| 00:06.52 | *** join/#asterisk f2knight (~ben@c-98-246-74-16.hsd1.or.comcast.net) |
| 00:07.15 | f2knight | Q: Hi guys |
| 00:07.20 | batfastad | To receive events notifications in my PHP script through AMI, do I have to keep that PHP script perpetually connected to AMI? Eg: without issuing the Action: Logoff command and closing the socket? |
| 00:08.06 | f2knight | batfastad, not a lot of exp with the AMI, but my understanding of it is that you must be connected to get anything out or in to it. |
| 00:08.58 | f2knight | Q: Does anyone use Flowroute? |
| 00:10.12 | *** join/#asterisk LiENUS (quake@209.168.144.219) |
| 00:10.13 | batfastad | f2knight: that's what I'm thinking too. I can only find documentation about sending commands and receiving the immediate response. No documentation about hanging around waiting for events to be received |
| 00:10.23 | LiENUS | hey there |
| 00:10.50 | LiENUS | if i wanted to interface a large number of phones to an asterisk server and have it so you can dial an extension to reach a specific desk |
| 00:11.01 | LiENUS | what hardware am i looking for? they are regular plain ole telephones not voip |
| 00:12.24 | LiENUS | do i just need a digium analog card with enough ports to give one to each desk? ie if i have 48 phones id need two 24 port modular analog cards? |
| 00:13.45 | carrar | Might be cheaper to replace the phones |
| 00:14.00 | carrar | and use used voip phones |
| 00:14.26 | carrar | then to go buy channelbanks |
| 00:14.50 | LiENUS | carrar, but am i right is that the hardware needed? |
| 00:15.15 | *** join/#asterisk jayson_r (~jayson@71.76.46.81) |
| 00:15.20 | carrar | I would probably use a T1 card and a ADIT600 channelbank |
| 00:15.22 | carrar | but yes |
| 00:15.27 | carrar | that woudl work also |
| 00:15.33 | *** join/#asterisk JonathanRose (~Jon@207.98.203.74) |
| 00:16.01 | LiENUS | oh theres a diff alternative? |
| 00:16.20 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
| 00:16.55 | carrar | You can use Cisco (was linksys) ATA's (SIP to Analog) devcies |
| 00:16.57 | f2knight | batfastad, My understanding would be that you would have to have a daemon running (a small server app) polling. Much the same way you do with FastAGI.. though again I am not a expert nor have I used AMI so just my light understanding of it |
| 00:17.15 | carrar | LiENUS, there are lots of ATA options |
| 00:17.15 | LiENUS | carrar, well im curious bout the channelbank option |
| 00:17.25 | LiENUS | ATA stands for? |
| 00:17.29 | LiENUS | analog terminal adaptor? |
| 00:17.33 | carrar | channel back is gonna be more expensive probably |
| 00:17.36 | LiENUS | telephony? |
| 00:18.00 | carrar | but is more of a perm solution in my mind |
| 00:18.18 | f2knight | LiENUS, as carrar said cheap IP Phones,. Grandstream 502's are pretty affordable under $50 each if you do not need any features on the phone, |
| 00:18.21 | LiENUS | well its for a guy who joined an networking related channel and when i suggested he try here he said he didnt want to try asking you guys and said you were jerks even though hes never been in here heh |
| 00:18.26 | carrar | again would be cheaper to replace phones |
| 00:18.31 | LiENUS | so id rather stear him to an expensive option :P |
| 00:18.41 | LiENUS | steer* |
| 00:18.52 | carrar | steer him to replacing phones |
| 00:19.02 | LiENUS | yeah but thats cheap |
| 00:19.05 | carrar | You can get cheap cisco & polycom phones used |
| 00:19.17 | carrar | it's not "cheap" |
| 00:19.19 | carrar | it's cheaper |
| 00:19.25 | carrar | the phones will work better |
| 00:19.30 | carrar | then using analog phones |
| 00:19.34 | carrar | less issues |
| 00:19.36 | LiENUS | i mean cheap is in costs less on his pocketbook |
| 00:19.39 | carrar | less echo probably |
| 00:20.12 | LiENUS | 24 voip phones at 50$ a pop is 1200$ a 24 port digium analog interface card is 1600 |
| 00:20.24 | carrar | plus the hardware |
| 00:20.29 | carrar | to connect the phones |
| 00:20.34 | carrar | breakout strips |
| 00:20.52 | f2knight | LiENUS, If you want to make him 'pay' buy NEW Cisco phones, Cisco Switches and Call Manager Then have him pay for you to go to school to learn how to do it all |
| 00:20.52 | LiENUS | hell im betting he wouldnt even have to run new cable to switch to voip, chances are hes got cat5 at least in the walls |
| 00:21.20 | carrar | can use those phones on your existing network |
| 00:21.44 | f2knight | LiENUS, The 'better' method is to pop in Cheap sip phones, and asterisk box, |
| 00:22.05 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 00:22.05 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 00:22.18 | LiENUS | carrar, yeah really at this point im thinking bout my own work and how they keep asking me rather than our phone guys to get us an incoming call volume number |
| 00:22.23 | f2knight | LiENUS, then decide how you want to trunk calls to the outside world. T1 = Contract and commitment. VoIP = Cheap instant setup |
| 00:22.27 | LiENUS | since its some integrated toshiba pbx system i have no clue how to |
| 00:22.55 | batfastad | f2knight: And I don't suppose there's any pre-written daemons out there to do this are there? I could write something in PHP but it probably won't be the most bullet-proof or efficient thing in the world |
| 00:22.58 | LiENUS | would be nice to use asterisk instead |
| 00:22.59 | f2knight | LiENUS, if you go T1 get an external T1 device something like redbox. it will make life a lot easier and you can make him 'pay' for it. |
| 00:23.41 | f2knight | batfastad, there is 'nothing' written that I know of .. there are tool kits to help you write them. but each server will be unique so you will have to build it for yourself. |
| 00:24.08 | carrar | LiENUS, I would stay with Digium cards all the way |
| 00:24.20 | batfastad | f2knight: Although thinking about it, that's probably what op_server.pl is used for in the Flash Operator Panel package. Wonder if there's anything in there that could help... will run off to check |
| 00:24.23 | f2knight | batfastad, I use Python. and have a cusomized version of pyst that I use because i find it 'EASIER' to use then learning the more 'hardened' framework. |
| 00:25.37 | LiENUS | carrar, i think if i do this at work im going with voip |
| 00:25.52 | carrar | it's the best way |
| 00:26.09 | f2knight | LiENUS, if you are using legacy PBX then yes just use some digium cards to interface to it.. Or AudioCodec FXS/FXO device are really nice to work with supper easy too |
| 00:26.12 | LiENUS | though i do have a fax machine |
| 00:26.16 | LiENUS | how would you handle that? |
| 00:26.29 | carrar | T.38 at 9600 baud |
| 00:26.31 | LiENUS | rather i have a requirement of a fax line to act as a backup |
| 00:26.32 | carrar | over SIP |
| 00:26.37 | LiENUS | really, sip |
| 00:26.45 | carrar | if you have a SIP carrier that can support it |
| 00:26.49 | carrar | otherwise |
| 00:27.00 | LiENUS | well i think we're actually going to be bringing in analog lines |
| 00:27.10 | LiENUS | our credit card machines and fax machine will all share the same line |
| 00:27.15 | batfastad | f2knight: Been meaning to learn python for years as it always seems like it should be a bit more "consistent" for tasks like this as it has a more fundamental set of functions. is that a correct assumption? |
| 00:27.18 | carrar | use a T1 to a digium T1 card and another T1 port to a analog channelbank for faxmachines |
| 00:27.20 | f2knight | I recommend just using a single line for your fax from the local Bell. works 100% all the time. |
| 00:27.30 | carrar | it works very well |
| 00:27.36 | LiENUS | credit card machines will actually be running over internet but in case it goes down it'll use fax line |
| 00:27.38 | carrar | but then you have a T1 from a carrier |
| 00:27.51 | LiENUS | f2knight, so not even bother running it through the pbx? |
| 00:28.15 | f2knight | batfastad, I like it because it is very 'logical' to me. and is easy to understand. You can also do in about 3 lines of simple code what it takes about 10 to do in java or php |
| 00:28.45 | f2knight | LiENUS, if you are not using asterisk for a PBX then no don't bother. Just get a gateway device. |
| 00:28.58 | f2knight | LiENUS, big quesiton is how are you getting your trunks.? |
| 00:29.03 | LiENUS | pots most likely |
| 00:29.08 | LiENUS | for simplicity |
| 00:29.18 | LiENUS | might actually do t1 but doubtful |
| 00:29.30 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 00:29.34 | carrar | pots != simplicity |
| 00:29.43 | f2knight | LiENUS, then you need to have a device that acts as FXS to answer and FXO to send the calls to the legacy pbx. |
| 00:29.53 | LiENUS | carrar, for 8 lines? |
| 00:30.01 | carrar | For any lines |
| 00:30.17 | batfastad | f2knight: I might give it a try again then. Last time I was put off I think as it didn't seem to have the same wide array of functions as PHP... it looked like you had to build your library of functions yourself. Is that right? |
| 00:30.21 | f2knight | LiENUS, IF you are using SIP to get and place calls you can use just a single FXO device interfacing to an asterisk box you host someplace or sometimes even right to your provider |
| 00:30.45 | LiENUS | carrar, its cheaper than going t1 route heh |
| 00:30.55 | f2knight | batfastad, depends on what 'functions' you are using. :) |
| 00:30.56 | carrar | and cheaper is ALWAYS better |
| 00:30.56 | LiENUS | the only way t1 ends up comparable in price is to go integrated |
| 00:30.57 | carrar | haha |
| 00:31.06 | carrar | best of luck |
| 00:31.13 | LiENUS | well hows pots going to be more reliable than t1? |
| 00:31.22 | LiENUS | err not pots |
| 00:31.26 | LiENUS | hows t1 going to be more reliable than pots |
| 00:31.43 | carrar | It's going to less problematic |
| 00:31.47 | carrar | in a number of ways |
| 00:31.52 | LiENUS | its the same copper coming in |
| 00:31.56 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 00:32.01 | carrar | You best read up |
| 00:32.14 | LiENUS | more of it for 8 phone lines than a t1 but its running over the same copper between us and the co |
| 00:32.22 | f2knight | LiENUS, T1 / well PRI, lets you modify callerID you get SLA agreements, POTS you do not. If you have more then 14 lines you are most often better with a T1... that is 14 lines that need to be used at once. |
| 00:32.32 | LiENUS | also with our present t1 average time to fix is... |
| 00:32.34 | LiENUS | 3 months |
| 00:32.36 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:32.41 | LiENUS | but we got a 100$ refund |
| 00:32.56 | LiENUS | and i got cussed out at 8am when our rep called my cell on my day off |
| 00:32.57 | f2knight | LiENUS, T1 have top priority on repairs over POTS |
| 00:33.06 | LiENUS | f2knight, lol you should tell at&t that |
| 00:33.26 | carrar | I would never recommend a biz goign with pots line, Always SIP over a QOS Circuit or a PRI |
| 00:33.55 | LiENUS | carrar, well pots lets me drop my phone lines and replace them with SIP cheaply |
| 00:33.59 | f2knight | LiENUS, carrar is right either run it as SIP, or hit the T1/PRI other wise stay home |
| 00:34.17 | LiENUS | and t1 for it to be in our budget wed need to do internet and phones over t1 |
| 00:34.20 | f2knight | What is the call volume like? |
| 00:34.29 | LiENUS | which results in lovelyness when the t1 goes out |
| 00:34.32 | carrar | LiENUS, then why bother with Asterisk |
| 00:34.38 | carrar | just keep analog |
| 00:34.44 | f2knight | pretty much |
| 00:34.54 | LiENUS | carrar, because i'm constantly getting asked to pull our call volume out of the existing pbx |
| 00:35.07 | carrar | wah |
| 00:35.13 | f2knight | if your budget is 350 a month.. the average cost of a bundled t1 then you need to reevaluate your phone system |
| 00:35.21 | LiENUS | 350$ a month? |
| 00:35.34 | LiENUS | where in gods name do you live? |
| 00:35.38 | f2knight | US |
| 00:35.44 | LiENUS | we're paying over 500 a month and thats on the cheapend for this area |
| 00:36.21 | f2knight | My T1 on the east coast costs me 650 a month with over 80 numbers parked on it. |
| 00:36.31 | carrar | LiENUS, you need to do some research on prices and write up a plan with costs and present it to your work |
| 00:36.34 | LiENUS | yeah ours is 500$ a month with 8 phone numbers |
| 00:36.43 | f2knight | my T1 on the west cost with only 48 numbers on it costs 462 mo. |
| 00:36.51 | LiENUS | with a sla that means didly |
| 00:36.57 | LiENUS | we had intermittent outage over a period of 3 months |
