| 00:00.30 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 00:02.10 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 00:04.20 | [TK]D-Fender | cmendes0101: funny that says it SUCCEEDED |
| 00:04.28 | [TK]D-Fender | cmendes0101: "sip show registry" |
| 00:04.30 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 00:04.55 | [TK]D-Fender | cmendes0101: <--- SIP read from UDP:206.15.156.221:5060 ---> SIP/2.0 200 OK CSeq: 138 REGISTER |
| 00:05.03 | [TK]D-Fender | cmendes0101: You're "OK"'d |
| 00:11.08 | cmendes0101 | [TK]D-Fender: ok yes it is in "sip show registry". This is another section that I was checking out. It says the same thing but keeps retrying that exceeds maximum reties. That should be ok tho since it does show up in the registry right?http://pastebin.com/c8sEVJbP |
| 00:12.01 | [TK]D-Fender | cmendes0101: Have you tried GETTING a call? |
| 00:12.27 | cmendes0101 | yes, does not work. |
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| 01:06.25 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 01:17.27 | *** join/#asterisk shaiss (~shaiss@166.205.10.119) |
| 01:18.12 | shaiss | Could anyone help me out with this http://goo.gl/2j28. I'm a new Asterisk user and would like some advice. Thank you |
| 01:19.37 | shaiss | That link is shortend and is on the digium asterisk forum |
| 01:20.05 | Russ | infobot, who is dave? |
| 01:20.06 | infobot | methinks dave is the renowned inventor of the FR-clapper |
| 01:21.34 | cmendes0101 | So I can't get incoming calls or outgoing(outgoing errors from broadvoice saying device is not registered to make calls). The sip registry is showing it is registered but will output this every once and a while http://pastebin.com/ex0sZN31 |
| 01:21.40 | Russ | shaiss, at least for me, teliax was a good option for low budget testing |
| 01:23.04 | shaiss | Looks like they're a provider, so that may be an option |
| 01:23.52 | Russ | $10 outlay for about 500 to 1,000 minutes of testing |
| 01:24.49 | shaiss | Russ, cheap enough, that's a good solution there. Thank you! |
| 01:24.58 | *** join/#asterisk coppice (~chatzilla@m121-202-81-176.smartone-vodafone.com) |
| 01:25.04 | *** join/#asterisk mutineer612 (~mutineer6@68-117-44-195.dhcp.roch.mn.charter.com) |
| 01:25.08 | shaiss | Russ, would you mind if I gave you credit on my post for that tip? |
| 01:27.12 | Russ | I don't mind |
| 01:28.05 | shaiss | Cool, thank you for the help there. Any other suggestions? |
| 01:28.34 | Russ | use softphones for testing? |
| 01:30.47 | shaiss | Once Asterisk is running on my vm machine. I'd like to be able to test out some of the features |
| 01:31.15 | shaiss | ie. conference calling, paging, check messages, ect... |
| 01:31.41 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
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| 01:32.02 | shaiss | So rather then having to go on ebay or craigslist and spend $30 or more on an actual ip phone that will not be used after the testing stage. I was thinking of using a software based phone |
| 01:32.10 | mutineer612 | Anyone have any experience with Asterisk and virtualization? I'm interested to know what sort of issues or successes have been had using esxi, xen or kdm. |
| 01:33.19 | shaiss | mutineer612, I have a similar question but looking to run AsteriskNow in a vm for testing |
| 01:33.59 | shaiss | mutineer612, I know I'm going to have to cofig vmware to bridge connection so it gets an ip on the network |
| 01:34.47 | mutineer612 | Yeah, I'm honestly interested in Switchvox but that channell seems pretty dead most of the time so I thought I would try my luck here. |
| 01:36.39 | mutineer612 | It does not sound like virtual servers can share digium hardware to guest os very well, but I'm thinking as a SIP/IAX only server that should should be able to get some kind of consolidation ratio. |
| 01:38.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 01:38.15 | shaiss | mutineer612, thats a bit above my experience. But if there on the same subnet why would they not be able to talk to each other? |
| 01:38.34 | mutineer612 | There is even mention of Switchvox offering some kind of Xen based licensing for 200 concurent users across 20 virtual machines... it's still posted on the site but I don't think they support/offer it any longer. |
| 01:39.46 | shaiss | mutineer612, I'm curious, why would you want to have multiple pbx servers? |
| 01:40.00 | shaiss | mutineer612, redundancy? |
| 01:40.30 | mutineer612 | shaiss, it's not so much an issue of IP connectivity, but rather the hardware and hypervisors ability to handle many simultaneous users, and process all of the call signaling and rtp voice streams. |
| 01:40.47 | *** join/#asterisk coppice (~chatzilla@m121-202-50-146.smartone-vodafone.com) |
| 01:42.00 | mutineer612 | shaiss, yes, redundancy and scalability. |
| 01:42.36 | shaiss | mutineer612, curious again, how many users do you have? |
| 01:47.35 | mutineer612 | No more than a few hundred at any given location... but with virtualization I'm hoping to avoid having to tie together many physical servers. It's the question of the day... is it better to have a distributed model linking on premise systems together via IAX and SIP trunking, or centralize the system in a single data center leveraging virtualization to reduce the hardware footprint and provide the redundancy (possibly usi |
| 01:47.36 | mutineer612 | ng vmotion?) needed to compete with the big boys... ie: Avaya, Cisco, Shoretel etc. |
| 01:49.34 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 01:51.17 | shaiss | mutineer612, That sounds like a cool project. Much more complex then my 12 user system ;) |
| 01:51.56 | *** part/#asterisk RobH (~robh@wikimedia/RobH) |
| 01:52.13 | mutineer612 | shaiss; lol, nothing wrong with that! My only problem is trying to get help with who to talk to about such projects. |
| 01:54.50 | shaiss | mutineer612, do you have a website? I'm curious to see what you guys do |
| 01:57.08 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 01:58.21 | mutineer612 | Sorry shaiss, cannot disclose at this time. |
| 01:58.54 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 02:03.20 | shaiss | mutineer612, np, good luck on that project |
| 02:03.22 | *** join/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 02:03.28 | nny | lol http://www.reddit.com/r/linux/comments/cxstd/tutorial_how_to_get_asterisk_set_up_and_making/ |
| 02:03.29 | mutineer612 | thx |
| 02:03.33 | nny | USE TRUXBOX111! |
| 02:04.23 | nny | i have done enough PR for the day, anyone who lurks reddit can add further. I like how the blogger bitched about the crankiness of the channel, how dare free help be so unaccomidating |
| 02:04.47 | nny | unaccommodating* |
| 02:06.41 | TJNII | I used to be a redditor. Then I happened across one of their pro-CP threads and decided I didn't want to be a part of that community. |
| 02:07.23 | nny | well, it's not really a blanket community |
| 02:07.30 | TJNII | True. |
| 02:07.34 | nny | you can create reddits that discuss your love of undercooked fish |
| 02:07.54 | nny | although I imagine something as henious as that was removed |
| 02:08.05 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 02:08.13 | nny | it's owned by Conde Naste. I doubt they want CP related to their service |
| 02:08.41 | nny | just goes to shows the world is comprised of an infinte number of assholes :D |
| 02:08.47 | nny | er types* |
| 02:09.01 | nny | the former feels true sometimes though lol |
| 02:09.03 | TJNII | No, I'd say assholes is correct. |
| 02:09.07 | nny | :D |
| 02:09.43 | TJNII | Especially now that the internet is promoting two-tone perception disorder and everybody thinks anyone who doesn't agree with them must be an ass. |
| 02:10.08 | shaiss | infobot, what is CP? |
| 02:10.09 | infobot | it has been said that cp is Copies one or more files to another file or directory. Syntax: cp (file1) (file2) ...(fileN) (destination). Where file1 through fileN are the files to copy, and destination is the destination file or directory. Example: cp ../frog joe copies the file ../frog to the file or directory joe. |
| 02:10.35 | *** part/#asterisk shaiss (~shaiss@166.205.10.119) |
| 02:11.43 | *** join/#asterisk snadge (~snadge@starbug.ugh.net.au) |
| 02:12.00 | snadge | question regarding using sipdroid as a voip client to asterisk |
| 02:12.14 | snadge | it seems to default to using 64kbit pcm over wifi.. is this acceptable? |
| 02:12.29 | snadge | it works alright, but im having an issue with dtmf |
| 02:12.39 | snadge | i press the number once.. and it triggers 4 or 5 times |
| 02:12.54 | nny | um |
| 02:12.57 | snadge | so i cant access my voicemail etc |
| 02:13.01 | nny | lol infobot |
| 02:13.34 | snadge | im actually the admin for the asterisk server.. so i thought maybe i can tweak the protocol or try something different |
| 02:14.01 | snadge | it works with other soft phones though.. so i blame sipdroid ;) |
| 02:15.05 | snadge | hmm.. maybe i should enable g722 support |
| 02:15.21 | snadge | but thats probably proprietary and requires violating several laws |
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| 02:20.16 | *** part/#asterisk nny (~Scott@cpe-071-076-058-253.sc.res.rr.com) |
| 02:24.40 | *** join/#asterisk coppice (~chatzilla@m121-203-209-59.smartone-vodafone.com) |
| 02:33.05 | snadge | ergh it appears dtmf problems are common with sip clients |
| 02:33.08 | snadge | particularly android ones |
| 02:33.32 | snadge | fucking useless.. maybe i should've just bought an iphone |
| 02:36.13 | *** join/#asterisk jetlag (jetlag@pool-173-61-213-156.cmdnnj.east.verizon.net) |
