk is blank, I'm guessing this is related. The provider just plugs their ears and sings really loud if you try to talk about asterisk/trixbox with them, so I can't get any help there. Their helpful setup document (3 page powerpoint that looks like a 9 year old made it) gives these detailed instructions: quote:{provider name}.net by default supports the following codecs: Based on this information, what should my register string be?
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| # ? Apr 22, 2010 15:24 |
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I just got thrown into a new client who uses Asterisk, I'm excited to learn! Thanks for this thread.
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| # ? Apr 27, 2010 21:10 |
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ab0z posted:My trunk is not registered, that is why incoming calls don't work. any input?
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| # ? Apr 29, 2010 14:01 |
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It doesn't look like you need a registration string. I'm not an Asterisk/PBX guy, but from listening to the people around here, it sounds like you have a static trunk, and they just send SIP traffic to the IP they have on file (I assume you gave them your IP?) You'll have to set your firewall to pass SIP to Asterisk and set Asterisk up to handle the SIP as it comes in. At least, I hope they have your IP, since if they don't do registration/user/password, I don't know how else they'll know where to send the invites. Take a Wireshark capture to see if they're even sending you invites, I think would be the first place to start. Other than that, I'm no help.
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| # ? Apr 29, 2010 14:29 |
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Panthrax posted:It doesn't look like you need a registration string. I'm not an Asterisk/PBX guy, but from listening to the people around here, it sounds like you have a static trunk, and they just send SIP traffic to the IP they have on file (I assume you gave them your IP?) You'll have to set your firewall to pass SIP to Asterisk and set Asterisk up to handle the SIP as it comes in. At least, I hope they have your IP, since if they don't do registration/user/password, I don't know how else they'll know where to send the invites. Take a Wireshark capture to see if they're even sending you invites, I think would be the first place to start. Other than that, I'm no help. Ok, that makes sense. I'll check our configs. I thought we forwarded ports when we first set up the server, but that may have changed. What range - 5060-?, TCP?
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| # ? Apr 29, 2010 14:35 |
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Default SIP signaling is UDP 5060. Media is UDP and can be anywhere 1024-65534, usually even ports only, but the provider might give you a smaller range, depending.
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| # ? Apr 29, 2010 17:52 |
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Panthrax posted:It doesn't look like you need a registration string. I'm not an Asterisk/PBX guy, but from listening to the people around here, it sounds like you have a static trunk, and they just send SIP traffic to the IP they have on file (I assume you gave them your IP?) You'll have to set your firewall to pass SIP to Asterisk and set Asterisk up to handle the SIP as it comes in. At least, I hope they have your IP, since if they don't do registration/user/password, I don't know how else they'll know where to send the invites. Take a Wireshark capture to see if they're even sending you invites, I think would be the first place to start. Other than that, I'm no help. That's actually a really good start. I've been wicked busy, so I haven't been working on this thread, but I'll try to answer this guys question. The way Sip registrations work is that the peer (your PBX) registers with the provider (the SIP trunk) using a username and password. The provider then logs what IP you come from, so it knows where to send the calls when one comes in for you. If they don't require registration, then they have to know what IP you are coming from, if not, then anyone can connect to their IP and send calls. If you can make calls out, but can't take calls in, then it's almost 100% a firewall issue. Install tcpdump on the box, and do a trace and then open it with wireshark and see if you are getting anything from the provider.
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| # ? Apr 29, 2010 22:28 |
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Why deal with a provider that won't support asterisk? I'd drop em and use something like vitelity or flowroute, both of which provide asterisk setup guides that show you exactly how to set it up, customized with your account information.
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| # ? Apr 30, 2010 12:35 |
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This particular provider is a mediocre local DSL company which we for some reason used to resell for. After years of underwhelming product and unimpressive support in the internet services category, naturally the thing to do would be to move our business critical voice communication service to them.
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| # ? Apr 30, 2010 13:46 |
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| # ? May 19, 2013 07:27 |
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We're looking to upgrade our lovely old phone system to something better. We have 6 analog lines, I believe, and about 17 people, probably 20+ soon. I looked at Digium's website and got a quote, but I'm not sure if that's the best option. Nobody here has any Asterisk experience. Jerk McJerkface, can you put me in touch with your sales guy or help me out here? We need something that's easy to use/customize, and we don't have to call stupid goons to come out and fix every drat week like our current phone system (some ghetto Samsung thing or something, I'm not even sure).
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| # ? May 7, 2010 00:06 |