| 00:37.08 | LiENUS | every time we called up to report it theyd blame it on our pbx |
| 00:37.19 | carrar | make them proove it |
| 00:37.22 | LiENUS | after 3 months they finally took a look and turns out it was on a bad pair |
| 00:37.30 | LiENUS | carrar, well eventually they addmitted it was their fault |
| 00:37.33 | LiENUS | and refunded us 100$ |
| 00:37.44 | carrar | so then SLA != diddly? |
| 00:37.57 | LiENUS | and shortly after they fixed it |
| 00:38.00 | f2knight | you need to ready your SLA carefully |
| 00:38.08 | LiENUS | not shortly after, rather after i got them to refund it |
| 00:38.11 | LiENUS | which took another 3 months |
| 00:38.16 | LiENUS | the rep who handled refunding it |
| 00:38.22 | LiENUS | called my cell at 8am on my day off to cuss me out |
| 00:38.23 | carrar | just imagin if you had noise or echo on your pots lines |
| 00:38.27 | carrar | haha |
| 00:38.32 | carrar | good luck getting that fixed |
| 00:38.39 | f2knight | if you are using a CLEC, your SLA is mostly saying if the circuit is down they will credit you UP TO so much a day if it is out for 24 hours. |
| 00:39.21 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 00:39.40 | LiENUS | f2knight, part of the issue is we have no choice |
| 00:39.43 | f2knight | it really sounds like your whole reason for the ASterisk box is to make it easier for you to do CDR ? is that right? |
| 00:40.12 | carrar | stick a asterisk box inline of your existing PRI |
| 00:40.18 | carrar | so it can do accounting |
| 00:40.20 | LiENUS | f2knight, thats how im going to sell it to the bosses, ultimately id like to quit tying us to the same copper |
| 00:40.42 | f2knight | and your lines come in now over pots? |
| 00:40.43 | LiENUS | f2knight, our sla sucks |
| 00:40.45 | f2knight | how many |
| 00:40.51 | LiENUS | all our lines right now (8) come in over t1 |
| 00:41.11 | LiENUS | our sla effectively says if they can get 80% of ordered throughput they consider it provisioned and we have to pay the agreed price |
| 00:41.16 | f2knight | t1 or a bonded t1? do you get internet over that same t1`? |
| 00:41.27 | LiENUS | along with a lot of other crap |
| 00:41.33 | LiENUS | t1 integrated so internet is over the same t1 |
| 00:41.43 | f2knight | okay so you DO NOT HAVE a t1 |
| 00:41.54 | carrar | he DOES have a T1 |
| 00:42.01 | carrar | it's muxed |
| 00:42.09 | f2knight | sorry carrar, that not a real t1. :) |
| 00:42.10 | LiENUS | we don't? |
| 00:42.18 | f2knight | it a marketing ploy |
| 00:42.18 | carrar | T1 by definition |
| 00:42.20 | carrar | yes |
| 00:42.24 | carrar | he has one |
| 00:42.29 | f2knight | t1 by def is 24 pairs |
| 00:42.33 | LiENUS | it is? |
| 00:42.37 | f2knight | digital 64k trunks. |
| 00:42.37 | carrar | 24 pairs? |
| 00:42.37 | LiENUS | i thoguht t1 by def was 2 pairs |
| 00:42.38 | carrar | hahah |
| 00:42.40 | carrar | WTF |
| 00:42.41 | LiENUS | its 24 channels |
| 00:42.51 | carrar | no |
| 00:42.53 | LiENUS | thought* |
| 00:42.54 | carrar | thats a voice T1 |
| 00:43.07 | carrar | T1 is just signaling |
| 00:43.14 | f2knight | and a data t1 is 1.5mb 64k * 24 |
| 00:43.20 | carrar | ESF/B8ZS |
| 00:43.25 | f2knight | ISDN is 1 channel on that t1 |
| 00:43.52 | f2knight | most of the intergrated t1 services are HDSL circuits |
| 00:44.12 | f2knight | and most all int. t1 providers circuits are NOT on the first response list for repairs. |
| 00:44.36 | LiENUS | carrar, afaik our t1 is 4 pairs, and its some dynamic integrated t1 where the channels are either used for voice or used for data, as lines are used it drops available channels for data to use for voice |
| 00:44.40 | f2knight | if you get your T1 as a REAL t1 pure voice from the bell not a clec you will have top tier service |
| 00:44.52 | carrar | LiENUS, A T1 is always 2 pairs |
| 00:44.56 | carrar | 4 wires |
| 00:45.01 | LiENUS | err meant two pairs 4 wires |
| 00:45.11 | carrar | 2 = TX and 2 = RX |
| 00:45.59 | LiENUS | f2knight, well part of the issue is the clec refuses to put in a work order to have it fixed |
| 00:46.13 | carrar | LiENUS, I would get them to conver that T1 to a full data only T1 |
| 00:46.19 | f2knight | so your not using at&t, |
| 00:46.20 | carrar | then do SIP over it as well as yoru internet |
| 00:46.25 | carrar | with QoS |
| 00:47.04 | carrar | Assuming your carrier will do that for you |
| 00:47.09 | f2knight | LiENUS, whos the clec? |
| 00:47.15 | carrar | and if they wan't, find a different carrier |
| 00:47.15 | *** part/#asterisk batfastad (~benbradle@87-194-37-131.bethere.co.uk) |
| 00:47.18 | LiENUS | earthlink i think nowadays |
| 00:47.23 | f2knight | LOL |
| 00:47.30 | f2knight | Sorry , okay get another provider |
| 00:47.32 | LiENUS | previously was deltacom before them |
| 00:47.50 | f2knight | Stay away from Nuvox and Cebyond bewarned. |
| 00:47.51 | carrar | LiENUS, where are you at? |
| 00:47.56 | LiENUS | louisiana |
| 00:47.57 | carrar | egypt? |
| 00:48.06 | carrar | BFE louisiana? |
| 00:48.09 | carrar | or near a city? |
| 00:48.10 | LiENUS | f2knight, i dont think that'll be an issue since afaik they dont have any sort of presense in louisiana |
| 00:48.12 | LiENUS | in baton rouge |
| 00:48.20 | carrar | So big city |
| 00:48.25 | carrar | you should have choices |
| 00:48.43 | LiENUS | newp not really |
| 00:49.00 | f2knight | okay carrar hes all yours. |
| 00:49.05 | LiENUS | as i said they were the cheapest at 535 total iirc a month for t1 + 8 voice lines |
| 00:49.17 | carrar | LiENUS, worse case, get two internet circuits, one for data and one for vocie if you can't get your carrier to do QoS |
| 00:49.26 | carrar | and go SIP |
| 00:49.27 | LiENUS | beyond that its cable and voip, which comes close to the same price for less bandwidth or dsl and pots |
| 00:49.44 | f2knight | LiENUS, sorry but people have given some sound advice and you seem to not want to listen so I will let you go on your way. |
| 00:49.53 | carrar | or stay with what you have |
| 00:50.05 | Micc | LiENUS, two cable connections would be cheaper than a T1. |
| 00:50.07 | LiENUS | f2knight, well the sound advice was to get a diff carrier, its att or earthlink more or less here |
| 00:50.27 | LiENUS | Micc, you cant buy business cable internet without their voip offering |
| 00:50.33 | LiENUS | and their voip offering is like 60$ per line |
| 00:50.34 | f2knight | he could always just buy a module and reporting software for the PBX he has. but hey telephony 101 was never a class offered |
| 00:50.51 | LiENUS | f2knight, we have a maintainance contract with someone else |
| 00:51.06 | Micc | LiENUS, I don't think thats even legal. |
| 00:51.07 | LiENUS | if i do that we'll have thrown away the money we pay them to do it |
| 00:51.38 | f2knight | then you tell the maint. people that you need a way of pulling CDR details from the system |
| 00:51.42 | LiENUS | Micc, the cable company is presently suing another telco who wants to put fiber up claiming putting fiber up is illegal |
| 00:51.45 | f2knight | and have them impliment it and be done |
| 00:51.59 | LiENUS | i dont think our cable company here cares much about actual law |
| 00:52.13 | LiENUS | f2knight, thats not my position to do it, dont have the authority to |
| 00:52.14 | carrar | living in louisiana is illegal |
| 00:52.22 | carrar | You should move |
| 00:52.32 | LiENUS | all i get is told to find some way to get the data |
| 00:52.37 | f2knight | carrar, I hear the moon has a good view |
| 00:52.53 | LiENUS | lemme see if i can find ye old cable companies bid |
| 00:52.55 | f2knight | LiENUS, did you ever think to call the maint. company and ask? |
| 00:53.10 | LiENUS | f2knight, i dont have the authority to do that, they wont talk to me on the phone |
| 00:53.21 | carrar | why |
| 00:53.56 | f2knight | then you don't have the authority to make the changes you were told to make. |
| 00:54.10 | LiENUS | f2knight, that was my response |
| 00:54.18 | f2knight | you interface ANYTHING with your PBX then you will likely be told by your maint company that you voided it |
| 00:54.36 | f2knight | sounds like you need to resolve some internal issues with your company first. |
| 00:54.38 | carrar | I'm not sure you have authority to be here |
| 00:54.52 | f2knight | carrar, is right |
| 00:55.02 | f2knight | whats your telephony ID number? |
| 00:55.09 | LiENUS | you know, the funny thing is if i got dsl i would actually get a better sla than the t1 |
| 00:55.09 | carrar | This channel is for CAN DOERS |
| 00:55.24 | LiENUS | carrar, where you work, everyone has the ability to make any purchase for the company they want? |
| 00:55.36 | LiENUS | given that i would have to pay them to come in to install said equipment |
| 00:55.37 | *** join/#asterisk coppice (~chatzilla@m121-202-97-222.smartone-vodafone.com) |
| 00:55.37 | carrar | I can |
| 00:55.43 | carrar | I have a limit |
| 00:55.48 | carrar | but I can make purchases |
| 00:56.00 | f2knight | same here. |
| 00:56.09 | f2knight | If I am told to make something work and i know its under contract |
| 00:56.14 | carrar | I also can approve projects that cost more |
| 00:56.22 | carrar | if said project was aproved |
| 00:56.24 | f2knight | and if i make any changes it will void it I get permission to make it happen |
| 00:56.40 | LiENUS | i have to get approval for all purchases |
| 00:56.41 | f2knight | in most cases the boss adds me to the authorization list to make all the changes |
| 00:57.11 | f2knight | well carrar maybe we could be nice and spell it out for him. |
| 00:57.24 | carrar | LiENUS, Sounds you would be better off automating getting reports out of your old systme |
| 00:57.27 | f2knight | 1) call the main company and get a quote on the cost to put in a cdr device. |
| 00:57.30 | carrar | via RS232 I'm guessing? |
| 00:57.34 | f2knight | 2) present it to your boss |
| 00:57.41 | f2knight | 3) if denyed do nothing |
| 00:57.47 | f2knight | 4) find a back bone |
| 00:57.52 | f2knight | 5) get a new job |
| 00:58.03 | LiENUS | f2knight, or 1) finish law school 2) get new job |
| 00:58.09 | carrar | LiENUS, how do you interface your old PBX? |
| 00:58.16 | f2knight | how old are you LiENUS |
| 00:58.22 | LiENUS | carrar, i dont do anything with the old pbx |
| 00:58.23 | LiENUS | f2knight, 26 |
| 00:58.27 | f2knight | are you interning. for a cheap ass lawyer |
| 00:58.29 | carrar | LiENUS, you said you do reports? |
| 00:58.56 | f2knight | what the model and make of the PBX? |
| 00:58.57 | LiENUS | on the pbx? no i dont touch the phone system |
| 00:59.17 | f2knight | carrar, hes going to make a wonderful lawyer. |
| 00:59.29 | LiENUS | toshiba strata ctx 100? |
| 00:59.41 | LiENUS | f2knight, because i stay within the defined duties of my job? |
| 01:00.05 | carrar | "Room to Grow by Simply Adding to Existing Equipment" |
| 01:00.13 | carrar | Sounds like a great system!~ |
| 01:01.03 | carrar | that pbx supports remote system access |
| 01:01.15 | carrar | You should be able to do some sorts of reports |
| 01:01.23 | pdtpatrick | Question .. what software have you guys used to resample a .wav file? im trying to use sox and resample a 44100 Hz file but after conversion .. the file does not play |
| 01:01.42 | LiENUS | f2knight, you must not have much experience with attorneys or they behave different under civil law vs common law |
| 01:02.20 | carrar | I use sox |
| 01:02.49 | carrar | sox + google |
| 01:03.05 | *** join/#asterisk locojay (~locojay@pool-108-6-12-127.nycmny.fios.verizon.net) |
| 01:03.51 | locojay | hi i just added res_jabber and gtalk modules. unfortunatly i get "TLS connection cannot be established. Please install OpenSSL and its development libraries on this system" in asterisk -rvvvv |
| 01:03.54 | f2knight | LiENUS, Took me 4 min. to locate and find software that you run on your computer to get the information from your strata system |
| 01:04.00 | locojay | openssl + dev is installed |
| 01:04.29 | locojay | do i have to recompile after installing openssl-dev? |
| 01:04.37 | f2knight | on top of that the software is free and is built just for the system. |
| 01:04.50 | carrar | loco, I would think so |
| 01:05.01 | locojay | k ill give it a try |
| 01:05.08 | LiENUS | f2knight, and you proprose i accomplish this how without doing something to interfere with the company in charge of maintainance? |
| 01:05.17 | f2knight | pdtpatrick, make sure you convert it to mono |
| 01:05.35 | carrar | LiENUS, tell your company to assign this project to you |
| 01:05.42 | carrar | so you can do what is necessary to completely |