| 02:39.00 | drmessano | I wouldn't hold that against someone |
| 02:39.20 | drmessano | No, really, don't hold your iPhone against someone.. Backhoe fade |
| 02:40.35 | mutineer612 | what dtmf mode does you sip client support? |
| 02:41.04 | mutineer612 | RFC2833? |
| 02:42.33 | mutineer612 | Does your client have the same problem even when mashing the numbers on the dialpad? |
| 02:43.56 | *** join/#asterisk sogi (sogi@triton.intrak.tuke.sk) |
| 02:43.59 | sogi | hi all |
| 02:44.06 | sogi | I have a problem |
| 02:44.23 | sogi | calling ext# to ext# |
| 02:44.35 | sogi | it ring 2 times and then goes to voicemail |
| 02:44.37 | sogi | Nobody picked up in 20000 ms |
| 02:44.43 | sogi | but it dont wait 20s |
| 02:44.52 | sogi | where can be the problem? |
| 02:45.06 | fenrus | pastebin the dialplan |
| 02:46.05 | sogi | it is freepbx machine |
| 02:46.10 | sogi | so dialplan is so wid |
| 02:46.10 | sogi | e |
| 02:46.18 | fenrus | go to #freepbx |
| 02:46.29 | fenrus | ~freepbx |
| 02:46.29 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 02:47.09 | sogi | yes sorry...I only wondering if somebody dont have the same problem with pure asterisk |
| 02:49.17 | *** join/#asterisk NateHB (~chatzilla@hoc.handsonconsultants.com) |
| 02:53.21 | *** join/#asterisk xNinja (3e96a3ac@gateway/web/freenode/ip.62.150.163.172) |
| 02:53.40 | xNinja | hello...i have question about spa3000...as i am new to voip "sip" stuff |
| 02:54.29 | xNinja | does the spa3000 act as a sip proxy server ? so i can through my net in my cell phone connect to my spa3000 at home and dial using my phone land line and vise versa ? |
| 02:54.59 | xNinja | like when i am outside the country i can using the net to pickup my home line to dial free inside my country ? |
| 02:55.59 | sogi | you can use your cell internet connection to connect spa3000 as sip client |
| 02:56.04 | sogi | if spa3000 knows it |
| 02:56.10 | sogi | asterisk sure do :) |
| 02:57.59 | xNinja | in my case i need asterisk server ? |
| 02:58.32 | sogi | http://www.voipuser.org/review_8.html |
| 02:58.34 | xNinja | or spa3000 is enough to handle 1 line to just cut the costs of my cell phone bells |
| 02:58.44 | sogi | if it cannot act as sip server |
| 02:58.47 | sogi | it wont work |
| 03:02.03 | xNinja | so it acts as sip server? i am getting confused because i see peoples using it with asterisk i am wondering why |
| 03:02.32 | xNinja | if it can do the job as sip server and using my home land line...thats all i want then why i need asterisk |
| 03:02.43 | xNinja | for big space of voice mails or complex setup ? |
| 03:04.12 | xNinja | because in my country using land line to call landline or to call cellphone its free of charge for both persons....so i just want a home sip server to dial through my home land line so that i wont pay any to the cellphone carrier.......so spa3000 is the solution for me right ? |
| 03:04.17 | xNinja | just want to be sure :D |
| 03:09.34 | drmessano | SPA-3000 cannot be a "SIP Server" |
| 03:14.13 | *** join/#asterisk jetlag (~jetlag@pool-173-61-209-62.cmdnnj.east.verizon.net) |
| 03:17.58 | xNinja | any recommended cheap sip server for home ? |
| 03:18.05 | xNinja | embedded' |
| 03:23.29 | drmessano | embedded? |
| 03:24.32 | xNinja | what do i need for my situation >? |
| 03:24.52 | WIMPy | Some people install Asterisk on their router. |
| 03:25.28 | drmessano | Some people with more than beginner asterisk knowledge do, yes |
| 03:26.31 | drmessano | xNinja: Spend $100 or so and get a nice Atom box from Newegg.. add some RAM and an HD and now you have a small PC for asterisk |
| 03:27.14 | drmessano | You should also look up "embedded", because you're misusing the term here a bit |
| 03:27.31 | xNinja | why asterisk and why complex solution isnt there an embedded sip server to use for my own,..i dont even know if the term sip server is the right term for my situation |
| 03:28.10 | drmessano | shakes his head |
| 03:28.22 | drmessano | What do you want your SIP server embedded in? |
| 03:28.25 | xNinja | do you fully understand what i need >? |
| 03:28.32 | drmessano | Do YOU? |
| 03:28.51 | DogBoy | you can install it on a dockstar too |
| 03:28.52 | WIMPy | Some routers have such functionality readily available, like e.g. Darytek vigors, but I don't know if they allow external SIP clients. |
| 03:29.03 | DogBoy | they're on sale now at amazon for 30 bux |
| 03:29.45 | snadge | im not sure what dtmf mode sipdroid uses |
| 03:29.48 | DogBoy | drmessano is right though it's not for the new asterisk user |
| 03:30.01 | snadge | apparently dtmf has never worked properly on any android software voip clients ;) |
| 03:30.09 | xNinja | all i want is something small in my home which can act as sip server/proxy whatever...something i can connect to it from my cell phone internet so that i dial from the cell phone sip client "fring" for example to pickup my phone landline so that i wont pay anything to cell phone carrier |
| 03:30.37 | DogBoy | sounds weird |
| 03:31.04 | DogBoy | you want to make calls over wifi on your phone xNinja ? |
| 03:31.18 | drmessano | xNinja: You need (1) A device to interface with the analog line and (2) a SIP proxy/server.. SPA-3102 is a good start, and maybe Milkfish running on DD-WRT |
| 03:31.23 | drmessano | But that's 2 boxes |
| 03:31.41 | xNinja | not wifi, i have an internet connection in my cellphone which i pay the cellphone isps for it |
| 03:31.53 | drmessano | DogBoy: Cell phone SIP client > Wifi > SIP server > <Interface to Analog home phone line> |
| 03:32.18 | drmessano | So he can use his cell for the handset, but use the analog home phone for the call |
| 03:32.21 | xNinja | draytek has it all ? i saw it but i dont know if its a full solution for me ? |
| 03:32.31 | DogBoy | I wonder how well that would work |
| 03:32.34 | drmessano | xNinja: No |
| 03:32.38 | xNinja | drmessano: exactly http://cgi.ebay.com/UNLOCKED-LINKSYS-SPA3000-SIP-VOIP-FXS-FXO-PHONE-ADAPTER_W0QQitemZ330431894712QQcategoryZ99269QQcmdZViewItemQQ_trksidZp3907.m263QQ_trkparmsZalgo%3DSIC%26itu%3DUCI%252BIA%252BUA%252BFICS%252BUFI%252BDDSIC%26otn%3D8%26pmod%3D330440055859%26po%3DLVI%26ps%3D63%26clkid%3D6605664481723324845 |
| 03:32.39 | drmessano | xNinja: No interface to the phone line |
| 03:32.40 | xNinja | ops |
| 03:32.46 | DogBoy | I don't think I would make calls like that will driving for instance xNinja |
| 03:32.51 | DogBoy | while |
| 03:33.17 | drmessano | xNinja: So SPA-3102 + A SIP Server is going to be two different pieces of hardware then.. |
| 03:33.45 | WIMPy | xNinja: I don't have experience with voice on the Drayteks, but I'd expect it to do all you need. |
| 03:33.53 | drmessano | .... |
| 03:34.03 | drmessano | The Draytek has an analog phone interface? |
| 03:34.04 | xNinja | thereis a windows software pbx...do i just need that with a digium card for example and thats it ? |
| 03:34.08 | DogBoy | were they like a badguy on dr who |
| 03:34.10 | drmessano | and runs a SIP server? |
| 03:34.21 | WIMPy | drmessano: Either FXO/FXS or S0. |
| 03:34.32 | drmessano | xNinja: No, Windows SIP server wont work |
| 03:35.07 | snadge | whats the best protocol to use between an android phone and asterisk server |
| 03:35.09 | drmessano | There's nothing that will work with a Digium card that I have ever heard of |
| 03:35.15 | snadge | it seems to only be able to negotiate PCMU or PCMA |
| 03:35.22 | snadge | whatever they are.. and both seem to have an issue with dtmf |
| 03:35.34 | drmessano | snadge: PCMU and PCMA are codecs, not protocols |
| 03:35.44 | xNinja | "any easy solution" ? |
| 03:35.44 | snadge | ok sure.. whats the best codec to use then |
| 03:35.53 | drmessano | xNinja: There is none |
| 03:35.53 | xNinja | if router it would be so great |
| 03:36.02 | drmessano | xNinja: Been trying to tell you that.. and I am done wasting my time |
| 03:36.21 | drmessano | xNinja: I gave you several scenarios and explanations.. Good luck to you |
| 03:36.42 | xNinja | thanks a lot drmessano i really appreciate it...sorry for wasting ur time |
| 03:37.30 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:42.13 | DogBoy | <xNinja> do you fully understand what i need >? |
| 03:42.13 | DogBoy | <drmessano> Do YOU? |
| 03:42.15 | DogBoy | lol |
| 03:42.35 | DogBoy | back to the drawing board xNinja |
| 03:42.53 | xNinja | ? |
| 03:43.02 | xNinja | i am searching but confused |
| 03:44.54 | xNinja | DogBoy: can you point me to what i have to search for exactly for my case ? |
| 03:45.07 | xNinja | pbx home router for example ? |
| 03:46.27 | DogBoy | get the book |
| 03:46.37 | DogBoy | read it or at least read parts of it |
| 03:46.50 | DogBoy | what you're asking about is not straight forward by any means |
| 03:47.13 | DogBoy | it is doable but imo is not going to work that well |
| 03:47.18 | WIMPy | wonders how that is going to help select te right product. |
| 03:47.41 | DogBoy | knowing what you are doing is a great place to start |
| 03:47.59 | DogBoy | selecting a product is not the first step |
| 03:48.59 | WIMPy | If what he told us all his needs, it sounds more like the only step. |
| 03:49.47 | xNinja | :) |
| 03:50.06 | carrar | do you fully understand what he needs |
| 03:50.17 | DogBoy | what us? I said what I have to say, step in and take over WIMPy |
| 03:50.37 | carrar | not 99% mind yo u |
| 03:50.44 | carrar | FULLY |
| 03:50.53 | DogBoy | 1. here is what you buy 2. now set it up like this |
| 03:50.59 | WIMPy | What I understand is just a device that lets you connect a SIP phone to a POTS line. |
| 03:51.26 | DogBoy | yea but you're kind of misguiding him possibly |
| 03:51.32 | DogBoy | yea it's easy just buy this |
| 03:52.09 | DogBoy | it's not easy and buying something is what you do after you figure out what you are doing |
| 03:52.11 | carrar | xNinja, I don't fully understand what you need |
| 03:52.40 | WIMPy | Not a big deal, but 1. some might not allow the sip client on the wan side and 2. might not be easily available in the US. But the Vigor sounds like a good bet. |
| 03:52.57 | DogBoy | he wants to make sip calls over his data plan on cellphone to a device at home that will proxy the call out to POTS |
| 03:53.25 | xNinja | i think wimpy said it very simply connect a sip phone to POTS line "landline" if that what it means, so i can diaup through the sip phone using the landline...thats it |
| 03:53.26 | carrar | cake |
| 03:53.29 | carrar | * can do that |
| 03:53.32 | carrar | make it happen |
| 03:53.53 | xNinja | DogBoy: no no |
| 03:54.20 | carrar | dogboy, doesn't fully understand what you need |
| 03:54.25 | xNinja | i just want my sip client to connect to my home line to use it for calling....thats it |
| 03:54.30 | DogBoy | or care much |
| 03:54.33 | carrar | heh |
| 03:54.41 | xNinja | carrar: :D |
| 03:54.58 | DogBoy | it's like a homework question anyway |
| 03:55.00 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 03:55.02 | carrar | xNinja, assuming you can make SIP calls from your cell phone and you have a outside IP at home ye |
| 03:55.05 | carrar | s |
| 03:55.18 | xNinja | yeah |
| 03:55.21 | carrar | ~book |
| 03:55.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 03:55.30 | carrar | go forth and prosper |
| 03:57.18 | xNinja | [06:50] <WIMPy> What I understand is just a device that lets you connect a SIP phone to a POTS line. |
| 03:57.29 | xNinja | carrar: is that the same as what you mean ? |
| 03:57.36 | carrar | What do you mean |
| 03:57.49 | *** join/#asterisk ajm0716 (~ajm0716@c-76-97-74-68.hsd1.ga.comcast.net) |
| 03:57.54 | carrar | You need some hardware |
| 03:58.01 | carrar | with the right hardware * will do what you need |
| 03:58.04 | *** join/#asterisk shido6 (~shido6@c-24-4-217-126.hsd1.ca.comcast.net) |
| 03:58.12 | xNinja | yeah hardware to forward the calls ? |
| 03:58.26 | DogBoy | heh |
| 03:58.26 | carrar | Hardware for * to connect to a POTS LINE |
| 03:58.43 | xNinja | like sangoma card ? |
| 03:58.48 | xNinja | spa3000 cant do that ? |
| 03:59.02 | carrar | I'll recommend Digium hardware |
| 03:59.37 | xNinja | so again i have to make a server... |
| 03:59.54 | carrar | like a TDM410P Card |
| 04:00.04 | carrar | with a FXO card |
| 04:00.07 | DogBoy | yes absolutely you need to run asterisk |
| 04:00.15 | carrar | yes |
| 04:00.18 | WIMPy | xNinja: The spa can connect to your line, but you can't connect your phone to it. You'd have to run Asterisk somewhere to connect both. |
| 04:00.25 | carrar | doesn't need to be a high end server |
| 04:00.27 | xNinja | i dont mind test and get the card but the thing is...i want something small and easy because i want to give it to my family/friends as a solution to cut the cell phone costs |
| 04:01.07 | carrar | build your own PC then |
| 04:01.10 | xNinja | WIMPy: you mean spa can let my sip client connect to my landline and dial ? |
| 04:01.10 | carrar | be creative |
| 04:01.30 | carrar | Let your Imagination run with the flying horses |
| 04:01.37 | WIMPy | xNinja: No, I said no. |
| 04:01.44 | WIMPy | Not on it's own. |
| 04:01.46 | xNinja | carrar: i told u its not the problem...i can read and test but i need easy solution because of them |
| 04:02.16 | DogBoy | this is mildly entertaining I have to admit |
| 04:02.22 | carrar | xNinja, What did I recommend then? |
| 04:02.31 | carrar | please repeat |
| 04:02.34 | WIMPy | Look at the Vigors or do DIY mini-PCs. |
| 04:02.40 | carrar | I need to FULL KNOW YOU UNDERSTAND what I said |
| 04:02.46 | carrar | FULLY |
| 04:03.03 | carrar | COMPLETELY |
| 04:03.07 | carrar | 100% |
| 04:03.14 | carrar | Like a Captin to a private |
| 04:03.29 | carrar | get it and get it right the first time |
| 04:04.03 | carrar | HARRO |
| 04:05.07 | xNinja | i appreciate the help and you trying to help carrar but please dont try to make fun of me... |
| 04:05.17 | carrar | Oh by no means |
| 04:05.45 | carrar | I'm nothing fully serious in this channel |
| 04:05.48 | carrar | err |
| 04:05.51 | carrar | I'm nothing but fully serious in this channel |
| 04:07.02 | carrar | Just go buy a Zotac Intel Atom |
| 04:07.05 | carrar | and mini case |
| 04:07.10 | carrar | and Hard Drive |
| 04:07.25 | carrar | and find a cheap ATA |
| 04:08.15 | carrar | off Ebay |
| 04:08.15 | xNinja | its all about the card right |
| 04:08.15 | carrar | What card? |
| 04:08.15 | carrar | if you go with a minipc you don't have room for a Digium card |
| 04:08.15 | xNinja | cards like digium |
| 04:08.43 | carrar | and you'll need a SIP to POTS device (ATA) |
| 04:08.50 | DogBoy | you can use one of these for the pc: |
| 04:08.51 | DogBoy | http://www.amazon.com/Seagate-FreeAgent-DockStar-Network-STDSA10G-RK/dp/B002MRRU6G/ref=sr_1_11?ie=UTF8&s=electronics&qid=1281042532&sr=8-11 |
| 04:08.57 | DogBoy | 30 bux |
| 04:09.00 | DogBoy | haha |
| 04:10.41 | carrar | xNinja, you want cheap and crappy or most expensive and quality? |
| 04:10.49 | carrar | s/most/more/ |
| 04:11.17 | xNinja | cheap but not crappy :D something for home use 1 user only |
| 04:11.21 | DogBoy | cheap is not necessarily crappy |
| 04:11.26 | xNinja | yep |
| 04:12.12 | carrar | guess your stuck with a $30 Ebay SPA3000 |
| 04:12.59 | carrar | any other questions? |
| 04:13.11 | xNinja | what i need worth going in range of $600 |
| 04:13.22 | carrar | depends |
| 04:13.32 | carrar | Audiocodes are much nicer |
| 04:14.01 | DogBoy | translates, that means it's gonna sound nicer |
| 04:14.05 | carrar | Item image |
| 04:14.06 | carrar | Audiocodes MP104 MP-104 FXO Analog Gateway SIP VoIP |
| 04:14.18 | carrar | get more lines |
| 04:14.21 | carrar | do more calls |
| 04:14.27 | carrar | have more controls |
| 04:14.27 | xNinja | i am getting sick being confused of all those terms pbx/fxo spa3000 is doing what i need and not doing :( |
| 04:14.34 | carrar | ~$400 ebay |
| 04:14.49 | xNinja | i just want something simple very simple for one user one line |
| 04:15.13 | xNinja | sip client to home landline to make calls thats all |
| 04:15.33 | carrar | buy the cheap spa3000 then |
| 04:15.42 | xNinja | it will do what i need !? |
| 04:15.56 | carrar | if you can figure out how to configure it |
| 04:16.18 | xNinja | after all it will ? |
| 04:17.03 | carrar | http://www.voip-info.org/wiki/view/Asterisk@Home+Handbook+Wiki+Chapter+5#5312SPA3000FXOasaVoIPToPSTNGatewayOutbou |
| 04:17.07 | xNinja | do i really have to read the book and understand all those things :D |
| 04:17.25 | carrar | You just want it to magically work? |
| 04:17.29 | carrar | everything? |
| 04:17.36 | DogBoy | yes |
| 04:17.44 | DogBoy | it might be a good idea if you knew what you were doing |
| 04:17.53 | DogBoy | nobody is going to do it for you |
| 04:18.04 | carrar | You are going to need to plug into the MATRIC NEO and upload the asterisk module |
| 04:18.04 | shido6 | well |
| 04:18.08 | carrar | MATRIX |
| 04:18.13 | shido6 | thats what switchvox is for |
| 04:18.21 | carrar | yeah |
| 04:18.25 | shido6 | but um..... |
| 04:18.27 | shido6 | <ahem> |
| 04:18.28 | carrar | but thats not free |
| 04:18.29 | shido6 | its not free |
| 04:19.08 | DogBoy | he wants to make some cheap thing he can give friends and family |
| 04:19.37 | carrar | Personally I'd go with finger painting with water colors |
| 04:19.42 | DogBoy | yea |
| 04:20.04 | DogBoy | xNinja, get the book, read up, anything else you are WASTING YOUR TIME and OURS |
| 04:20.44 | carrar | If you are going to use Asterisk (which is what this channel is about) you're going to need to learn it |
| 04:21.19 | xNinja | thanks for everythings i appreciate it |
| 04:21.36 | carrar | just use skype |
| 04:21.44 | carrar | it's free |
| 04:21.47 | carrar | it's one click |
| 04:22.24 | xNinja | np:) |
| 04:31.04 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 04:31.29 | drmessano | lol |
| 04:37.39 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 04:42.08 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 04:54.18 | ChannelZ | scuse me, ummm, which buttons do I press to call on the Skypes? k thanks |
| 05:00.57 | carrar | The shinny red candy lik button! |
| 05:03.27 | ChannelZ | teh one with the vibrator on it? |
| 05:03.50 | shido6 | the one with the $ on it |
| 05:09.16 | *** part/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 05:11.24 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 05:22.00 | drmessano | Do the Skype's use's the tube's? |
| 05:25.03 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.194) |
| 05:30.20 | *** join/#asterisk Piffer (~piffer@ip70-173-209-212.lv.lv.cox.net) |
| 05:31.21 | Piffer | Anyone got AsteriskNOW to see the hard drive in VMware Server 2.x ? Hoping to take it for a spin, before putting it on a physical box. |
| 05:33.36 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 05:36.03 | *** join/#asterisk HarrisAsterisk (~harisalip@119.155.116.253) |
| 05:37.02 | Piffer | ah fixed, removed scsi and added ide... |
| 05:37.56 | HarrisAsterisk | hi, plz help, I want to set DID for Dahdi FXO channel (dummy DID for POT FXO) using asterisk 1.4 |
| 05:40.50 | HarrisAsterisk | please |
| 05:41.15 | Piffer | I have no idea how to do that. sorry |
| 05:46.10 | HarrisAsterisk | thanks Piffer for interest. Any one have idea ?????? |
| 05:48.27 | kaldemar | HarrisAsterisk: be more elaborate on what you want. |