| 01:05.50 | carrar | complete it |
| 01:05.55 | f2knight | again you will make a wonderful lawyer LiENUS, you don't listen. |
| 01:05.58 | LiENUS | carrar, id rather ditch the system entirely if im going to do that |
| 01:06.07 | carrar | then get a budget |
| 01:06.11 | carrar | and do that |
| 01:06.16 | carrar | as that is the btter choice |
| 01:06.29 | carrar | but get approval |
| 01:06.33 | f2knight | okay I have never done this before .. but LiENUS I am sorry you have earned yourself an ignore status. |
| 01:06.34 | carrar | so you can make these changes |
| 01:06.43 | LiENUS | f2knight, you ever been in one of those places that have a sign that says "you break it you bought it?" |
| 01:07.24 | *** join/#asterisk gxdssoft (~gxdssoft@200.106.107.214) |
| 01:07.26 | carrar | no one is askign that you break stuff |
| 01:07.32 | carrar | You do need skills |
| 01:07.48 | carrar | 5ki11Z y0 |
| 01:07.55 | LiENUS | carrar, and its a live system in use for business critical functions, that i've never used before and dont have a spare to explore first |
| 01:08.10 | carrar | then pass it off to someone who ahs time |
| 01:08.17 | f2knight | locojay, yes you need to recompile. AN easier way is to run the script 'contrib/scripts/install_prereq' |
| 01:08.27 | f2knight | locojay, it will get everything you need |
| 01:08.28 | LiENUS | i'm not a big fan of running in and messing with something i know nothing about |
| 01:08.41 | carrar | Then you won't like asterisk |
| 01:08.42 | locojay | cool thanks |
| 01:08.52 | f2knight | locojay, then redo the ./configure ; make menuselect; make ; make install |
| 01:09.47 | locojay | worked thanks |
| 01:09.48 | LiENUS | carrar, asterisk i can easily put in a virtual machine ahead of time to test configuration changes |
| 01:10.04 | f2knight | NP locojay |
| 01:10.06 | locojay | i now have another prob : JABBER: encryption failure. possible bad password. but pwd is ok . |
| 01:10.24 | f2knight | Maybe that would be a hint to LiENUS to start listening to the advice people give. |
| 01:10.49 | LiENUS | for that matter if i wanted to right now i could set up a voip only asterisk setup with no additional hardware at all |
| 01:11.06 | f2knight | locojay, you using google voice? |
| 01:11.15 | locojay | yeah just testing |
| 01:11.26 | LiENUS | f2knight, well, it goes something like this. if i touch configuration on the toshiba pbx our contract is void and they get a free ticket to blame all issues on me |
| 01:11.55 | f2knight | locojay, okay well I will give some advice as I use it my self.. for starters I can never get it to answer the calls reliably it just ignores about 50 % of the time. so be prepared |
| 01:12.00 | raden | Is there a way to program all my polycom phones at one or one phone at a time about the only option ? |
| 01:12.02 | f2knight | locojay, outbound works wonderful though |
| 01:12.35 | f2knight | locojay, you need to have your gtalk and jabber file set exactly |
| 01:12.59 | locojay | k i just used this manual: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/11/adding_google_voice_to_freepbx.html |
| 01:13.00 | f2knight | locojay, https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 01:13.01 | LiENUS | now in theory we could sue them and prove that i didnt actually change anything. but that would cost quite and put us looking for a new phone maintainance company |
| 01:13.06 | locojay | thanks |
| 01:13.17 | locojay | 50% does not sound so good |
| 01:13.41 | locojay | mayber better to forward to a did. unfortunately i totally missed to get a sipgate us |
| 01:14.00 | f2knight | locojay, if you are lucky.. it is a known issue. it is because google is changing the API all the time. they might even be doing voice detection who knows. |
| 01:14.50 | locojay | big brother :) |
| 01:14.57 | f2knight | raden, Polycom/cisco phones are pretty much programed by setting up a tftp server and creating a config file for each devices mac address they then configure on boot |
| 01:15.18 | raden | f2knight, Good to know ! |
| 01:15.29 | f2knight | locojay, no i dont think that I think they are detecting voice to see that it is not being used as a trunk service to asterisk or other phone systems :) |
| 01:15.40 | LiENUS | sound advice generally does not include breaking contracts round these parts |
| 01:15.45 | raden | f2knight, so a tftp server a lil magic on my L3 switch for DHCP tftp boot and we should be good |
| 01:15.59 | f2knight | raden, now the fun part... there is really nothing i know of to make said config files except you makeing a custom script. |
| 01:16.30 | f2knight | raden, thats about the sum of it. thats how you do firmware updates and everything. |
| 01:16.32 | raden | f2knight, I know where to get sample scripts : ) |
| 01:16.45 | f2knight | raden, pass them along if you will :) |
| 01:17.12 | raden | f2knight, you have a email addy ? I have the locations stored on my work computer but ill be happy to send em to you I have like 10 example configs |
| 01:20.15 | f2knight | raden, just sent you a private message and thanks :) |
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| 01:48.30 | ghostmediapro1 | Remote Sip Ext 501 connect but no audio http://pastebin.com/9utydqEJ and Remote Sip Ext 503 Connects, recieves audio but after a few secounds drops the call/terminate/hangup http://pastebin.com/Yj8At0qe Please Assist. = complete call and sip set debug http://pastebin.com/VMN4R4yP = externhost=pbx.ghostmediapro.com localnet=192.168.0.0/255.255.255.0 nat=yes = ping pbx.ghostmediapro.com from terminal resolv http://pastebin.com/DVPMqjsE = ifconfig fr |
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| 01:55.26 | ghostmediapro1 | http://pastebin.com/tKkKf8Sg |
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| 02:20.11 | f2knight | ghostmediapro1, you are natted... do you have ports opened in your firewall router? |
| 02:21.50 | ghostmediapro1 | f2knight yes http://pastebin.com/RSW23HG2 router screen shots at the end of file |
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| 02:57.47 | lister_ | is there any way for asterisk to visually represent a call flow? |
| 02:58.14 | lister_ | i'm writing some software for call centers to track their transfers and hold times etc by extracting the info from a SOAP packet, but it seems fairly complicated and was hoping something already exists... |
| 02:58.52 | lister_ | not sure what to google though cos most 'call flow' type articles are about IVR etc |
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| 03:18.57 | WIMPy | You want to listen on AMI. |
| 03:19.26 | WIMPy | It will send events when call states change or the dialplan is traversed. |
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| 04:58.53 | f2knight | lister_, that is because call flow is the routing of the calls, or the 'flow' if your talking about visually representing the call flow then there are programs already. visualdialplan for example... If on the other hand you are trying to track the 'transfers' and 'hold' times then you have to know how the calls are being processed. Are they using Call Queues? If so QueueMatrics and OrderlyQueues already exists, and yes they are very c |
| 04:58.53 | f2knight | omplex applications and require the system setup to use 'there' software. If on the other hand you are using normal diallogic then you are going to have to look at the log files, and tuck in to the AMI interface and watch events and and do your stuff that way.. but I am not up on AMI, from what I have seen there is not a lot out other then interface stuff for you to write to it.. |
| 05:01.49 | nix8n82 | I need to build an asterisk cluster, and I'm not sure what tools I need. I need to be able to send ami to originate calls and have it distributed among the servers in a load balanced and failover way. I also need incoming calls to be load balanced and be able to failover.It would also be nice to have remote sip user be able to register to it as if it is a single server. What open source tools do I need to do this and/or is there any commercial products that |
| 05:01.49 | nix8n82 | <PROTECTED> |
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| 05:03.12 | lister_ | f2knight, thanks for the reply, looking at orderlyq now. we don't actually even use asterisk, but i'm thinking it would be easier to implement this than write call center software myself. |
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| 05:05.08 | [TK]D-Fender | nix8n82: SER <- |
| 05:06.52 | luckman212 | Polycom UCS v4.0.0 is out |
| 05:10.24 | nix8n82 | lister_ vicidial might do that as it is a call center software program. However it does not handle any newer version of asterisk than 1.4 |
| 05:10.53 | f2knight | nix8n82, Kamailio (OpenSER) , Radius, MySQL and if you need voicemail / IVR Asterisk to handle just the media, if the devices might be natted you will need RTPProxy/MediaProxy as well as a possible STUN server. HB for monitoring server status and oh yea probably some other scripts and utills like tftp, nagios/cacti or even puppet. |
| 05:11.01 | nix8n82 | Thanks [TK]D-Fender |
| 05:12.16 | f2knight | nix8n82, then you have to put all the pieces together. kamailio is where your devices register to.. it stores its information in a mysql db, then you have to learn alot about sip headers and how they work bevcause you have to build the entire processing stack |
| 05:12.47 | f2knight | nix8n82, piece of cake really. Should have it setup your first time in oh about 6 weeks.. |
| 05:13.05 | nix8n82 | sweet is that all |
| 05:13.14 | f2knight | nix8n82, give or take a bout 6-8 months for all the other stuff |
| 05:13.31 | f2knight | nix8n82, give or take a bout 6-8 months for all the other stuff |
| 05:13.51 | luckman212 | does anyone in here have access to the polycom resource center? |
| 05:13.59 | nix8n82 | well that really screws the two weaks they want me to do it in |
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| 05:14.26 | f2knight | nix8n82, not likely going to happen then. |
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| 05:14.48 | f2knight | nix8n82, bigger quesiton . do you want FAILOVER or REDUNDANCY |
| 05:15.04 | f2knight | nix8n82, that is loadbalance |
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| 05:15.31 | f2knight | nix8n82, FAILOVER simply means when one system goes down the secondone gets used. |
| 05:15.59 | f2knight | nix8n82, loadbalancing means that there is smarts to handling which system gets to process a call. |
| 05:16.12 | nix8n82 | I need loadbalancing |
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| 05:17.04 | f2knight | nix8n82, unless you have a tone of Networking and clustering experiense its not going to happen in your time frame. You can call in the consultants. We are always for Hire :) |
| 05:17.36 | f2knight | nix8n82, failover on the other hand,... you could pull off by the end of the week. |
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| 05:19.01 | nix8n82 | So the big question is what do you charge to setup a cluster? |
| 05:19.58 | nix8n82 | loadbalancing kind of implies failover as well? |
| 05:21.16 | carrar | $1,000,000 |
| 05:22.29 | carrar | nix8n82, Wouldn't you rather learn the software youself and set tha tup? |
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| 05:23.52 | nix8n82 | carrar I would but my problem is I have maybe a month and a half to get it done |
| 05:24.57 | carrar | Why not just go purchase a shoretel pbx? |
| 05:26.31 | carrar | or hire someone to setup Asterisk with a failover box |
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| 05:31.43 | nix8n82 | I need to be able to handle 5000 calls at once or more |
| 05:40.31 | carrar | handle it to where? |
| 05:41.11 | carrar | calls coming in via SIP or PRI's? |
| 05:42.29 | WIMPy | 5000 on PRIs? I think that's not the right price class. |
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| 05:43.48 | carrar | not a issue as long as you build it right |
| 05:44.10 | carrar | and or you may need to use a SIP router in front if it's all SIP based |
| 05:44.29 | carrar | But you lack any usfull details |
| 05:44.40 | carrar | as typical for questions in here :) |
| 05:45.37 | WIMPy | Well, I'd like to see you doing 5000 calls on PRIs. |
| 05:45.54 | carrar | not a issue, lots of people do |
| 05:45.57 | carrar | Bring those in via DS3's |
| 05:46.12 | carrar | mux em out |
| 05:46.24 | WIMPy | Still a bit tiny, isn't it? |
| 05:46.32 | carrar | heh |