| 05:49.37 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 05:50.58 | ChannelZ | an FXO channel = connecting to POTS. Your DID is under control of the phone company. |
| 05:51.57 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 05:54.50 | HarrisAsterisk | yes. i am using POTs line, not PRI , SIP. so phone company didn't assign DID on pot lines |
| 05:55.18 | HarrisAsterisk | so I want to assign dummy DID for each FXO channels |
| 05:57.26 | HarrisAsterisk | I read on web that it is possible with asterisk 1.4, but method is not define, which edition in conf file should i made ? (also there is some example using FreePBX but i am not using FREEpbx) |
| 06:04.08 | carrar | something like |
| 06:04.08 | carrar | callerid="Jenny <8675309>" |
| 06:04.08 | carrar | channel => 1 |
| 06:09.15 | Russ | what is the best way to avoid expression expansion in AEL? |
| 06:09.42 | kaldemar | HarrisAsterisk: define "assign DID". do you mean setting a caller id for outgoing calls? |
| 06:10.01 | carrar | well he can't set callerid for outbound pots line |
| 06:10.11 | carrar | so has to be inbound |
| 06:11.06 | carrar | Russ, don't use it? |
| 06:11.43 | *** join/#asterisk programmerq (~jefferya@saffron.programmerq.net) |
| 06:12.14 | Russ | you mean AEL? |
| 06:13.50 | Russ | I'd expect: |
| 06:13.52 | Russ | 211 = "health-human"; |
| 06:13.56 | Russ | to work, but it does not |
| 06:13.57 | programmerq | Hello-- I'm trying to design a tech support chat software. Using an existing PBX (especially if both support chats and calls were taken) seems like a good idea in theory-- queues are already implemented, conference features, transfers, etc. As best as I can tell, no one has done this with asterisk. Would re-purposing asterisk to take chats instead of calls be feasible, or am I completely going in the wrong direction? |
| 06:14.44 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-pcgrheadaqktzugf) |
| 06:15.30 | carrar | Why not just install a public XMPP server for chats? |
| 06:16.20 | carrar | use a web based client as the front end |
| 06:17.00 | programmerq | carrar: XMPP is designed to use an IM model-- all those features of a PBX aren't there like queues, conferences, transfers, etc. |
| 06:17.54 | carrar | ok public IRC server |
| 06:17.59 | programmerq | same problem |
| 06:18.02 | programmerq | no queues |
| 06:18.04 | carrar | what problem |
| 06:18.05 | programmerq | no transfers |
| 06:18.13 | programmerq | etc |
| 06:18.40 | ChannelZ | HarrisAsterisk: I really have no idea what you're asking |
| 06:18.49 | carrar | use a irc server with php front end |
| 06:18.52 | programmerq | my question isn't about the obvious low hanging fruit when it comes to solving this problem-- I'm trying to think outside the box and do it different |
| 06:19.01 | ChannelZ | The phone company has to assign you DIDs or nobody can call you. |
| 06:19.03 | carrar | have it pop people in after checks |
| 06:19.15 | carrar | to the correct channel |
| 06:19.25 | carrar | for 1 on 1 support |
| 06:20.24 | programmerq | carrar: my question isn't about the best way to solve this problem-- my question is specifically "Is this feasible to re-purpose asterisk to be used for chats?" |
| 06:20.30 | programmerq | but thank you for the input |
| 06:20.43 | HarrisAsterisk | not callerID , i want to set DID (DNIS), for inbound calls |
| 06:20.48 | carrar | asterisk is voip |
| 06:21.16 | programmerq | carrar: it was designed for voice, but it can also do video, so why couldn't it do text too? |
| 06:21.35 | HarrisAsterisk | callerid ="jenny <8348904904>" did not set DID but set Caller id |
| 06:21.36 | programmerq | wouldn't I just have to implement my own "codec"? |
| 06:22.01 | ChannelZ | HarrisAsterisk: You want to see what number someone dialed in on? Is that it? |
| 06:23.37 | *** join/#asterisk oej (~olle@95.209.106.251.bredband.tre.se) |
| 06:24.41 | Piffer | Sounds like HarrisAsterisk wants to setup some DIDs on his end, so incoming calls to a numer he configured will be answered by his Asterisk server. |
| 06:25.17 | Piffer | But with a pots line, the telco has already set a number for that line. |
| 06:25.43 | ChannelZ | that being the case, it's called "configuring chan_dahdi.conf" |
| 06:26.13 | Piffer | with a pots line, there's only one number as far as I know, otherwise he'll need a PRI or SIP trunk |
| 06:27.23 | HarrisAsterisk | yes. want to find dialed number (DID/DNIS) |
| 06:28.33 | Piffer | ah that's different... and I have no idea how to do that, so I'm going to bed. Good night everyone ;-) |
| 06:28.34 | *** part/#asterisk Piffer (~piffer@ip70-173-209-212.lv.lv.cox.net) |
| 06:28.41 | kaldemar | HarrisAsterisk: that's not something you get on FXO. |
| 06:29.09 | kaldemar | HarrisAsterisk: caller id is all you can get. |
| 06:29.13 | HarrisAsterisk | Piffer you are almost right. PRI/SIP provide DID by self but not POTS lines, so I need to set dhadi to provide dummy DID on incomming call |
| 06:30.58 | ChannelZ | You know the phone number associated with a phone line. You know what port it's plugged into on your TDM card. If line 1 is 555-555-5555, plugged into channel 1, any call coming in on DAHDI/1 must be because they dialed 555-555-5555. Set a channel variable if you want |
| 06:31.13 | HarrisAsterisk | but if you see it can be set in Freepbx, & some people has been set this, according to my web search, but how they |
| 06:31.13 | kaldemar | HarrisAsterisk: you can't set DAHDI to do it. you must do it in your dialplan (extensions.conf), though i can't see a point in setting it. |
| 06:31.15 | ChannelZ | Or if you just want to handle the call differently, set a different context for each channel |
| 06:32.21 | HarrisAsterisk | one channel will be able to handle different campaign based on did |
| 06:32.31 | ChannelZ | context=line1 |
| 06:32.34 | ChannelZ | channel => 1 |
| 06:32.38 | ChannelZ | context = line2 |
| 06:32.42 | ChannelZ | channel => 2 |
| 06:32.49 | WIMPy | Interesting stuff. Nerver thought about such things. |
| 06:33.34 | kaldemar | HarrisAsterisk: handling FXO's by DID is just making things more complex. use different contexts as ChannelZ said. |
| 06:33.50 | HarrisAsterisk | yes kaldemar, but DID/CID are channels variable which can not be set from dialplan |
| 06:34.06 | ChannelZ | or you can send them all to the same context, and look at ${CHANNEL} to and GotoIf your head off depending on what channel it came in on |
| 06:34.58 | ChannelZ | HarrisAsterisk: You can set your own variable.. again in chan_dahdi.conf: |
| 06:35.09 | ChannelZ | setvar=DID=555-555-5555 |
| 06:35.12 | ChannelZ | channel => 1 |
| 06:35.24 | ChannelZ | setvar=DID=888-888-8888 |
| 06:35.26 | ChannelZ | channel => 2 |
| 06:35.58 | ChannelZ | then ${DID} should contain whichever depending on what channel. Though that doesn't seem useful unless you're just going to log that somewhere |
| 06:36.03 | HarrisAsterisk | waaah good. thnaks ChannelZ, hope this hint can do something |
| 06:36.39 | ChannelZ | If you just want to handle the calls differently just separate them by context |
| 06:36.44 | *** part/#asterisk programmerq (~jefferya@saffron.programmerq.net) |
| 06:41.03 | HarrisAsterisk | hummmm agree. actuall on POTs _X. didn't work, so I use extension s, which didn't launch DID base routing, so i was unable to seperate by dialed numer for pots lines. But i will now apply your suggestions first "var in dahdi" :-) then context |
| 06:41.17 | HarrisAsterisk | Thanks a lot for all of you |
| 06:41.26 | ChannelZ | That's why I said separate them by context |
| 06:41.50 | ChannelZ | It's easier than writing a bunch of 'GotoIf' and such in the dialplan to look at a DID variable |
| 06:42.06 | Russ | using the 'console dial <extension>@<context>' is really handy, but I can't seem to set the outgoing callerid |
| 06:42.17 | Russ | or, at least ${CALLERID(num)} doesn't show anything |
| 06:44.05 | ChannelZ | did you set the callerid for the console channel in console.conf? |
| 06:44.18 | Russ | I did, format +1NXXNXXXXXX |
| 06:44.22 | HarrisAsterisk | agree, but in near future I have to replace POTS with PRI & SIP line, where campaign will have to seperated by DID. I mean on single PRI different DID will be received for various campaigns. so i was looking to set server in same direction |
| 06:44.24 | HarrisAsterisk | thansk |
| 06:45.24 | HarrisAsterisk | ChannalZ i am implementing your suggestion right now |
| 06:46.01 | ChannelZ | Russ what does the extension that you're dialing from the console do? |
| 06:46.45 | Russ | a whole bunch of stuff |
| 06:47.01 | Russ | but I have a Verbose(calledid ${CALLERID(num)}) in there |
| 06:47.10 | Russ | I see callerid, but no number after it |
| 06:52.36 | Russ | simple test case would be context foo { 111 => { Verbose(from ${CALLERID(num)}); }; }; |
| 06:52.44 | Russ | console dial 111@foo |
| 06:53.01 | ChannelZ | Abort, Retry, Fail: |
| 06:53.10 | Russ | r |
| 06:53.39 | ChannelZ | sorry I dunno |
| 06:57.22 | HarrisAsterisk | ChannelZ setvar is not working, is supported in 1.4 ? |
| 06:57.33 | *** join/#asterisk Tim_Toady (~moi@77.49.3.102.dsl.dyn.forthnet.gr) |
| 06:57.46 | *** join/#asterisk rootlinux (~rootlinux@201.143.28.20.dsl.dyn.telnor.net) |
| 06:57.57 | HarrisAsterisk | i use setvar=DID=6670541 |
| 06:57.58 | HarrisAsterisk | channel => 3 |
| 06:58.41 | HarrisAsterisk | chan_dahdi |
| 06:58.42 | HarrisAsterisk | setvar=DID=6670541 |
| 06:58.42 | HarrisAsterisk | channel => 3 |
| 06:58.42 | HarrisAsterisk | dialplan |
| 06:58.42 | HarrisAsterisk | noop( ${DID} |
| 06:59.30 | *** join/#asterisk hariom (~hariom@122.170.17.101) |
| 06:59.49 | ChannelZ | did you reload? |
| 07:00.08 | hariom | which default prompts should be used? sln16 or gsm? |
| 07:00.31 | ChannelZ | hariom: whatever your most common codec being used is |
| 07:00.41 | ChannelZ | ulaw is good as it can be transcoded into the others |
| 07:01.08 | hariom | I have been using gsm till now. But I heard that quality is poor. |
| 07:01.09 | ChannelZ | sln16 isn't of much use unless you have devices supporting other high bandwidth codecs |
| 07:01.22 | hariom | ok |
| 07:01.34 | ChannelZ | What do you make/receive calls with? |
| 07:02.27 | hariom | I have converted all prompts from wav to gsm. I am using SIP to Mobile and mobile to SIP termination. Mobile means GSM network. |
| 07:03.09 | ChannelZ | HarrisAsterisk: Oh.. perhaps setvar isn't in 1.4... |
| 07:03.29 | HarrisAsterisk | :-( |
| 07:04.19 | ChannelZ | It's in sip.conf but I don't see it in the 1.4 sample configs for DAHDI |
| 07:04.25 | HarrisAsterisk | can i update channel variable from chan_dahdi |
| 07:04.44 | HarrisAsterisk | ok :-( |
| 07:05.37 | HarrisAsterisk | Do you (or anyone) know if there's a way to set EXTEN directly |
| 07:05.38 | ChannelZ | So you're stuck with looking at ${CHANNEL} |
| 07:05.44 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:06.06 | ChannelZ | The extension is read-only but you can Goto to jump someplace else |
| 07:15.30 | Russ | DumpChan tells me the same thing... |
| 07:15.33 | Russ | 'CallerIDNum= (N/A)' |
| 07:15.46 | Russ | er...why N/A, and not, say blank |
| 07:16.10 | ChannelZ | probably just what the source turns nothingness into |
| 07:18.47 | *** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:20.47 | ChannelZ | is this a generic console or is it using alsa or oss? |
| 07:21.49 | Russ | jeez, I can't even set callerid by doing a CALLERID(num)=... |
| 07:21.56 | Russ | alsa |
| 07:22.26 | ChannelZ | I wonder if the callerid string needs to be set in alsa.conf instead? |
| 07:22.37 | Russ | pretty sure its in channels.conf |
| 07:22.41 | Russ | er, console.conf |
| 07:22.55 | ChannelZ | but they kind of mirror each other, is console.conf even used? |
| 07:23.14 | Russ | they don't like very mirror'y to me |
| 07:23.32 | ChannelZ | well insomuch as each has their own context, autoanswer settings, etc. |
| 07:24.03 | ChannelZ | I haven't played with the console much but my recollection was that when loading chan_alsa that chan_console isn't used |
| 07:24.43 | Russ | when loading chan_alsa, chan_oss isn't used |
| 07:25.06 | Russ | Set(CALLERID(num),1__________); |
| 07:25.16 | Russ | Verbose(e.164 dialing ${EXTEN} from ${CALLERID(num)}) |
| 07:25.18 | Russ | wtf |
| 07:25.29 | Russ | you'd think that'd be sufficient |
| 07:25.41 | ChannelZ | Set(CALLERID(num)=12345) |
| 07:26.19 | Russ | well whack me with a hammer |
| 07:26.57 | ChannelZ | Let it be known, I was given permission! |
| 07:32.52 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 07:37.29 | Russ | wonders if there is an application for making it as if the user had dialed a paticular number |
| 07:40.31 | ChannelZ | besides just Goto? |
| 07:40.45 | Russ | I'm trying to do something for the challenge of it |
| 07:40.53 | Russ | I have a _NXX! extension |
| 07:41.04 | Russ | inside that, I want to jump to another extension and continue processing |
| 07:41.18 | Russ | which works fine for all at once dialing, but doesn't for dial as you go |
| 07:41.27 | Russ | hmm |
| 07:41.50 | Russ | actually...I can just test if the length of ${EXTEN} is 3 |
| 07:43.53 | *** join/#asterisk Da-Geek (~Da-Geek@80.235.230.186) |
| 07:54.38 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 07:57.40 | redax | hi |
| 07:58.16 | redax | why don't calls externnotify when new messages arrives? |
| 08:04.00 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:10.01 | *** join/#asterisk troffasky (~r00t@no.ipdetails.info) |
| 08:10.49 | hariom | Hi, I have changed the ASTARGS="" in safe_asterisk to ASTARGS='-U <normal user>' where <normal user> is the user I am currently logged in. But when I see "ps -U root -u root u | grep asterisk" I see that asterisk is running as root. |
| 08:10.53 | hariom | Why is it so? |
| 08:11.41 | hariom | According to Chapter 13 of the book, "ps aux | grep asterisk" should show only the user in which it should run but in my case it is showing root as well as the normal user. |
| 08:14.09 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:15.49 | redax | hariom: runnin `/usr/sbin/safe_asterisk -U asterisk' ? |
| 08:16.02 | redax | or /usr/sbin/safe_asterisk -U asterisk -G asterisk |
| 08:16.12 | redax | if you want group as well |
| 08:16.16 | hariom | only -U |
| 08:16.25 | redax | it works here |
| 08:16.41 | hariom | I made changes in safe_asterisk file. ASTARGS="-U asterisk" |
| 08:17.27 | redax | should be ok as well |
| 08:18.01 | redax | nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} |
| 08:18.03 | hariom | I am on Ubuntu 10.04 |
| 08:18.21 | redax | this how asterisk started... ${CLIARGS} is what you gave to safe_asterisk |
| 08:18.37 | redax | ASTARGS normally empty, except if you specified anything |
| 08:18.54 | hariom | ok |
| 08:19.43 | redax | so, I don't know what's your prob. this way though if you add -U asterisk for safe_asterisk, it shouldn't work as well |
| 08:20.03 | redax | so you have kinda different problem. do you have the asterisk user really? :) |
| 08:21.00 | hariom | No, its not asterisk but sys so I have done something like this: ASTARGS="-U sys" |
| 08:22.22 | redax | is you asterisk startup well, if you do `safe_asterisk -U sys' or even `asterisk -U sys -qvvvvvc' |
| 08:23.08 | hariom | It starts well with asterisk -cvvvv |
| 08:23.18 | hariom | I can see this asterisk process in System monitor |
| 08:23.36 | hariom | asterisk -cvvvv is done without root permission |
| 08:24.10 | hariom | ok. I got problem |
| 08:24.36 | hariom | I was running sudo /etc/init.d/asterisk start; I should run without sudo bcoz now its in normal users |
| 08:25.09 | hariom | redax: Thanks for helping. |
| 08:28.32 | hariom | Ok, there is another problem I am encountering on my another PC. When I install asterisk and do "make install" it gives errors: gzip: stdin: unexpected end of file |
| 08:28.32 | hariom | tar: Unexpected EOF in archive |
| 08:28.57 | hariom | tar: Error is not recoverable: exiting now |
| 08:28.58 | hariom | make[1]: *** [/home/sys1/asterisk-bin/lib/asterisk/sounds/.asterisk-core-sounds-en-wav-1.4.19] Error 2 |
| 08:29.36 | hariom | make: *** [datafiles] Error 2 |
| 08:31.30 | redax | your archive not downloaded correctly. maybe smaller than the real |
| 08:31.53 | redax | uhh. have to go. |
| 08:31.57 | redax | see ya later |
| 08:32.17 | hariom | ok, will try again |
| 08:33.24 | Diffen2 | Hello, sorry to ask this question again but is it possible to have one external number that creates multiple conferencerooms via a pin code? |
| 08:36.35 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:38.01 | Russ | have the extension read the pin with Read() after a suitable background message |
| 08:38.15 | Russ | then you can call the appropriate conference extension with the supplied pin |
| 08:40.07 | Diffen2 | Russ hmm the thing is that there shouldnt be any conference room from the beginning. they should be created when you enter a pin. if there isnt a room with the pin it will create a room. if a second caller enter the same pin they will end up in the same conference. if the second caller enter a different pin, he will start a new confernece |
| 08:40.56 | Russ | I see |
| 08:41.01 | Diffen2 | :) |
| 08:41.08 | Russ | Like I said, do it manually with Background and Read |
| 08:42.22 | Diffen2 | hmm but doent i need to have a bunch of conference rooms created? |
| 08:44.08 | Russ | Just find the next empty one |
| 08:44.15 | Diffen2 | ahh ok |
| 08:44.53 | Diffen2 | i thought i should use the D option of the conference room |
| 08:46.22 | Diffen2 | hmmm i think i will need to do some read regarding the Read() command. Russ thank you for you help. |
| 08:46.49 | Russ | I don't know very much about conference rooms, just that there is an application command with something about a conference room number, and a pin |
| 08:51.29 | Diffen2 | ok |
| 08:51.41 | Diffen2 | i will try your solution out russ. it might work |
| 08:53.09 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
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| 09:04.04 | shamelessn00b | hi all |
| 09:19.53 | Russ | why oh why does the ITU release their information in tables contained in pdf's? |
| 09:26.51 | Russ | beautiful tables of NXX codes, international dialing rules, etc |
| 09:27.56 | shamelessn00b | Hey whats the best way to set up asterisk as a media gateway, IAX, SIP?? |
| 09:28.55 | shamelessn00b | We are 48 E1s terminating at our servers, I want all those E-1s to go on 1 server and then that server would originate calls to other servers based on what service was requested |
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| 09:33.50 | Russ | sigh...ITU |
| 09:33.52 | Russ | "Dialling Procedures" |
| 09:39.18 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 09:42.59 | Russ | ah, that is beautiful |
| 09:43.18 | Russ | they have a word version, and abiword <file> --to=xml does such a wonderful job |
| 09:43.42 | troffasky | careful |
| 09:43.43 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:43.58 | troffasky | if they hear you saying that, the next version will be a GIF pasted into a Word document |
| 09:44.48 | Russ | some xpath magic or maybe xslt and we'll have some dialplans |
| 09:46.13 | Russ | does Chile have phones with Y and Z keys or something? |
| 09:46.27 | Russ | their international prefix is 1YZ0 |