| 05:47.02 | carrar | Thats what, 7 DS3's of PRI's? |
| 05:47.21 | carrar | 8 |
| 05:47.43 | WIMPy | How do you get to 8? |
| 05:47.53 | carrar | PRI = 23 channels |
| 05:47.59 | carrar | 28 PRI's per DS3 |
| 05:48.16 | carrar | Though that many, you'd be using NFAS |
| 05:48.25 | carrar | you could gain some channels back |
| 05:48.26 | WIMPy | Oh, yes, the US version. |
| 05:48.51 | carrar | nix8n82, Whats the purpose |
| 05:48.55 | WIMPy | I would end up at 11 E3, which doesn't seem to make much sense. |
| 05:48.55 | carrar | still haven't answeed |
| 05:49.22 | carrar | yeah, I would aggregate these through opensips |
| 05:50.50 | carrar | get some Crisco AS5400, convert the DS3's to SIP |
| 05:51.56 | carrar | Let the opensip route calls to least use asterisk servers or use opensips to actually register the phones as well |
| 05:52.12 | carrar | Always easy without any sort of details :) |
| 05:53.00 | carrar | meanwhile |
| 05:53.36 | nix8n82 | carrar to call people..on behalf of other people and connect them. |
| 05:54.37 | carrar | How does that work |
| 05:54.46 | carrar | make to calls and bridge them together? |
| 05:54.49 | carrar | two |
| 05:55.25 | carrar | Do you need to play any audio before conencting the calls? |
| 05:55.38 | carrar | if not, you don't even need asterisk |
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| 05:57.19 | nix8n82 | yeah some cases I need to play audio before connecting the call. |
| 05:57.56 | carrar | Well for those call you could pass the call out to a asterisk server that was was availble |
| 05:58.26 | carrar | 'bank of asterisk boxes |
| 05:58.29 | carrar | ' |
| 05:59.08 | carrar | keep track of how many calls each asterisk has so you don't overload them |
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| 06:02.53 | carrar | anyways, you are about as responsive as a door knob |
| 06:03.05 | carrar | it's very doable |
| 06:03.26 | __derek__ | Hi all, can anyone tell me if voice traffic goes through the asterisk PBX or not(when using SIP)? |
| 06:03.56 | WIMPy | __derek__: tcpdump |
| 06:04.04 | kaldemar | __derek__: depends on your configuration and the network setup. |
| 06:04.14 | __derek__ | hmm ok |
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| 06:05.00 | __derek__ | let me try tcpdump |
| 06:05.05 | f2knight | __derek__, in most cases voice does go through asterisk |
| 06:05.27 | f2knight | esp if you have canreinvite=no set |
| 06:05.46 | __derek__ | ok, I though that asterisk just connected the 2 extensions... |
| 06:06.17 | f2knight | sure and if it does call recording it is in the media path, |
| 06:06.26 | __derek__ | ok |
| 06:06.26 | f2knight | if it has a zap card its in the media path |
| 06:06.43 | __derek__ | yeh |
| 06:06.57 | f2knight | if it does ANYTHING with media... liek playing a message recording audio.. voicemail. its in the media path |
| 06:07.50 | __derek__ | ok, like conferences |
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| 06:09.37 | nix8n82 | sorry carrar thanks for your input...I'm not good at multi tasking |
| 06:11.33 | __derek__ | voice data is sent using higher numbered ports via rtp, right? |
| 06:12.58 | carrar | typically |
| 06:16.52 | f2knight | __derek__, 10000-20000 usually |
| 06:16.52 | __derek__ | thanks guys for all your help... I think for my setup voice data doesn't go through the PBX as bandwidth usage by the PBX is just 5.8kbits/sec |
| 06:17.24 | __derek__ | ok, but is that dependent on Asterisk or the SIP client? |
| 06:17.40 | carrar | yes |
| 06:18.27 | carrar | rtp.conf |
| 06:18.35 | __derek__ | ok |
| 06:18.57 | __derek__ | yup it is 10000 to 20000 |
| 06:19.33 | __derek__ | Thanks f2knight , carrar |
| 06:19.37 | carrar | (if the audio is coming asterisk) |
| 06:21.42 | __derek__ | Yeh, I connected to a conference and asterisk opened up udp 12658 |
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| 06:23.40 | __derek__ | OTOH I started a voice call and asterisk opened 2 udp ports in that range |
| 06:30.02 | f2knight | __derek__, asterisk often has to create 2 legs for a single call |
| 06:31.49 | __derek__ | I figured that one port per sip client |
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| 06:32.33 | MDesade | hello hello |
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| 06:33.55 | MDesade | i am looking to buy 2x TDM410's with echo cancellation modules cards, to replace my 2x TDM400's.... does anyone know if i can populate the new cards, with the modules from the old TDM400s or? did they change EVERYTHING making it all incompatible??? |
| 06:34.53 | *** join/#asterisk astrutt (astrutt@brain.ratman.org) |
| 06:37.28 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:38.02 | schmidts | good morning |
| 06:41.03 | ChannelZ | MDesade: they're compatible so far as I know |
| 06:43.03 | ChannelZ | __derek__: incoming will use a port in the range of your configured rtp.conf but outgoing will use whatever the remote side requested. |
| 06:43.31 | MDesade | excellent... that is good to know |
| 06:44.01 | MDesade | obviously, i should call digium tomorrow and find out for sure |
| 06:44.34 | ChannelZ | MDesade: http://blogs.digium.com/2008/02/04/digium-the-tdm410-and-the-genie/ -- but yes it wouldn't hurt to give a quick call |
| 06:46.59 | MDesade | excellent... (despite quoting disney's aladdin) |
| 06:49.20 | __derek__ | ChannelZ: why are the UDP(rtp) ports necessary? Isn't signalling done on the usual sip ports? |
| 06:50.18 | carrar | rtp is audio, signaling is sip/sdp |
| 06:50.40 | carrar | though sdp can be in rtp |
| 06:51.01 | __derek__ | hmm ok |
| 06:51.22 | carrar | but then it's not really caled rtp, it's call sdp |
| 06:51.22 | __derek__ | I was trying to understand wrt http://en.wikipedia.org/wiki/Session_Initiation_Protocol#SIP_messages (the diagram) |
| 06:51.41 | carrar | ~book |
| 06:51.42 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 06:51.46 | ChannelZ | SIP just handles call setup/teardown and other control type stuff. |
| 06:51.53 | ChannelZ | The actual audio streams use a different port. |
| 06:52.34 | ChannelZ | see "MEDIA (RTP) PATH" on that diagram |
| 06:53.55 | __derek__ | ChannelZ: I saw the "MEDIA (RTP) PATH" and that made me wonder why 2 UDP ports in the 10000 - 20000 range were opened on asterisk when I made a new voice call |
| 06:54.12 | __derek__ | from what I understand, the ports need to be opened only on the clients? |
| 06:54.13 | ChannelZ | well you should have one going out and one coming in |
| 06:54.30 | ChannelZ | (plus the SIP port) |
| 06:54.44 | kaldemar | __derek__: your asterisk stays on the media path, that's why. |
| 06:55.19 | kaldemar | __derek__: that example on wikipedia doesn't apply as is anyway because asterisk is not a proxy, but a B2BUA. |
| 06:55.34 | __derek__ | kaldemar: ahh that explains it |
| 06:55.38 | ChannelZ | UDP is uni-directional. So you're sending audio out of one port but receiving audio in on another. And yes if you're doing so through asterisk, you might very well have 2 sets of these - one set from the device (your phone) to Asterisk, and the other to the remote end. |
| 06:59.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:01.17 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:09.49 | f2knight | __derek__, you can do a show channels on the console too to see what channels are setup |
| 07:10.13 | f2knight | well 'core show channels' |
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| 08:28.00 | *** join/#asterisk porche (~kursad@78.184.202.199) |
| 08:28.35 | porche | Hi, Does anyone know the ways to use asterisk as ivr server for alcatel pbx? |
| 08:30.30 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 08:35.43 | *** part/#asterisk MDesade (~chatzilla@ip70-162-84-98.ph.ph.cox.net) |
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| 08:47.24 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 08:47.29 | devil_evoxxx | hi all guys :) |
| 08:49.15 | irroot | hi there how goes |
| 08:49.24 | *** join/#asterisk ChrisInSydney (~Chris@CPE-58-173-104-6.nwqt1.ken.bigpond.net.au) |
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| 09:36.28 | *** join/#asterisk ironm (~ironm@fwj00.e-fon.ch) |
| 09:40.08 | ironm | hello. Please allow me one question according to snmp module of asterisk. Even the snmp module is loaded and answers to snmpget (s. directly below) I am getting an errror when trying to use snmpwalk (s. further below) |
| 09:40.12 | ironm | [root@voipswitch02 asterisk]# snmpget -m ALL -v1 -c EquiniX localhost sysUpTime.0 |
| 09:40.21 | ironm | DISMAN-EVENT-MIB::sysUpTimeInstance = Timeticks: (408776) 1:08:07.76 |
| 09:40.23 | ironm | # --- |
| 09:40.37 | ironm | [root@voipswitch02 asterisk]# snmpwalk -v 3 -u asteriskUser -n "" -l authPriv -a SHA -A password-bla -x AES -X password-bla localhost asterisk |
| 09:40.50 | ironm | ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID |
| 09:40.52 | ironm | # --- |
| 09:41.08 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 09:41.26 | ironm | I use asterisk 1.6.1.20 and have this behavior on 32 and 64 Bit systems |
| 09:41.49 | ironm | Do you have any idea what is wrong? Thank you in advance for any hints |
| 09:43.15 | ironm | all mib files are in there: |
| 09:43.17 | ironm | -rw-r--r-- 1 root root 521 Oct 11 10:16 /usr/share/snmp/mibs/digium-mib.txt |
| 09:43.34 | ironm | -rw-r--r-- 1 root root 17319 Oct 11 10:16 /usr/share/snmp/mibs/asterisk-mib.txt |
| 09:51.53 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:57.23 | coppice | irroot: do you know why the current FAX support in * doesn't supposrt FAX polling? the old rxfax and txfax did |
| 09:58.58 | irroot | coppice no clue sorry by old you talking about the app_fax from soft-switch.org site 1.2/1.4 ?? |
| 10:00.52 | coppice | app_fax was a later revision. the stuff from soft-switch.org was app_rxfax and app_txfax. they all supported polling. Now it seems a compatible parameter is detected just so * can report that polling is not supported |
| 10:01.01 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 10:01.49 | coppice | app_rxfax and app_txfax were originally just test programs to exercise the FAX engine. for a long time they became the normal way people accessed the FAX engine |
| 10:06.19 | *** join/#asterisk raj-darkmystery (~SR@114.143.184.114) |
| 10:06.28 | raj-darkmystery | guys need some guidance |
| 10:06.41 | raj-darkmystery | i am new in voip or you can say asterisk |
| 10:09.38 | raj-darkmystery | i wanted to know how i can transfer a call to another extension if the extension is not available,,, was scrawling through google and found many posts but are confusing as well... can any one guide me ? |
| 10:12.22 | irroot | coppice perhaps has something to do with supporting the digium fax stack |
| 10:13.04 | *** join/#asterisk iulhk (~carryoff@116.71.183.7) |
| 10:15.08 | raj-darkmystery | i wanted to know how i can transfer a call to another extension if the extension is not available,,, was scrawling through google and found many posts but are confusing as well... can any one guide me ? |
| 10:15.10 | raj-darkmystery | asterisk and voip is new for me actually :-/ |
| 10:17.27 | *** join/#asterisk raj-darkmystery (~SR@114.143.184.114) |
| 10:17.45 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:19.01 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 10:19.46 | *** part/#asterisk porche (~kursad@78.184.202.199) |
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| 10:21.35 | coppice | irroot: that would make it a rather sadly limited stack |
| 10:23.09 | coppice | irroot: I kinda discounted that possibility, as the digium stack is from commetrex, and should be very complete |
| 10:25.09 | irroot | i am not sure just guessing it should not be too hard to put it back ?? you see in trunk ive moved faxdetect into a framehook this could start using some of the spandsp DSP bits if available in res_fax_spandsp |
| 10:25.25 | irroot | same applies to the v.21 detector |
| 10:35.09 | *** join/#asterisk nicola_pav (~chatzilla@62.219.238.196) |
| 10:36.08 | nicola_pav | hello. I want to integrate asterisk+ freepbx with opensips. I have already 2 accounts registered to opensips and I am forwarding all media to asterisk. extensions can talk to asterisk when dialing for example *43. but extensions can call each other |
| 10:36.50 | nicola_pav | how can i let asterisk or freepbx search the opensips database (already added in res_mysql.conf and extconfig.conf) for the other extension and allow the call between them? |