| 09:48.25 | Russ | ah, 1YZ is a carrier code |
| 09:53.29 | shamelessn00b | noobl question, does IVR need echo cancellation |
| 09:54.43 | shamelessn00b | if your just playing back files from * and responding to user DTMFs, would you need echo cancellation? |
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| 11:36.45 | Diffen2 | Hello. Is it possible to use Diversion header only when calls are using followme? we want to show the A-number in the displays of the followme devices. |
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| 11:42.40 | jpmcallister | Im geting this NOTICE in CLI: |
| 11:42.40 | jpmcallister | handle_request_subscribe: Received SIP subscribe for peer without mailbox: 2229 |
| 11:42.47 | jpmcallister | but the peer has a mailbox |
| 11:43.14 | jpmcallister | it is defined in users.conf |
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| 12:33.16 | ypsltd | can anyone give me some pointers on how to use the volume() option on my 1.6 installation? need to set gain on sip channels |
| 12:34.37 | [TK]D-Fender | ypsltd: "core show function VOLUME" <- instructions. |
| 12:35.32 | ypsltd | i see but how do I actually add the gains to my setup i.e. I don't know where to put the lines or what to write in my dialplan |
| 12:36.02 | [TK]D-Fender | ypsltd: the instruction GIVE you a sample, and you put it wherever you need it |
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| 12:36.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:36.59 | ypsltd | ok thanks, so I literally enter the line "Set(VOLUME(TX)=6)" without quotes somewhere in my dialplan? |
| 12:38.03 | [TK]D-Fender | ypsltd: yes |
| 12:39.14 | ypsltd | ok thanks, sorry to be a pain but I tried this and asterisk reload but no joy (in extensions.conf): [from-internal] |
| 12:39.14 | ypsltd | include => from-internal-xfer |
| 12:39.14 | ypsltd | include => bad-number |
| 12:39.14 | ypsltd | exten => s,1,Set(VOLUME(TX)=6) |
| 12:39.26 | [TK]D-Fender | ypsltd: Just FYI, 6db of boost is FOUR times the volume |
| 12:39.35 | *** join/#asterisk mrgabu (~gbdurante@187.38.128.6) |
| 12:39.39 | [TK]D-Fender | ypsltd: and do NOT flood like that again, use a PASTEBIN |
| 12:39.40 | [TK]D-Fender | ~pb |
| 12:39.41 | 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.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. |
| 12:39.46 | ypsltd | ok maybe that's a bit much! am I missing something obvious? |
| 12:39.56 | *** part/#asterisk mrgabu (~gbdurante@187.38.128.6) |
| 12:40.02 | [TK]D-Fender | ypsltd: that line looks fine. is it being CALLED? |
| 12:40.23 | ypsltd | not sure how to check sorry |
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| 12:41.36 | ypsltd | thanks Ijust did again and works a treat! |
| 12:41.44 | [TK]D-Fender | ypsltd: What do you think "s,1" is relevant in that context? |
| 12:44.45 | ypsltd | sorry i was wrong it's not working. I'm afraid I do not understand the dialplan enough to know what to add to my context |
| 12:46.37 | [TK]D-Fender | ypsltd: put it before you call out. Or before anything you need to beard at a different volume. |
| 12:47.09 | ypsltd | ok how do I do that? |
| 12:49.36 | [TK]D-Fender | ypsltd: Sounds like you have absolutely no clue how your own dialplan is set up. And neither do we. |
| 12:49.57 | [TK]D-Fender | ypsltd: If you don't even know where you want it... there is nothing we can do. |
| 12:50.27 | ypsltd | lol that is true, I am using elastix and freepbx. I can pastebin anything you want to see if you like? If I can get this volume altered I will be jumping for joy! |
| 12:51.29 | [TK]D-Fender | ypsltd: No, that confirms it. You are not in control of your dialplan. FreePBX is. |
| 12:51.35 | [TK]D-Fender | ypsltd: And it is NOT suppotred here. |
| 12:52.06 | [TK]D-Fender | ypsltd: You aer going to have to do some odd custom work to hack this into your setup, and it will not be vialable for several places. |
| 12:52.12 | [TK]D-Fender | ~freepbx |
| 12:52.13 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 12:52.15 | [TK]D-Fender | ^^^ |
| 12:53.34 | ypsltd | ok thanks for your advice I will try #freepbx but I have a feeling they will not have a clue. Looks like I may have a lot of readying to do |
| 12:55.33 | [TK]D-Fender | ypsltd: If you have 1 provider you need to pump up you'll need to make a small custom context to receive the call into, and another context for a "custom trunk" which will mod the volume before jumping back to your normal extens it will create for you ... or just dial out the peer itself. |
| 13:09.25 | *** join/#asterisk kalimc (~root@173-203-51-116.static.cloud-ips.com) |
| 13:10.15 | kalimc | What is the best way to go about troubleshooting a faxing issue with Hylafax? I'd prefer something close to a step-by-step elimination |
| 13:11.16 | kalimc | I couldnot track down a Hylafax irc channel, is there such a monster? |
| 13:11.29 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:12.49 | [TK]D-Fender | kalimc: http://www.hylafax.org/content/Support |
| 13:13.33 | kalimc | thank-you kind sir |
| 13:14.34 | *** join/#asterisk kalimc (~root@173-203-51-116.static.cloud-ips.com) |
| 13:16.28 | kalimc | One more thing, is it necesarry to have %100.00 timing tests with zttest? The hylafax manual suggests it, however, I have not been able to see anything over 99.9% |
| 13:17.30 | pabelanger | should be good |
| 13:17.32 | [TK]D-Fender | kalimc: Perhaps you could give some actually useful details about your working environment. |
| 13:17.39 | pabelanger | trying moving to a different slot |
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| 13:21.13 | kalimc | We moved it to a different slot, but will always share with the on-board video card. The environment is Asterisk 1.4.22.1, libpri 1.4.7, |
| 13:21.32 | kalimc | looking up iax version, and hylafax version |
| 13:22.18 | WIMPy | kalimc: Why do you have viedo interrupts enabled? |
| 13:22.49 | kalimc | I didn't see a place to disable video interrupts |
| 13:23.11 | kalimc | Its phoenix bios |
| 13:24.07 | kalimc | I am looking in the bios now |
| 13:25.14 | kalimc | the only obvious setting in the bios is to disable the onboard device, however I will google it now |
| 13:26.03 | pabelanger | Disable anything you are not using, IE: USB, serial / parallel ports. |
| 13:27.01 | kalimc | ok, will do. |
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| 13:31.08 | kalimc | pabelanger: I also disabled the PCI slots not in use, are you familiar with "Enable Master"? |
| 13:31.57 | pabelanger | no |
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| 13:52.27 | hariom | Hi, I am constantly getting this error while doing make install from scratch. I have downloaded complete * twice. tar: Error is not recoverable: exiting now |
| 13:52.45 | hariom | make[1]: *** [/home/sys1/asterisk-bin/lib/asterisk/sounds/.asterisk-core-sounds-en-wav-1.4.19] Error 2 |
| 13:53.16 | hariom | make: *** [datafiles] Error 2 |
| 13:53.28 | pabelanger | ~pb |
| 13:53.29 | 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.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. |
| 13:53.31 | pabelanger | hardwire: ^^ |
| 13:53.47 | pabelanger | hariom: ^^ |
| 13:53.51 | hariom | oh yea, I thought its very small (just 2 lines) so didn't use it |
| 13:54.01 | hariom | I know pb |
| 13:54.15 | pabelanger | Also, why 1.4.19? Have you tried 1.4.34? |
| 13:54.38 | hariom | Its not * 1.4.19. It is sounds-en-wav-1.4.19 |
| 13:54.47 | hariom | I am using 1.6.2.10 |
| 13:56.02 | pabelanger | Ya, sorry. need coffee |
| 13:56.05 | pabelanger | I think it is a bug |
| 13:56.07 | p3nguin_ | "sounds" is separate? |
| 13:56.14 | pabelanger | try 1.6.2.11-RC1 |
| 13:56.26 | pabelanger | hariom: Centos? |
| 13:56.42 | hariom | pabelanger: ubuntu Hardy |
| 13:57.10 | pabelanger | Ya, I'm sure this is a regression bug with 1.6.2.10. Fixed in .11-RC1 |
| 13:57.23 | hariom | I have installed * before and never got this problem. But this even after deleting all directories, I am getting this error. |
| 13:57.46 | hariom | But, the same version has got installed in Ubuntu 10.04 |
| 13:57.50 | pabelanger | Its a regression, introduced in 1.6.2.10 |
| 13:58.09 | pabelanger | Not all OSs are affected |
| 13:58.23 | pabelanger | Only 1 I knew of was Centos |
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| 13:59.14 | pabelanger | hariom: https://issues.asterisk.org/view.php?id=17716 |
| 13:59.19 | pabelanger | https://issues.asterisk.org/view.php?id=17722 |
| 14:02.06 | hariom | thanks |
| 14:02.19 | hariom | so you suggest to upgrade to rc1? |
| 14:02.59 | pabelanger | yes |
| 14:03.31 | pabelanger | If you still have a problem, I would recommend opening a new issue on the tracker |
| 14:04.10 | hariom | How would one update the existing *? |
| 14:04.25 | hariom | I would like to update my ubuntu 10.04 machine as well |
| 14:04.25 | p3nguin_ | by installing a new version. |
| 14:04.37 | hariom | ah, thats a big process again |
| 14:04.39 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:05.04 | p3nguin_ | Not really, especially if you remain within the same branch. |
| 14:05.39 | p3nguin_ | Download the source, compile it, roll it into a package for your distro, install said package. |
| 14:06.43 | kalimc | What is the easiest way to get the HylaFax version? |
| 14:08.17 | hariom | pabelanger, p3nguin: what if I manually install sound packages and dselect sounds in menuselect? |
| 14:08.35 | *** join/#asterisk maxhbp204 (~chatzilla@122.170.39.24) |
| 14:09.11 | pabelanger | Yes, is a option |
| 14:09.38 | p3nguin_ | I've never even heard of this "sounds" thing you're trying to compile. |