| 10:36.53 | nicola_pav | any hints please? |
| 10:37.25 | nicola_pav | extensions cannot call each other* |
| 10:38.36 | puzzled | hi |
| 10:38.46 | kaldemar | nicola_pav: dial the opensips. |
| 10:39.15 | nicola_pav | kaldemar: I am afraid i did not get u |
| 10:39.47 | kaldemar | nicola_pav: disregard that. what's the point in forwarding media to asterisk? |
| 10:40.17 | nicola_pav | kaldemar: i want to separate registration from media |
| 10:40.39 | nicola_pav | i want asterisk to handle the media |
| 10:40.53 | kaldemar | asterisk is not an RTP proxy. |
| 10:41.13 | kaldemar | sounds like you try to use it as one. |
| 10:42.31 | nicola_pav | so integrating asterisk with opensips just to handle IVRs, announcements and vociemail? |
| 10:43.29 | kaldemar | http://www.opensips.org/Resources/DocsTutAsterisk |
| 10:43.38 | nicola_pav | i dont want asterisk to handle the registration |
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| 12:11.04 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:15.44 | Naikrovek | yawns. |
| 12:16.01 | irroot | WIMPy ping |
| 12:16.11 | Naikrovek | somehow knocks his whiteboard off the wall which his flying fists of yawning. |
| 12:30.29 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
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| 12:54.31 | n3hxs | did that wake you up Naikrovek? |
| 12:54.56 | Naikrovek | it woke me up more, yeah. |
| 12:57.11 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-juafqjvbxxppsohq) |
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| 13:05.24 | *** join/#asterisk atan (atan@unaffiliated/atan) |
| 13:14.40 | Katty | stretches |
| 13:15.19 | n3hxs | hands Katty a coffee. |
| 13:15.27 | tzanger | we need a kattycam for that kind of stretching action |
| 13:15.42 | n3hxs | the only cam shows the birds outside. |
| 13:15.56 | tzanger | yeah I know |
| 13:16.05 | Katty | there will be no kattycams |
| 13:16.11 | tzanger | haven't trained the birds to knock the cam 'round the other way yet |
| 13:16.13 | Katty | i would break the camera. |
| 13:17.57 | n3hxs | I don't think so. |
| 13:18.22 | tzanger | I've seen pix. that's a lie and you know it |
| 13:18.30 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 13:19.32 | *** join/#asterisk topro (~topro@host-62-245-142-50.customer.m-online.net) |
| 13:22.43 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 13:23.08 | topro | are there special settings which need tuning to make asterisk work reliably (i.e. more lag-tolerant) on a wifi-sip setup? wired sip phones work without issues. wifi-sip handsets tend to drop calls. Investigating whether its wifi related or asterisk settings which might improve situation. |
| 13:26.21 | Katty | oh god. |
| 13:26.28 | Katty | i'm being dragged along to a sales call |
| 13:26.32 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 13:26.35 | Katty | cries |
| 13:27.15 | SunTsu | rolls a ball of tissues through the channel |
| 13:33.24 | *** join/#asterisk Lantizia (~Lantizia@cpc22-stok16-2-0-cust96.1-4.cable.virginmedia.com) |
| 13:34.24 | Lantizia | how might I test if a string is in a string in asterisk (i'd much rather test if a number is in an array - but i see no proper arrays) |
| 13:34.29 | Lantizia | this is in dial plan btw |
| 13:35.28 | [TK]D-Fender | Lantizia, can you reword that... |
| 13:36.11 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:36.53 | Lantizia | ok, is there a function to test if a string is inside another string... or instead maybe how do I make a proper array to hold some numbers and check to see if a given number is in that array |
| 13:38.16 | [TK]D-Fender | Lantizia, Don't believe the is a FUNC for that. You could make your own macro for it though. And there is no "array" structure in * yet |
| 13:40.01 | Lantizia | yeah I know there isn't any arrays which is why I asked about checking for strings in strings e.g... IF($[INSTR(${EXTEN}\,$[LISTEXTEN]=true]?meow:woof) |
| 13:40.19 | Lantizia | where INSTR checks to see if say "200" is in a string of "200,201,202,203,204" |
| 13:41.01 | [TK]D-Fender | Lantizia, As I said I ddon't believe ther is a FUNC for that. |
| 13:44.03 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 13:44.13 | p3nguin | Does 10 provide array support? |
| 13:44.26 | ironm | hello. Please allow me one question according to snmp module of asterisk. Even the snmp module is loaded and answers to snmpget (s. directly below) I am getting an errror when trying to use snmpwalk (s. further below) |
| 13:44.33 | ironm | [root@voipswitch02 asterisk]# snmpget -m ALL -v1 -c EquiniX localhost sysUpTime.0 |
| 13:44.39 | *** join/#asterisk serafie (~erin@nat/digium/x-frxgmylhlbhfqmay) |
| 13:44.42 | ironm | DISMAN-EVENT-MIB::sysUpTimeInstance = Timeticks: (408776) 1:08:07.76 |
| 13:44.52 | ironm | # --- |
| 13:45.00 | ironm | [root@voipswitch02 asterisk]# snmpwalk -v 3 -u asteriskUser -n "" -l authPriv -a SHA -A password-bla -x AES -X password-bla localhost asterisk |
| 13:45.08 | p3nguin | ~pb |
| 13:45.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:45.09 | ironm | ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID |
| 13:45.33 | ironm | oh .. I see .. thank you p3nguin |
| 13:45.59 | ironm | <PROTECTED> |
| 13:46.23 | ironm | Do you have any idea what is wrong? Thank you in advance for any hints. All MIBs are in place |
| 13:51.19 | *** join/#asterisk jetlag (~jetlag@pool-71-168-246-8.cmdnnj.east.verizon.net) |
| 13:54.13 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-134-68-195.in-addr.btopenworld.com) |
| 13:54.35 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 13:54.43 | StaRetji | Howdy |
| 13:55.14 | StaRetji | folks, is it possible with asterisk pbx to leave voicemail message for mobile number user - directly? |
| 13:55.40 | StaRetji | I mean, leave voicemail without mobile users phone ringing? |
| 13:55.41 | [TK]D-Fender | StaRetji, * can't do anything more with the PSTN than you can with a normal phone |
| 13:56.08 | StaRetji | [TK]D-Fender: thx, I know it can be done with mobile phone |
| 13:56.19 | StaRetji | so, I wonder if it will work with * |
| 13:56.39 | n3hxs | If you do the same thing with * that you do with a normal phone. |
| 13:56.55 | WIMPy | It you can. |
| 13:57.12 | [TK]D-Fender | stAnd how do you do it with a mobile phone? Also * isnt' a mobile phone. It would matter as to how * is expected to actually communicate with the PSTN |
| 13:57.13 | StaRetji | I know that some operators will let you call 06xxxxxxx* |
| 13:57.33 | StaRetji | and will prompt voicemail without ringing users mobile |
| 13:57.38 | WIMPy | Asterisk can be a mobile phone. |
| 13:58.01 | kaldemar | StaRetji: is something stopping you from dialing 06xxxxxxx* on such operators? |
| 13:58.21 | kaldemar | StaRetji: it is up to the operator, not asterisk. |
| 13:58.27 | [TK]D-Fender | StaRetji, Then you need to know what those prefixes are and be calling them using a tech that will pass it on accordingly |
| 13:58.54 | StaRetji | kaldemar: nope, I found it on google, so I don't know any number at that specific operator, but, I will google for my mobile operator setting and try it |
| 13:59.36 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 13:59.41 | StaRetji | well, I tried my own with * at the end |
| 13:59.50 | [TK]D-Fender | StaRetji, There is no "standard". Any solution is going to be limited and possibly difficult to isolate where you can apply it |
| 13:59.51 | StaRetji | asterisk said, no dialplan rules matched |
| 14:00.21 | kaldemar | StaRetji: you of course must make an extension that matches. |
| 14:00.23 | StaRetji | actually, my grandstream gxv3000 said that |
| 14:00.45 | StaRetji | calling that number without * works |
| 14:01.04 | StaRetji | so I'm a bit confused what extension should be made |
| 14:01.17 | StaRetji | for example 061234567 will ring my phone |
| 14:01.38 | StaRetji | 061234567* will give that message, no dialplan rules matched |
| 14:03.03 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 14:04.03 | atan | I hear a Rogers customer |
| 14:04.04 | WIMPy | irroot: pong. BTW: I haven't seen anything on the ML, yet. |
| 14:04.06 | atan | Whoop |
| 14:04.41 | irroot | WIMPy its there now gmail is blocked just leting you know it be working ok |
| 14:05.05 | WIMPy | Ok, hope to see it soon then. |
| 14:05.31 | irroot | ive changed the incoming channel in addition to be <name>/<port>-<id> where i make it mISDN locally to not have to change dialplan |
| 14:06.24 | WIMPy | You check the channel name for the port? |
| 14:07.17 | WIMPy | But you did realize that you can put ports in different contexts? |
| 14:07.22 | irroot | WIMPy i do so all come into one context and let asterisk sort it out only needed when no msn/did |
| 14:07.39 | *** join/#asterisk master_of_master (~master_of@87.181.65.165) |
| 14:07.58 | irroot | yeah i know this but want LCR out of the picture |
| 14:08.05 | WIMPy | And you can also enure that won't happen by configuring MSNs for the port. |
| 14:08.47 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
| 14:10.10 | irroot | WIMPy if i have any issues ill buzz you please buzz me if you have questions |
| 14:10.36 | WIMPy | sure |
| 14:11.22 | WIMPy | BTW: I did a patch to make MSN configuration work with partial numbers, compared from the right, just like the PSTN does. |
| 14:14.01 | irroot | nice but we dont get any info on most our lines here unless we pay extra |
| 14:14.09 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 14:14.29 | WIMPy | That's for connecting phones. |
| 14:15.24 | irroot | WIMPy ah indeed we dont get ISDN phones locally only lines |
| 14:16.25 | WIMPy | You doun't get the called number on your lines, unless you pay extra? |
| 14:17.23 | irroot | WIMPy yeah its stupid but thats how it is |
| 14:17.40 | WIMPy | Evil teclos. |
| 14:19.09 | irroot | the worst |
| 14:22.39 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:f49c:1819:b29a:cf01) |
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| 14:25.54 | devil_evoxxx | hi irroot , how go on? |
| 14:26.12 | irroot | well thx |
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| 14:27.32 | devil_evoxxx | when you use pri-card , you alwais mount echo cancellation module ? (only for faxing) |
| 14:29.37 | irroot | devil_evoxxx explain mount ?? i use oslec module loaded not the hardware module faxing and echo canceleation dont work well |
| 14:30.15 | devil_evoxxx | ops :) Mount i intend the hw module |
| 14:31.09 | [TK]D-Fender | hardware module faxing <- huh? |
| 14:31.28 | devil_evoxxx | no :) the echo cancellation module |
| 14:31.33 | devil_evoxxx | hw |
| 14:32.25 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:32.29 | puzzled | WIMPy: I have a mISDN_dsp_oslec unknown symbol issue similar to the report here: http://permalink.gmane.org/gmane.linux.isdn.i4l.user/4235 Have you seen this or know a solution? |
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| 14:35.47 | WIMPy | puzzled: Looks like a version conflict to me. |
| 14:37.13 | WIMPy | Are you using parts of an older kernel together with parts of a newer standalone version? |
| 14:37.14 | devil_evoxxx | i've got two pri, i use one E1 for faxing out and the second E1 for inbound/outbound voice. So i've got to decide if buy the hw echo cancellation module or not. what do you recomend? |
| 14:37.48 | WIMPy | Buy half an EC module :-) |
| 14:37.53 | coppice | buy half a module |
| 14:37.59 | [TK]D-Fender | If you intend to resell or reuse it later, then get it, otherwise don't |
| 14:38.12 | [TK]D-Fender | because EC kills faxing <- |
| 14:38.26 | [TK]D-Fender | it would have to be disabled on all calls (normally automatic) |
| 14:38.26 | WIMPy | It can be a good thing for voice, but if you use one interface for fax only, you should disable it on that port. |
| 14:38.32 | puzzled | WIMPy: using a standalone build of the modules copied to /lib/modules/<kver>/extra/mISDN and using depmod overrides to force the kernel to not use the original kernel modules but instead use my new ones |
| 14:38.33 | devil_evoxxx | yes, but can i disable the hw module sw , right? |
| 14:38.52 | [TK]D-Fender | " hw module sw" <- huh? |
| 14:39.07 | devil_evoxxx | echo hw module |
| 14:39.14 | WIMPy | devil_evoxxx: yes |
| 14:39.40 | WIMPy | puzzled: Are you sure it doesn't lod any of the old modules? Maybe it would be a good idea to remove them. |
| 14:39.49 | devil_evoxxx | fine :) so is now time to buy with echo-canc hw module |
| 14:40.29 | puzzled | WIMPy: my idea too. working on that right now |
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| 14:46.47 | Madkiss | is there something else that allows me to connect two local calls apart from a meetme or confbridge-conference? |
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| 14:49.39 | [TK]D-Fender | Madkiss, "Bridge" |
| 14:51.43 | Madkiss | [TK]D-Fender: and can I combine this somehow with the G-argument of the Dial()-command? |
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| 14:52.56 | [TK]D-Fender | Madkiss, You could do a Redirect to a dialplan extension that dials another local channel using G. In that loacl channel you do the actual bridge. That might work. |
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| 15:05.02 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 15:05.05 | Madkiss | [TK]D-Fender: please forgive the dumb question earlier; what I was really looking for was not using G in Dial() but instead just let the normal call proceeding happen, and that is "If a Call is andered by Answer() and then a normal Dial() is executed, the two calls will be connected anyway" :) |
| 15:05.10 | Ionic | Hey |
| 15:05.32 | Ionic | How comes asterisk made a version change from 1.8 to 10.0.0? Do I have to think binary? |
| 15:05.51 | WIMPy | ~asterisk10 |
| 15:05.52 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 15:05.56 | WIMPy | Ionic: ^^ |
| 15:06.20 | WIMPy | We are better than Firefox. |
| 15:06.31 | [TK]D-Fender | Madkiss, that made even less sense. |
| 15:07.01 | Madkiss | no worries. it does what I expect it to do now. |
| 15:07.19 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 15:07.34 | Ionic | Ah I see |
| 15:07.45 | Ionic | Mh |
| 15:07.54 | Ionic | My package manager doesn't let me install asterisk, nice |
| 15:08.53 | pabelanger | Ionic: I doubt asterisk 10 is available in your package manager |
| 15:09.14 | Ionic | pabelanger: nah, 1.8.7.0 |
| 15:09.38 | Ionic | Oh... what the |
| 15:09.44 | [TK]D-Fender | I'd run from any repo that carried a pre-release branch... |
| 15:10.22 | Ionic | RDEPEND="... h323? ( net-libs/openh323 )" DEPEND="${RDEPEND} !net-libs/openh323" |
| 15:10.24 | Ionic | Uhm... |
| 15:10.45 | Ionic | Yeah, sure... lol |
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| 15:19.04 | Katty | cries. |
| 15:19.32 | Qwell | Katty: not allowed! we must fix you. |
| 15:20.23 | [TK]D-Fender | ... |
| 15:25.15 | WIMPy | I've got an interesting phenomenom when using sip/tls, which seems to be working again in todays version. |
| 15:26.22 | WIMPy | When a bridged call is hung up at the remote end, everything is fine, but if I call an extension that only does SayUnixTime() and Hangup(), the phone doesn't seem to get the call end. |
| 15:26.36 | WIMPy | Does that make any sense to anyone? |
| 15:26.55 | ChannelZ | did you answer() first? |
| 15:27.12 | WIMPy | yes |
| 15:28.27 | WIMPy | Oh, wait. I'm only using TLS in one direction, I think. |
| 15:30.56 | WIMPy | Oops. Asterisk to phone doesn't seem to work. |
| 15:31.14 | *** join/#asterisk celord (~celord@201.194.102.181) |
| 15:31.55 | Ionic | ~capi |
| 15:31.55 | infobot | i heard capi is Common ISDN Application Programming Interface. See http://www.capi.org for more info. |
| 15:32.03 | Ionic | Mh |
| 15:32.29 | *** join/#asterisk pietro (~pietro@93-62-151-43.ip23.fastwebnet.it) |
| 15:32.36 | Ionic | ~chan_capi |
| 15:32.36 | infobot | chan_capi is probably work by kapejod located at http://www.junghanns.net/asterisk/, or chan_cpi-cm from http://sourceforge.net/projects/chan-capi which is under active maintinance |
| 15:33.04 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 15:33.10 | puzzled | Ionic: that's ancient. use chan_capi from http://www.melware.org/ChanCapi |
| 15:33.11 | WIMPy | Err, no. Try cahn-capi.org. |
| 15:33.15 | Ionic | "active" :/ |
| 15:33.38 | puzzled | ftp://ftp.chan-capi.org/chan-capi |
| 15:33.43 | WIMPy | It does work with 1.8 finally. |
| 15:33.45 | Ionic | Yeah |
| 15:33.55 | Ionic | last edited 2010-04-06 |
| 15:33.56 | Ionic | Mh... |
| 15:33.59 | Ionic | "active" |
| 15:34.08 | puzzled | nice. maybe one day I'll have dahdi, sangoma, capi and misdn interfaces working |
| 15:34.24 | titter | So who wants to assist with getting the dynamic conference pin numbers to work again :) |
| 15:34.25 | WIMPy | All at once? |
| 15:34.31 | puzzled | Ionic: don't let the date fool you. it has a ton of advanced features |
| 15:34.38 | Ionic | Heh |
| 15:34.41 | Ionic | Well idk |
| 15:34.48 | Ionic | I've gotta install Asterisk first |
| 15:34.48 | puzzled | WIMPy: yes, when I figure out how to create the magic sauce :) |
| 15:34.50 | Ionic | Btw |
| 15:35.00 | Ionic | I don't need hylafax or something, right? |
| 15:35.05 | WIMPy | Yes, I'm pretty sure it isn't that long ago it started working with 1.8. |
| 15:35.08 | Ionic | I want to get rid of it anyway |
| 15:35.14 | puzzled | Ionic: off course, just because hylafax is cool |
| 15:35.20 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 15:35.26 | Ionic | puzzled: uh, no |
| 15:35.32 | Ionic | It's colloding with mgetty |
| 15:35.36 | puzzled | Ionic: but seriously you can use fax with chan_capi directly |
| 15:35.42 | Ionic | kk |
| 15:35.52 | Ionic | Well |
| 15:35.56 | WIMPy | puzzled: Sangoma didn't cross my way so far. But I've goit the others working on my test box. |
| 15:36.03 | Ionic | No idea how comfortable it will be but I'll find out |
| 15:36.23 | Ionic | I don't quite understand why I can't install both hylafax and mgetty at the same time |
| 15:36.24 | puzzled | Ionic: if you figure out how to fax from libreOffice directly via chan_capi let me know. I like that part about hylafax |
| 15:37.11 | Ionic | My basic problem is that hylafax doesn't work with mgetty :/ |
| 15:37.20 | Ionic | But I sort of need mgetty for my analog MODEM |
| 15:37.22 | WIMPy | Yes, fax is the "killer app" for capi, I guess. |
| 15:37.26 | Ionic | Or actually vgetty |
| 15:37.29 | WIMPy | Or maybe the only one. |
| 15:37.33 | puzzled | then use agetty |
| 15:37.47 | Ionic | ? |
| 15:38.00 | Ionic | You do know what mgetty/vgetty is, right? :p |
| 15:38.14 | puzzled | something annoying needed to make hylafax work |
| 15:38.15 | Ionic | WIMPy: oh, what about voice? %) |
| 15:38.40 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 15:38.42 | Ionic | puzzled: uh, no. hylafax won't work with mgetty and vice versa |
| 15:39.15 | WIMPy | The general trouble about capi is that it's somehow unpredictable. What you can or can't do with it depends on your vendors specific implementation. |
| 15:39.29 | puzzled | Ionic: unless I installed something that hylafax don't use, I do have mgetty on my hylafax box. and mingetty I see now |
| 15:39.39 | Ionic | puzzled: but vgetty really rocks, it's the only voice software for analog MODEMs I've so far encoutered |
| 15:39.58 | puzzled | uhm voice via analog modems? that's what sip is for :) |
| 15:40.01 | Ionic | puzzled: mingetty is something entirely different |
| 15:40.08 | Ionic | SIP sucks |
| 15:40.24 | puzzled | yes it does |
| 15:41.23 | Ionic | But I want to use voice and fax via this ISDN card over Asterisk now that it's working |
| 15:41.41 | Ionic | I tested basic functionality with capisuite and it worked decent enough |
| 15:41.51 | puzzled | slaps some sense into /dev/mISDNtimer with a udev rule |
| 15:41.51 | drift- | p3nguin he told me to speak to you , asterisk server is refusing me my ip and login and password does not work for my login drift |
| 15:42.19 | Ionic | I'll leave vgetty/mgetty attached to my analog modem just in case |
| 15:42.30 | Ionic | There's already VOCP running pretty fine |
| 15:42.39 | Ionic | Voice box and fax stuff, pretty nice |
| 15:42.46 | puzzled | Ionic: that's what my current asterisk box does with chan_capi and hylafax |
| 15:42.53 | puzzled | it's ancient though |
| 15:43.09 | Ionic | puzzled: heh... why hylafax, really? |
| 15:43.20 | Ionic | I don't see the reason for hylafax |
| 15:43.22 | Ionic | Maybe the tools |
| 15:43.34 | WIMPy | hylafax can connect directly to capi. |
| 15:43.43 | Ionic | So does Asterisk? |
| 15:43.48 | puzzled | because when I started using chan_capi eons ago it did not have fax so I installed hylafax & iaxmodem |
| 15:43.51 | WIMPy | That's probably still the best fax solution available. |
| 15:43.54 | Ionic | Well, "directly" is wrong though |
| 15:44.09 | Ionic | It does need some capi4hylafax patch |
| 15:44.19 | Ionic | puzzled: oh I see |
| 15:44.29 | puzzled | iirc that capi4hylafax patch is no longer maintained |
| 15:44.41 | WIMPy | Oh :-( |
| 15:44.56 | *** join/#asterisk navaismo (~navaismo@201.123.42.167) |
| 15:45.07 | Ionic | So CAPI won't work in newer hylafax versions anyway? |
| 15:45.29 | puzzled | don't think so. it's from 2007 |
| 15:45.32 | puzzled | ftp://ftp.chan-capi.org/capi4hylafax/ |
| 15:45.37 | Ionic | yeah |
| 15:46.55 | Ionic | Hm... wtf |
| 15:47.04 | Ionic | chan_capi releases actually lowered in size |
| 15:47.12 | *** part/#asterisk titter (~Justin@c-76-101-15-40.hsd1.fl.comcast.net) |
| 15:47.14 | puzzled | code optimization :) |
| 15:47.31 | Ionic | At some point only though |
| 15:47.43 | Ionic | Nah it can't be code |
| 15:47.52 | Ionic | 100KB of *compressed* code? |
| 15:48.00 | Ionic | nowai |
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| 16:25.33 | Ionic | Mh wow |
| 16:25.39 | Ionic | chan_capi is throwling a lot of errors |
| 16:26.28 | Ionic | s/throwl/throw/ |
| 16:27.24 | Ionic | Do I need svn for Asterisk 1.8? |
| 16:28.15 | *** join/#asterisk cusco (~tralala@a79-168-174-232.cpe.netcabo.pt) |
| 16:28.23 | cusco | Hi |
| 16:28.39 | cusco | I know this doesn't make sense, but I still dare to ask |
| 16:29.16 | cusco | placing calls in a queue... would there be a way to state that a call has higher priority to a certain queuemember ? |
| 16:29.22 | Naikrovek | throwling is better. That's what my daughter does at the bowling alley. |
| 16:29.24 | cusco | so it tries to ring him first? |
| 16:29.50 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:31.28 | eduzimrs | hey guys i want to use *1.8.7.0 what versions of dahd-linux-complete and libpri should i use to work fine with a digium TE205P E1 ? |
| 16:32.02 | Qwell | eduzimrs: latest |
| 16:32.05 | Ionic | Yeah... going for svn |
| 16:32.46 | Ionic | Wow that's way better |
| 16:32.53 | Ionic | It compiled \o/ |
| 16:33.15 | eduzimrs | @Qwell yeap, but i did it before so i got on result of the command "pri show spans" the following -> PRI span 1/0: Provisioned, In Alarm, Down, Active |
| 16:33.29 | eduzimrs | @Qwell never conect its always "DOWN" |
| 16:34.00 | Qwell | sounds more like a config issue than a version issue.. |
| 16:34.29 | eduzimrs | the config its ok, i was using it before and i saved the config files |
| 16:36.06 | eduzimrs | because of this i asked about versions of dahdi and libpri |
| 16:37.27 | eduzimrs | @Qwell ideas? |
| 16:37.51 | Qwell | wonders why he bothers sometimes |
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| 16:44.35 | Katty | bothers Qwell |
| 16:44.48 | Qwell | Katty: You are allowed. |
| 16:45.13 | nhaynes | can anybody point me in the right direct? I have calls coming into my asterisk box that are answered by a switchboard operator. That operator does blind transfers to the user the call is for. I want to prepend a prefix to the caller-id so users know that the call was forwarded to them and it didn't come into through their did. |
| 16:45.52 | Qwell | nhaynes: Give the operator a separate outgoing context, that appends whatever to the CALLERID |
| 16:46.36 | nhaynes | wouldn't that also prepend that if they simply called an internal extension? |
| 16:46.53 | Qwell | I suppose so |
| 16:48.19 | [TK]D-Fender | nhaynes, ${BLINDTRANSFER} |
| 16:50.25 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-dpsyqpuxmlyvhtsa) |
| 16:53.17 | nhaynes | <[TK]D-Fender> Im not really sure what to modify in I assume extensions.conf to do that |
| 16:53.25 | nhaynes | exten => s,3,Noop(BLINDTRANSFER=${BLINDTRANSFER}) |
| 16:53.29 | nhaynes | that's what it currently looks like |
| 16:53.42 | nhaynes | i just want to prepend like FWD to blind transfers in the callerid name |
| 16:54.08 | [TK]D-Fender | nhaynes, If you want to prefix your transferred calls... then in the exten you transfer them to ... go check it. |
| 17:06.04 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:f49c:1819:b29a:cf01) |
| 17:06.10 | WIMPy | There is the Asterisk is not a PBX (toolkit) thing again. |
| 17:12.18 | *** join/#asterisk coppice (~chatzilla@183178203074.ctinets.com) |