| 14:09.39 | hariom | pabelanger: where to get rc1 ? |
| 14:09.58 | hariom | I see only 1.6.2.10 |
| 14:10.16 | pabelanger | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.6.2.11-rc2.tar.gz |
| 14:10.16 | hariom | p3nguin: I am not going to compile. Download the place in the sounds directory |
| 14:10.22 | pabelanger | looks like rc2 was released |
| 14:10.41 | maxhbp204 | Hello, I am from india, I am using sangoma A200 with Airtel Line, dahdi 2.3.0.1 and asterisk-1.6.2.9, i am able to call incoming and outgoing with calls, but in incoming call i am not getting callerid proper |
| 14:11.01 | maxhbp204 | i have set cidstart=polarity_In and cidsignalling=dtmf |
| 14:11.10 | maxhbp204 | after that i am getting 8 digits in callerid |
| 14:11.23 | maxhbp204 | but the full complete callerid of 10 digits is not coming |
| 14:11.29 | maxhbp204 | can any one help me in this please |
| 14:11.42 | pabelanger | Is CallerID enabled on the line? |
| 14:11.47 | maxhbp204 | yes |
| 14:12.02 | maxhbp204 | callerid is coming if i am setting the phone directly |
| 14:12.09 | [TK]D-Fender | maxhbp204: If you plug an analog phone right onto taht line, will it see 10 digits? |
| 14:12.17 | maxhbp204 | but if i am setting it in sangoma A200 then it is issue |
| 14:12.21 | maxhbp204 | yes |
| 14:12.33 | maxhbp204 | i am getting 10 digits |
| 14:12.51 | [TK]D-Fender | maxhbp204: try with "echotraining=no" |
| 14:13.03 | maxhbp204 | ok let me try with it, thanks for your help |
| 14:13.22 | maxhbp204 | and another issue with the card is that it is taking delay for hangup the call |
| 14:13.38 | maxhbp204 | like if i am hanging up the line from mobile then also it is hanging up after 4-5 seconds |
| 14:13.48 | maxhbp204 | it is working fine if i am using analog phone directly |
| 14:14.08 | hariom | It is issue with telecom operators. May be. I read somewhere |
| 14:14.52 | hariom | maxhbp204: In which state are you? |
| 14:15.03 | maxhbp204 | hariom: yes i am also thinking that, but if it is the case then it has to do same in analog too |
| 14:15.12 | pabelanger | maxhbp204: you need to check if your telco provides a disconnect tone, then compare it to your settings in indications.conf |
| 14:15.13 | maxhbp204 | i am using Airtel in ahmedabad |
| 14:15.38 | maxhbp204 | pabelanger: ok that i have to check |
| 14:15.40 | hariom | Is your analog phone line from same operator? |
| 14:15.43 | maxhbp204 | thanks for that help |
| 14:15.46 | pabelanger | Most don't provide a disconnect tone for analog. |
| 14:15.59 | maxhbp204 | hariom: yes |
| 14:16.03 | hariom | hmm.... |
| 14:16.48 | hariom | Is you line a pri ? |
| 14:16.51 | kalimc | Looking at my zaptel.conf file, I see me settings (span=1,0,2,esf,b8zs) - and referencing them to the documentation, wouldn't one want to select '1' to use the circuits timing? |
| 14:17.02 | maxhbp204 | [TK]D-Fender: i have to add this echotraining=no in chan_dahdi.conf? |
| 14:17.10 | maxhbp204 | it is not pri line |
| 14:18.10 | maxhbp204 | can i give chan_dahdi.conf configuration in pastebin? |
| 14:18.23 | [TK]D-Fender | maxhbp204: Correct. it si a channel parameter |
| 14:18.52 | maxhbp204 | ok |
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| 14:20.58 | maxhbp204 | http://pastebin.com/CpwUWsN4 |
| 14:21.03 | maxhbp204 | this is chan_dahdi.conf |
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| 14:25.09 | maxhbp204 | Any suggestions? |
| 14:28.50 | [TK]D-Fender | maxhbp204: I do not see you doing what I suggested |
| 14:31.02 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:32.40 | maxhbp204 | yes that i am doing test right now |
| 14:32.46 | maxhbp204 | thanks |
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| 14:36.46 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:38.36 | packetwalk | what's the best place to learn asterisk? |
| 14:40.57 | russellb | ~thebook |
| 14:40.58 | infobot | from memory, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 14:41.55 | ZeXr0 | That and trial and error |
| 14:42.24 | ZeXr0 | I guess that once you get the "difficult" part to work (connection to an SIP provider) making a callflow is really really easy |
| 14:43.14 | packetwalk | I'll start with the .pdf |
| 14:43.26 | packetwalk | thanks people |
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| 14:47.38 | Mango | Hi. I was hoping someone could give me some advice about setting up IAX. When I attempt to place a call, I see this in the CLI: |
| 14:47.39 | Mango | http://pastebin.ca/1912298 |
| 14:47.56 | Mango | ...and that's it; Zoiper eventually says "recovery on timer expire" |
| 14:48.36 | kalimc | Wow! Disabled all un-used items (usb/slots/misc devices), made an adjustment to zapta.conf, and I am not seeing any timings from zttest below 99.999 |
| 14:50.28 | Mango | It looks like Asterisk isn't responding to Zoiper, but I have no idea why :) |
| 14:51.13 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:21a:73ff:fe3f:1711) |
| 14:52.59 | kalimc | Forgive me for asking, but how important is it that the 'distance parameter' in the span setting (param #3) be set correctly? Would one see a difference if it were set to say '0' (0-133 feet) if the actual distance was say 500 feet? What is the impact of an incorrect setting? |
| 14:55.41 | pabelanger | kalimc: possible line quality issues |
| 14:56.52 | kalimc | thats what I expected, have there everr been instances where someone had is set incorrectly and saw an issue? |
| 14:58.07 | maxhbp204 | [TK]D-Fender: after applying that ,i have reload asterisk and also dahdi restart in asterisk too, but i am getting 8 digits now |
| 14:59.26 | maxhbp204 | can i have to add in span too? or in general section? |
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| 15:01.00 | maxhbp204 | http://pastebin.com/6nDw4YZ2 |
| 15:01.04 | maxhbp204 | please see it |
| 15:04.54 | pabelanger | kalimc: Nothing asterisk specific, but in other equipment I have. |
| 15:05.23 | kalimc | pabelanger: ok great. I'll measure and set it accordingly. |
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| 15:25.31 | kalimc | I see 4 spans in my settings, they all are using the timing setting of '1' (2nd parameter). Should subsequent spans after the first be set to use the 1'st span, or should they all match the span ID. i.e: 1,1,0 // 2,2,0 // 3,3,0. |
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| 15:26.17 | kalimc | they are all current setup as 1,1,0 // 2,1,0 // 3,1,0 // 4,1,0 - which I suppose makes sense, as there looks to be only 3 settings for that parameter |
| 15:28.40 | _zoom_ | hi, i need to enable h323 using ooh323, need help? |
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| 15:52.03 | mpdavis73 | i have what may be a below basics question - who provides and handles the telephone numbers? the ISP? would it have to be the ISP by definition? we already have IP phones |
| 15:52.36 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:53.19 | Naikrovek | the voice provider |
| 15:53.31 | Naikrovek | telephone companies handle number assignments when it comes to phones |
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| 15:59.07 | cfh | hi all, what can i do to show from the * cli the codec used in a bridge call ? |
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| 15:59.25 | mpdavis73 | i think i don't know enough yet to know what questions to ask - back to reading |
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| 16:09.48 | wcselby | o/ |
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| 16:47.53 | mpdavis73 | is there a different type of T1 as opposed to an internet T1 or is it just how it is used? (Reading chapter 7) |
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| 16:50.11 | kalimc | if 'pri' is not available from the asterisk prompt, however, in zttool, I am seeing everthing as ok, is there a way to reload the module? |
| 16:50.30 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 16:53.09 | kerframil | kalimc: module reload chan_dahdi |
| 16:53.46 | kerframil | kalimc: or zaptel, as the case may be |
| 16:54.06 | kalimc | thanks, I will try that |
| 16:55.53 | pabelanger | mykhyggz: Yes, same physical connection, different protocols that run atop. |
| 16:56.09 | kalimc | 'ERROR[5667]: chan_zap.c:10527 build_channels: Unable to reconfigure channel '1-23' |
| 16:56.50 | kalimc | strange that it will not reload as zttool shows a green light, zttest is looking ok, 99.999.. |
| 16:57.17 | kalimc | I cannot figure out why I am unable to load the module |
| 16:58.15 | kalimc | green light, card is listed in lspci, zttool = ok, zttest = ok. What should I look into next? |
| 17:00.56 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.168.54.dsl.dyn.forthnet.gr) |
| 17:08.40 | kerframil | kalimc: the device needs to be provisioned by zaptel_cfg before asterisk/chan_zap initialises. I'd suggest running zaptel_cfg -vvvv and looking for anything that may be amiss. |
| 17:09.26 | kerframil | kalimc: on a related note, you might consider moving to dahdi at some point as zaptel is an end-of-life implementation. |
| 17:11.50 | mpdavis73 | what would need to be done to change the VoIP protocol after implementation? |
| 17:12.56 | drmessano | protocol? |
| 17:13.05 | drmessano | Change it where? |
| 17:14.00 | mpdavis73 | still in early learning stage - reading chapter 8, GSM vs iLBC, etc |
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| 17:15.35 | drmessano | Those are codecs, not protocols |
| 17:16.16 | mpdavis73 | duh - was looking at the title chapter, not the section chapter |