| 17:16.44 | Ionic | Hum... what? |
| 17:16.48 | Ionic | [Oct 11 19:16:16] WARNING[155292]: pbx.c:4091 pbx_extension_helper: No application 'Wait,1' for extension (capi-in, 14595729, 1) |
| 17:17.19 | WIMPy | Typo in extensions.conf? |
| 17:17.27 | Ionic | Nah |
| 17:17.30 | Qwell | That syntax hasn't been valid in quite a long time. |
| 17:17.35 | Ionic | Uh |
| 17:17.50 | Ionic | It's what my card manufacturer provided as an example |
| 17:17.58 | WIMPy | Oh. Nostalgia. :-) |
| 17:18.01 | Qwell | That doesn't mean they're right. |
| 17:18.07 | [TK]D-Fender | WIMPy, It isn't what it used to be ;) |
| 17:18.17 | Ionic | Qwell: they were, in 2007 |
| 17:18.19 | Ionic | Probably |
| 17:18.38 | Qwell | That was 4 years and 6 major versions ago. |
| 17:18.43 | Ionic | :) |
| 17:18.45 | Ionic | Right |
| 17:20.04 | CrossWired | I have cdr setup in my context so I get all the details of connected calls using originate, but how do iget the status of those that go unanswered, or are busy? I'm using starpy |
| 17:21.10 | WIMPy | Use the dialplan to write DIALSTATUS and/or HANGUPCAUSE in to some CDR field. |
| 17:21.44 | *** join/#asterisk dijib (~nobodysho@70.24.92.19) |
| 17:21.58 | [TK]D-Fender | CrossWired, Dial a Local channel instead to do the dialout and have that leg report on it's progress |
| 17:22.13 | dijib | time to smoke a fattie |
| 17:22.41 | dijib | and then ive got a question about contexts and security. also call blocking. & followme |
| 17:22.44 | CrossWired | WIMPy: i'm doing that and its not capturing any of the nonanswered calls |
| 17:23.18 | CrossWired | [TK]D-Fender: you have any pointers for that setup? |
| 17:23.38 | [TK]D-Fender | CrossWired, Dial a local channel... how much is there to say? |
| 17:23.58 | [TK]D-Fender | CrossWired, Dialplan on both sides. you can then do stuff around the actual attempts |
| 17:24.06 | Ionic | Qwell: where do I find information about the new syntax and stuff? https://wiki.asterisk.org/wiki/display/AST/The+asterisk.conf+file or something? |
| 17:24.08 | WIMPy | If you don't get past Dial(), it seems the call was answered. |
| 17:24.31 | CrossWired | doesn't the AMI.originate command handle the dial? |
| 17:24.35 | [TK]D-Fender | Ionic, exten => extension,priority,application(parameters) |
| 17:24.45 | Ionic | Hmmm |
| 17:24.51 | [TK]D-Fender | Ionic, and it isn't "new", it's been the actual standard since before * 1.0 |
| 17:25.09 | [TK]D-Fender | Ionic, And enforcesd since 1.6 |
| 17:25.12 | [TK]D-Fender | (.0) |
| 17:25.13 | WIMPy | CrossWired: If you're using AMI, why do you need CDRs? |
| 17:25.23 | Ionic | Nah, I really need some example really, let's try some google search |
| 17:26.03 | CrossWired | WIMPy: i have CDR setup as a basic reporting for now, until I figureout how to get full data back from starpy |
| 17:26.44 | [TK]D-Fender | Ionic, ... |
| 17:26.45 | [TK]D-Fender | ~book |
| 17:26.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 17:26.58 | CrossWired | there's not alot of examples using of originate with starpy |
| 17:27.32 | CrossWired | unless i'm an idiot, which very well might be |
| 17:28.00 | CrossWired | next question, if i'm using starpy to handle my origination, should i also be using starpy to handle the dialplan? as of now its a simple dialplan |
| 17:28.15 | WIMPy | I've got no idea about starpy, but AMI is a very powrfull thing and will provide you with what you need. |
| 17:29.04 | CrossWired | one of the asterisk guys was in here last week and recommended it over the other python based library as that what they use, they actually have a fork on their github, thats why i went with it, |
| 17:29.08 | WIMPy | There's always a dialplan. But you can take over flow control if you want. |
| 17:29.20 | CrossWired | thanks for the help though |
| 17:30.50 | navaismo | ? book |
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| 18:40.23 | r0m|u | netsplit :P |
| 18:43.05 | f2knight | Q: I know lack of audio is usually related to NAT, so heres the situation... Asterisk box is on a public IP , clients are all natted. all clients phones ring. only one client is unable to hear audio. but here is the odd thing. They only can not hear audio if calling in over the DID, if I call her extension from another phone also natted she can hear it. I sould mention ALL the devices are natted on there own network scattered over th |
| 18:43.05 | f2knight | e country. Any suggestions? |
| 18:45.13 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 18:45.25 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
| 18:54.01 | navaismo | go to eat |
| 18:59.21 | *** join/#asterisk DarKnesS_WolF (~sherif@unaffiliated/sherif) |
| 18:59.36 | DarKnesS_WolF | good evening everyone |
| 19:02.31 | *** join/#asterisk DarKnesS_WolF (~sherif@unaffiliated/sherif) |
| 19:02.39 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 19:09.00 | ChannelZ | gooooood eeeeeeeveening |
| 19:11.12 | Naikrovek | hi. |
| 19:19.01 | *** join/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 19:19.08 | otwieracz | Hello. |
| 19:19.18 | otwieracz | Does Asterisk support SIMPLE in any way? |
| 19:23.23 | [TK]D-Fender | no |
| 19:23.57 | *** join/#asterisk zipytech (~ron@75-149-24-162-Pennsylvania.hfc.comcastbusiness.net) |
| 19:24.21 | zipytech | how can i increase the number of rings on an extentions |
| 19:25.12 | zipytech | Extensions |
| 19:25.24 | ChannelZ | Increase the timeout in your Dial() |
| 19:25.25 | WIMPy | 'core show application dial' see the timeout. |
| 19:25.26 | DarKnesS_WolF | oh [TK]D-Fender how are u mate it is been years :-) |
| 19:26.09 | [TK]D-Fender | DarKnesS_WolF, just under most likely... |
| 19:26.48 | ChannelZ | OT I know but anyone familiar with multicast packets and linux? |
| 19:27.18 | WIMPy | If you mention multicast RTP, it's on topic :-) |
| 19:27.42 | ChannelZ | ok let's pretend it's RTP :) |
| 19:27.57 | DarKnesS_WolF | [TK]D-Fender: cool it is been few years for me away but seems taht everyone i knew still around :-) that is cool ! |
| 19:28.01 | WIMPy | Ok, but what is the question? |
| 19:29.04 | DarKnesS_WolF | mmmm |
| 19:29.34 | ChannelZ | well, I'm trying to figure out why something isn't working. Running PS3 Media Server on a workstation on a different subnet than my PS3. PS3MS yells out on 239.255.255.250 but the PS3 never sees it. |
| 19:30.24 | otwieracz | [TK]D-Fender: Or maybe there's some proxy-like software for enabling SIMPLE support? |
| 19:30.33 | WIMPy | hmm. multicast routing. That could be interesting. |
| 19:30.47 | [TK]D-Fender | otwieracz, Nothing to do with *. |
| 19:30.56 | ChannelZ | Wondering if multicast packets are handled differently by default or should I be doing something to get them to broadcast onto the other subnet or... I have no idea how this works |
| 19:32.46 | WIMPy | As far as i can see, you either need to bridge them or you need some routing algo that will manage IGMP stuff. |
| 19:34.23 | ChannelZ | hmm. |
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| 19:37.07 | zipytech | got it thanks |
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| 19:43.41 | ChannelZ | I just can't seem to find any information either way if this is supposed to work 'automatically' or what. The TTL of the packets are only 4 so they can't go out too far but should be enough to hop subnets in my case; All LAN traffic on both interfaces is allowed in/out/forward... ifconfig says the interfaces have MULTICAST enabled |
| 19:44.45 | WIMPy | Multicast doesn't route magically. |
| 19:44.47 | ChannelZ | oh wait you said IGMP not ICMP |
| 19:45.35 | WIMPy | indeed |
| 19:47.16 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 19:47.21 | WIMPy | I'm not sure if you can du just multicast bridging. As I said: It might be interesting. |
| 19:48.10 | *** join/#asterisk navaismo (~navaismo@201.123.42.167) |
| 19:48.54 | eduzimrs | @Qwell: can u help about the issue before? |
| 19:48.54 | ChannelZ | yeah so far it's been :) |
| 19:49.54 | WIMPy | I think PIM would be the right direction. |
| 19:50.02 | ChannelZ | hmm.. if I add a rule for IGMP right at the top of my firewall I don't actually see any packets hit it. tcpdump basically just shows my workstation broadcasting UDP packets to 239.255.255.250:1900 |
| 19:50.22 | ChannelZ | PIM? |
| 19:51.09 | WIMPy | http://en.wikipedia.org/wiki/Protocol_Independent_Multicast |
| 19:51.10 | eduzimrs | anyone knows why the pri show spans status never goes to "UP" -> PRI span 1/0: In Alarm, Down, Active im having problem with e1 outgoing calls thats saying "everyone is busy...." |
| 19:51.50 | WIMPy | If it's not up, it's not working. |
| 19:52.15 | eduzimrs | i have compiled many times and never goes UP |
| 19:52.23 | WIMPy | That's a low level issue, either in configuration or the line. |
| 19:52.42 | eduzimrs | i really dont how to solve this |
| 19:53.20 | eduzimrs | WIMPy maybe a issue with dahdi and libpri versions? |
| 19:53.24 | eduzimrs | conflict? |
| 19:53.39 | ChannelZ | bad config? |
| 19:53.43 | ChannelZ | dead e1? |
| 19:53.45 | WIMPy | E1? T1? what does your dahdi/system.conf look like? have you checked cabling? Status LEDs on the NT? |
| 19:53.49 | eduzimrs | E1 |
| 19:54.04 | eduzimrs | system.conf is OK |
| 19:54.09 | WIMPy | Do you have anything else to test? Maybe a loopback plug? |
| 19:54.34 | eduzimrs | i dont |
| 19:54.41 | WIMPy | If you're sure the system.conf is correct, it must be even below that. |
| 19:55.11 | WIMPy | Status LEDs? |
| 19:55.33 | eduzimrs | ohh i have a loop cable |
| 19:55.36 | eduzimrs | LED - GREEN |
| 19:55.43 | eduzimrs | both led=green |
| 19:55.52 | WIMPy | Which one? |
| 19:55.54 | WIMPy | Which ones? |
| 19:55.58 | p3nguin | Some people's kids. |
| 19:56.09 | eduzimrs | both E1 leds |
| 19:56.23 | eduzimrs | my device is a digium Te205P |
| 19:56.28 | eduzimrs | 2 E1 |
| 19:56.28 | WIMPy | Where? The card? The NT? |
| 19:56.40 | eduzimrs | Card |
| 19:56.57 | WIMPy | Then it should be up. |
| 19:57.06 | eduzimrs | pri show spans - PRI span 1/0: Down, Active |
| 19:57.22 | eduzimrs | the ALLARM gone but it always "DOWN" |
| 19:57.23 | ChannelZ | Perhaps you forgot to pay your bill |
| 19:57.28 | WIMPy | The next step wpuld be line code in dahdi/system.conf |
| 19:58.00 | eduzimrs | span=1,1,0,ccs,hdb3,crc4 |
| 19:58.12 | eduzimrs | bchan=1-15,17-31 |
| 19:58.17 | eduzimrs | dchan=16 |
| 19:58.20 | eduzimrs | echocanceller=mg2,1-15,17-31 |
| 19:58.21 | WIMPy | Looks ok. |
| 19:58.35 | WIMPy | So chan_dahdi.conf next. |
| 19:58.39 | eduzimrs | loadzone and defaultzone = us |
| 19:59.12 | eduzimrs | i´ll paste chan_dadi 1 minute |
| 19:59.23 | WIMPy | Doesn't sound obvious when you're on E1, but that only asffects tones. |
| 19:59.35 | WIMPy | (AFAIK) |
| 19:59.42 | eduzimrs | http://pastebin.com/3ZE1feMS |
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| 20:00.32 | eduzimrs | about zones? |
| 20:00.37 | WIMPy | yes |
| 20:00.56 | eduzimrs | did u see chan_dahdi... ? look ok too yeah? |
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| 20:01.21 | WIMPy | Yes, nothing obvious to me. |
| 20:01.52 | ChannelZ | re: has your telco hit the switch? Have you any other equipment to plug the line into and see if it's actually functional? |
| 20:02.33 | WIMPy | What happens if you call a number on that line from elsewhere? |
| 20:02.36 | eduzimrs | yeap, i have already done that, the telcom is OK, and when pluged the status continues DOWN |
| 20:03.01 | eduzimrs | "EVERYONE IS BUSY OR CONGESTION AT THIS TIME...." |
| 20:03.23 | WIMPy | No, from another line. |
| 20:03.42 | eduzimrs | DAHDI/i1/08006430014-1 is circuit-busy |
| 20:03.48 | eduzimrs | nothing happens |
| 20:03.58 | WIMPy | Nothing??? |
| 20:04.21 | eduzimrs | u mean, a external call to my * server? |
| 20:04.28 | WIMPy | yes |
| 20:04.38 | eduzimrs | nothing, really |
| 20:05.01 | eduzimrs | pri status is down, will not work |
| 20:06.57 | WIMPy | I would have thought that UP/DOWN is the same that is shown on the cards LED(s). |
| 20:07.23 | WIMPy | No idea what's in between those two. |
| 20:08.13 | eduzimrs | no, when i plug off the cable pri status is as follows = "PRI span 1/0: In Alarm, Down, Active" |
| 20:08.22 | eduzimrs | "IN Alarm" |
| 20:08.43 | WIMPy | Yes, ok. |
| 20:08.51 | eduzimrs | i think down is the pri service |
| 20:09.03 | eduzimrs | maybe "service down" |
| 20:09.05 | WIMPy | Does pri debug show anything? |
| 20:10.05 | eduzimrs | many things but error, just this = PRI Span: 1 TEI=0 MDL-ERROR (G): T200 expired N200 times sending SABME in state 5(Awaiting establishment) |
| 20:10.59 | WIMPy | So no response what so ever. |
| 20:11.20 | eduzimrs | let me show my compile process |
| 20:11.29 | eduzimrs | about *, libpri, dahdi |
| 20:11.52 | eduzimrs | so i could made a mistake at compile process |
| 20:12.12 | WIMPy | You said, you have a loop cable? Have you tried that between the two ports of the card? |
| 20:12.18 | eduzimrs | yeap |
| 20:12.24 | eduzimrs | i did |
| 20:12.40 | WIMPy | Sounds unlikely if Asterisk has pri support, which it obviousely has. |
| 20:13.06 | eduzimrs | yeap |
| 20:13.24 | WIMPy | Did you get errors like: we think we're CPE and the other end thinks they're cpe as well? |
| 20:13.24 | eduzimrs | the LEDS turn GREEN and the ALARM status gone |
| 20:13.33 | eduzimrs | yeap |
| 20:13.37 | eduzimrs | no erros |
| 20:13.41 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 20:13.47 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 20:13.50 | WIMPy | Ok, good. |
| 20:13.51 | eduzimrs | they're cpe |
| 20:14.16 | WIMPy | No errors or just that error? |
| 20:14.24 | eduzimrs | no erros |
| 20:14.28 | eduzimrs | sry |
| 20:14.58 | WIMPy | Or does that only appear if both are net? Not sure :-( |
| 20:15.18 | eduzimrs | cpe_net? |
| 20:15.23 | eduzimrs | hum |
| 20:15.34 | WIMPy | You could configure one port to pri_net and see if you can talk to yourself. |
| 20:15.41 | eduzimrs | ok |
| 20:16.14 | WIMPy | If that works you know the problem is at the other end of that plug. |
| 20:16.32 | WIMPy | bbiab |
| 20:19.17 | eduzimrs | WIMPy now i got status "UP" |
| 20:20.21 | [TK]D-Fender | checkout time, BBL |
| 20:24.45 | WIMPy | Ok, so your setup seems to be ok. |
| 20:25.21 | eduzimrs | yeap |
| 20:25.35 | eduzimrs | i return to older values and guess,,, STATUS DOWN |
| 20:26.45 | SuperNull | anyone know if there is a physical difference for a hotswap backplane for SAS vs SATA ? .. yeah its not asterisk but.. :-) ILY |
| 20:29.25 | *** part/#asterisk otwieracz (~gonet9@v6.gen2.org) |
| 20:35.57 | *** join/#asterisk rainkid (~rainkid@cpe-66-65-54-130.nyc.res.rr.com) |
| 20:36.35 | rainkid | Trying to get call recording to work with Ww/Xx options. None of them do anything when hitting *1 in an active call. Am I missing something obvious? |
| 20:39.02 | eduzimrs | WIMPy: any idea? |
| 20:39.33 | eduzimrs | penelope*CLI> pri show spans |
| 20:39.34 | eduzimrs | PRI span 1/0: Up, Active |
| 20:39.34 | eduzimrs | PRI span 2/0: Up, Active |
| 20:42.44 | rainkid | Any tips, anyone? Console shows nothing upon hitting *1. monitor directory exists. |
| 20:43.49 | navaismo | rainkid do you include in the globals the DYNAMIC_FEATURES => automon |
| 20:44.12 | navaismo | and the feature is set to *1? |
| 20:44.21 | rainkid | features.conf:;automon => *1 |
| 20:44.33 | rainkid | Let me check globals section |
| 20:50.29 | WIMPy | eduzimrs: Unless you have the wrong configuration for your line, the issue lies outside of your Asterisk box. |
| 20:51.09 | WIMPy | It's the most likely version, but I don't know what your telco uses. |
| 20:53.08 | rainkid | navaismo: In extensions.conf [globals], DYNAMIC_FEATURES=>automon, in features.conf, automon => *1. Restarted asterisk. Tested. Nothing. |
| 20:53.25 | rainkid | console show no action when *1 is pressed in an active call. |
| 20:54.22 | WIMPy | rainkid: Does your DTMF work generelly? Have you tried wit Read() or by using VoiceMailMain() or something? |
| 20:54.43 | navaismo | rainkid the command features show show the *1 as current |
| 20:54.43 | rainkid | Yes. DTMF works fine for voicemail/automated systems, etc. |
| 20:54.59 | navaismo | code |
| 20:55.39 | rainkid | xten => _1NXXNXXXXXX,3,Dial(SIP/+${EXTEN}@${VOICEPULSE_GATEWAY_OUT_A},,wW) |
| 20:55.48 | rainkid | oops.. missing an 'e' there. |
| 20:56.48 | navaismo | [15:54] <navaismo> rainkid the command features show, show the *1 as current code? |
| 20:56.52 | rainkid | "One Touch Monitor *1 " |
| 20:57.13 | eduzimrs | WIMPy thanks for helping, ill call to the telco again to colect data about coding types and framing types |
| 20:57.38 | rainkid | should I be using wW or xX? Running 1.82 |
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| 20:58.20 | navaismo | for Xx use atomixmon feature |
| 20:58.46 | rainkid | I'm not familiar with atomixmon and automon. Is one preferred over the other? |
| 21:01.51 | navaismo | monitor give you 2 files, mixmonitor give one file qith the 2 audio mixed |
| 21:02.02 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:02.16 | *** join/#asterisk VaGoNeTaS (~Vag@200.111.138.172) |
| 21:02.27 | navaismo | again stupid fingers |
| 21:04.21 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-bdikbjccsvzjkxqs) |
| 21:04.27 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
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| 21:15.47 | Naikrovek | it's kind of amazing what people put in pastebin.com |
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| 21:16.11 | Naikrovek | I just click through the new pastes once in a while. lots of stuff in there. |
| 21:25.40 | p3nguin | I paste almost everything private with expiration so people don't find 'em in the new pastes list. |
| 21:27.33 | p3nguin | rainkid: You probably should be using w or x, or W or X, depending on call direction, but rarely w and W, or x and X. |
| 21:27.37 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 21:28.50 | p3nguin | Using both w and W will allow for both the callER and the callEE to toggle recording. That's typically not what people intend. |
| 21:29.12 | rainkid | Ahh. Good point. |
| 21:29.24 | p3nguin | core show application Dial |
| 21:29.28 | p3nguin | Read the options. |
| 21:29.32 | rainkid | Thank you |
| 21:29.39 | KavanS | heh I've heard of people finding data about job applicants from pastebin |
| 21:29.42 | KavanS | you'd be surprised. |
| 21:29.53 | pigpen | hey guys, having an issue with call parking. (asterisk 1.8.5.0) |
| 21:30.06 | Qwell | Naikrovek: yeah... I assume you're talking about the Callope one? |
| 21:30.09 | KavanS | a friend of mine mentioned he'd found something awhile back on a potential hire...just interesting what is pasted/stored these days |
| 21:30.11 | Qwell | Calliope* |
| 21:30.47 | p3nguin | Cantaloupe |
| 21:30.52 | pigpen | call gets put into park and immediately says "blabla got tired of being parked" |
| 21:30.59 | pigpen | and hangs up. |
| 21:31.02 | pigpen | (cool feature) |
| 21:31.06 | p3nguin | Check your timeout in features.conf. |
| 21:31.12 | pigpen | 45 seconds default. |
| 21:31.22 | p3nguin | Is it set to return back to the origin? |
| 21:31.31 | pigpen | ;-) |
| 21:31.43 | pigpen | (as I ask sheepishly) how? |
| 21:31.52 | p3nguin | There's a setting for that, too. |
| 21:31.59 | pigpen | heh?. |
| 21:32.08 | p3nguin | It's probably not important. |
| 21:32.19 | pigpen | I'll look it up. |
| 21:32.21 | pigpen | tks. |
| 21:32.25 | p3nguin | It should only do that step if the timeout is reached. |
| 21:32.37 | pigpen | oh, well, we are not that lucky. |
| 21:32.44 | p3nguin | Immediately hanging up is an interesting feature. |
| 21:32.58 | Naikrovek | Qwell: just lots of code for stuff, tons of pr0n passwords, links to 0day torrents, notes from people, encrypted text, just tons and tons of stuff |
| 21:33.01 | pigpen | yeah, gets in, then it dumps right out. |
| 21:33.18 | pigpen | seemed to be ok on 1.8.4, but after we upgraded to 1.8.5.... |
| 21:33.26 | pigpen | "boom" (silently) |
| 21:33.42 | p3nguin | Bring it up to current and it might go away. |
| 21:34.22 | Qwell | Naikrovek: xxx fan fic. |
| 21:34.52 | Naikrovek | yeah fan fic |
| 21:41.35 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-juafqjvbxxppsohq) |
| 21:50.11 | rainkid | finally got automixmon to work! |
| 21:51.25 | rainkid | Except... it records in g729. I can change the format by using MixMonitor(). Is there a way for triggering through DTMF also? |
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| 22:06.33 | Naikrovek | rainkid: i believe it just records the audio stream(s), it doesn't reencode them. |
| 22:13.48 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 22:15.09 | rainkid | Any idea how I set automixmon's filename for recorded audio? |
| 22:15.37 | rainkid | MixMonitor command is for dialplan recording, not DTMF triggered recording, so I'm not sure how to set it. |
| 22:36.19 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 22:52.53 | f2knight | is anyone using mysql with asterisk VM? |
| 23:04.23 | depressed | Is it possible to run Asterisk on 512ram and 512 swap? |
| 23:04.40 | depressed | Pardon me, I mean stable* |
| 23:07.23 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 23:14.04 | navaismo | like in an old Alix hardware yes |
| 23:14.36 | *** join/#asterisk raker (~raker@cpe-174-102-239-159.wi.res.rr.com) |
| 23:15.01 | [TK]D-Fender | depressed: Certainly. Depends on what kind of load you intend to put on it of course |
| 23:17.41 | Naikrovek | f2knight: lots of people use MySQL w/Asterisk |
| 23:17.49 | depressed | 40 trunks, 100~ concurrent calls, 20 extensions, music on hold, ulaw codec for all? |
| 23:18.10 | Naikrovek | how many teleconf rooms |
| 23:18.36 | Naikrovek | what cpu is in that server |
| 23:21.26 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:22.01 | Naikrovek | depressed: ansfer dat |
| 23:22.28 | depressed | Naikrovek: i'm getting it pulled from a rack |
| 23:22.48 | depressed | * so i can check. I've been using a third party vps and they've pissed me off |
| 23:23.17 | f2knight | Naikrovek, I am sure they do.. question is does anyone have a quick primer.? I need to know what to mod on the voicemail.conf to make it connect, and what table to add my 'users' in to. |
| 23:23.38 | Naikrovek | isn't that covered in the book? |
| 23:23.40 | Naikrovek | hmm. |
| 23:23.48 | Naikrovek | lemme dig up a URL for ye |
| 23:24.59 | Naikrovek | hmm maybe not. |
| 23:25.08 | Naikrovek | nevermind. i'm too tired to think straight |
| 23:25.32 | f2knight | Naikrovek, it might be but i did not see it. , and I am knees deep trying to figure out a why i get audio only sometimes |
| 23:25.49 | Naikrovek | well we can help with that |
| 23:26.10 | Naikrovek | but we need to see log files and call attempts and so forth and so on and some such and what not etc |
| 23:26.10 | depressed | Naikrovek: intel atom n280 1.66ghz |
| 23:26.15 | Naikrovek | atom |
| 23:26.21 | Naikrovek | that won't go far, i don't think. |
| 23:26.39 | f2knight | atom would be good for about 5 to 10 users I think |
| 23:26.52 | f2knight | with the dual 1.8 maybe 20-25 |
| 23:27.06 | f2knight | as long as it was loaded with some ram |
| 23:27.07 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 23:27.10 | Naikrovek | If you want to benchmark it with SIPP to find otu exactly what it can do, the community would appreciate it. |
| 23:27.17 | Micc | is there an easy way to get the peer name in the dialplan? |
| 23:27.33 | Naikrovek | all my * servers are dual dual core or dual quad core, but i'm oversized. |
| 23:27.58 | Naikrovek | servers can be had quite cheaply on ebay. $400 should get you a very good asterisk box, if you look in the right place(s) |
| 23:28.28 | f2knight | Naikrovek, I would love to send samples to trace it down but right now the office is empty I am waiting for someone to go over there. so i can 'test' |
| 23:28.37 | Naikrovek | ah |
| 23:28.40 | Naikrovek | okay well we're patient |
| 23:28.45 | Naikrovek | but call logs would help a lot |
| 23:30.03 | f2knight | Naikrovek, well there is a few issues.. 1) is the one phone rings, but when picked up does not hear or send audio... but only when dialed over the DID to her. If i take another device and dial her extension it all works.. |
| 23:30.18 | f2knight | and they can make outbound calls with audio bothways. |
| 23:30.33 | Naikrovek | hm |
| 23:30.38 | Naikrovek | did calls lose audio. |
| 23:30.43 | f2knight | nope |
| 23:31.00 | Naikrovek | sorry, s/did/DID/ |
| 23:31.02 | f2knight | because if I call in on the did i can dial all the other extensions and they hear and talk fine |
| 23:31.11 | Naikrovek | hmm |
| 23:31.15 | f2knight | so its not a DID issue. |
| 23:31.27 | Naikrovek | well we're gonna need some logs, when you can get them. sip debugs also will help. |
| 23:31.50 | f2knight | yea will have to wait till someone is there to pick up the phone |
| 23:31.52 | Naikrovek | i'm about to hit the sack, so it'll be someone else that helps, but be patient. |
| 23:32.01 | Naikrovek | evenings are slow in these parts |