| 17:16.28 | drmessano | Codecs are relatively trivial to change.. either just in Asterisk or in Asterisk and on the phone, depending on if the phone and Asterisk both freely negotiate a list of codecs |
| 17:16.29 | mpdavis73 | sorry |
| 17:17.13 | drmessano | So if Asterisk is forcing one with a single allow= for that peer, and the phone will auto negotiate it, you simply change in Asterisk and reload |
| 17:17.34 | mpdavis73 | ok - i am hoping our existing phones will work, we have Cisco 7961's |
| 17:17.52 | drmessano | Are you going to use SIP or Skinny? |
| 17:18.04 | kerframil | mpdavis73: Cisco's should support alaw, ulaw, gsm and g729 at least. mind you, there may be more restrictions if using the SIP firmware. |
| 17:18.15 | mpdavis73 | don't know enough to know yet |
| 17:18.58 | mpdavis73 | right now we are using a fully hosted network+tele |
| 17:20.23 | mpdavis73 | most likely SIP, i don't like the word 'proprietary' |
| 17:20.52 | mpdavis73 | depends on what i have to do to the phones, about half of our users are at remote offices |
| 17:21.06 | kerframil | mpdavis73: maybe so, but I've heard that they're not great as SIP devices. that's based on hearsay though as I stick to the SCCP firmware on my Cisco devices and have never tried using SIP with them. |
| 17:21.24 | drmessano | You don't like the word proprietary, but you're using Cisco? |
| 17:21.31 | drmessano | Are you MAD, man? |
| 17:21.37 | mpdavis73 | wasn't my decision |
| 17:21.45 | fenrus | does sccp work good in asterisk? |
| 17:21.48 | drmessano | Who made the decision? |
| 17:21.51 | fenrus | i've heard of intermittent crashes etc |
| 17:21.54 | mpdavis73 | my it director |
| 17:22.02 | drmessano | mpdavis73: Can you fire him? |
| 17:22.11 | mpdavis73 | he's my boss |
| 17:22.12 | kerframil | mpdavis73: all sources I've referred to suggest that features are lost by the SIP firmware and, worse, if you flash the SIP firmware I don't think there's any going back. on the other hand, chan_skinny is a bit weak. the best sccp drive in my opinion is the chan-sccp project (the fork managed by Federico Santulli) but it's third party so ... |
| 17:22.13 | mpdavis73 | lol |
| 17:22.21 | kerframil | driver, sorry |
| 17:22.39 | drmessano | mpdavis73: Get promoted above him, then fire him.. IT guys who pick Cisco because they make "really cool switches" should be shot, then hung, then shot again |
| 17:22.57 | mpdavis73 | we're paying about $20k a month for smoothstone |
| 17:23.02 | drmessano | O.o |
| 17:23.05 | fenrus | the cisco cli is neat. |
| 17:23.24 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 17:23.31 | mpdavis73 | he's on the exec team |
| 17:23.43 | mpdavis73 | most likely ceo in a year or two |
| 17:24.03 | kerframil | drmessano: that seems a tad excessive. I use Cisco phones and they work. moreover, he may be in a situation where there is a installed base of Cisco devices (for instance, when migrating from Call Manager or whatnot). |
| 17:24.25 | kerframil | drmessano: that said, I wouldn't recommend them to anyone starting out who isn't afraid to get their hands dirty |
| 17:24.43 | kerframil | is afraid, rather |
| 17:25.08 | mpdavis73 | we have gobs of 7961's, a completely managed network and tele, and our current ceo is chomping at the bit for the contract to be up |
| 17:26.08 | mpdavis73 | i have the advantage of having several OSS projects implemented |
| 17:26.09 | drmessano | Cisco phones work well with Cisco telephony solutions.. Their SIP implementation sucks, and while there's some OK implementations of SCCP out there for Asterisk, Cisco really isn't interested in you using their phones on other platforms.. If they were that interested, they wouldnt just toss a shitty SIP implementation out there to look less evil.. |
| 17:26.14 | mpdavis73 | so accounting loves me |
| 17:27.00 | kerframil | drmessano: oh, I agree entirely. I took the risk of sticking to SCCP and it's worked out. hence, the getting hands dirty disclaimer. it can be done though. |
| 17:27.16 | fenrus | i wonder what problems people have with SIP and cisco phones |
| 17:27.29 | fenrus | have a small office with about 10 of em', works as a charm |
| 17:27.30 | drmessano | kerframil: Even when you do it, it's still SCCP.. and it's not very stable. |
| 17:27.51 | drmessano | Which is why I avoid like the plague |
| 17:28.26 | [TK]D-Fender | avoids cliches like the plague... |
| 17:28.46 | kerframil | drmessano: they are rock solid here. but, as I conceded earlier, the only stable sccp driver is that provided by Federico (imo). I can well appreciate that a project operated by one person is not necessarily a choice that would be relished by the majority. |
| 17:29.01 | kerframil | drmessano: to each their own, and all that (oops, a cliche) |
| 17:29.02 | drmessano | I'm sure you COULD make some nice older H.323 devices from Polycom work with Asterisk.. or just use newer, more stable devices on a well tested, supported protocol in Asterisk |
| 17:29.08 | mpdavis73 | i have spare phones I could test with, i was planning on getting a test env at home |
| 17:30.13 | mpdavis73 | my problem is going to be, i think, the fact that our network and tele is through the same company |
| 17:31.15 | mpdavis73 | ALL of our networking equip is Cisco |
| 17:31.41 | mpdavis73 | i think we had one dell switch... |
| 17:33.56 | Katty | ALLO DUKE |
| 17:33.59 | Katty | how's ye ole chap eh? |
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| 17:49.13 | mpdavis73 | i think i need a beer before i read anymore |
| 17:49.42 | Katty | BOTTOMS UP |
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| 17:51.36 | mpdavis73 | general opinion - would the 7961 phones we have now be specific to our provider, or just the cisco firmware and our provider is using ccm? |
| 17:52.16 | Kobaz | bottoms up, tops down? |
| 17:53.57 | kerframil | mpdavis73: the latter, I should imagine. you may determine the firmware version from the phone's interface. |
| 17:54.29 | mpdavis73 | just found it - SCCP41.8-3-4SR1S |
| 17:54.58 | mpdavis73 | has a listing for expansion modules..hmm |
| 17:56.03 | kerframil | mpdavis73: that would be for devices such as the 7915 |
| 17:56.46 | mpdavis73 | ah, ok |
| 18:00.30 | mpdavis73 | G.722? |
| 18:00.44 | mpdavis73 | just a different codec? |
| 18:00.56 | kerframil | mpdavis73: yes, a different codec. and not one that is supported by these phones, iirc. |
| 18:01.19 | pabelanger | mpdavis73: Do you have to use SCCP? You should convert to SIP if possible |
| 18:01.42 | mpdavis73 | under device setting->media config->'Device Advertise G.722' Enabled |
| 18:02.00 | kerframil | mpdavis73: ah, ok. I'm probably out of date. |
| 18:02.31 | mpdavis73 | not sure, i need to be able to use our existing hardware, Cisco IP 7961 |
| 18:02.55 | mpdavis73 | still in early learning phase right now |
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| 18:04.00 | mpdavis73 | i'll probably end up taking a few phones home and signing up with a free sip provider for some testing |
| 18:04.11 | mpdavis73 | so i can figure out what would work for us |
| 18:05.05 | kerframil | mpdavis73: well, it's like this. if you flash the sip firmware, you'll be treading a more broadly supported path. however, the sip implementation in Cisco phones is weak and, as far as I know, you can't go back to sccp thereafter. or you could use the chan-sccp project I mentioned earlier. the functionality is superior (compares very favourably with ccm itself) but it's a one-man band (the chan-sccp code is not maintained within the scope of t |
| 18:05.31 | kerframil | mpdavis73: or you could use devices that do SIP better, obviously |
| 18:06.27 | mpdavis73 | but would require new hardware |
| 18:06.31 | kerframil | mpdavis73: right |
| 18:06.57 | RypPn | you can change from sip to sccp and vice versa |
| 18:07.11 | mpdavis73 | i'll have to look up some prices - i know we are paying $20k/month right now |
| 18:07.19 | kerframil | RypPn: regrettably, no. I think going from sccp -> sip is a one-way street, last I checked. |
| 18:07.28 | mpdavis73 | but that is for internet and tele |
| 18:07.53 | RypPn | I've done it on a 7940 on numerous occasions |
| 18:08.12 | RypPn | while using freeswitch and back to * |
| 18:08.14 | kerframil | RypPn: oh - how? |
| 18:08.34 | RypPn | different tftpboot structure for sip |
| 18:08.39 | mpdavis73 | i have gobs of phones due to downsizing, so a few bricked phones wouldn't be a big deal anyway |
| 18:08.52 | kerframil | RypPn: I was under the impression that SIP firmware will only look for newer instances of SIP firmware with respect to tftp provisioning but would love to be corrected if I'm mistaken |
| 18:09.35 | RypPn | no, it polls for the xml files for both protocols if you watch the RRQ's |
| 18:10.06 | RypPn | SIP and SEP |
| 18:10.29 | kerframil | RypPn: sounds good; I'll check it out |
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| 18:12.25 | mpdavis73 | RypPn: as far as basic operation, what would be different on the phone display? |
| 18:13.12 | RypPn | mpdavis73 under which protocol? Or preseneted against asterisk as opposed to CCM? |
| 18:14.17 | kerframil | mpdavis73: it depends on how it would be driven in the non-sip case. for example, chan_skinny is not feature rich. some branches of chan_sccp support advanced features such as native hardware conferencing and the SRST protocol. |
| 18:14.33 | mpdavis73 | I would probably switch to SIP for asterisk, but if the end-user experience is much different i would probably stay with sccp until i could replace devices |
| 18:14.50 | RypPn | The closest to a ccm presentation is under sccp-b as kerframil states, V2 is solid, V3 with the cool stuff like shared lines, a bit flakey |
| 18:15.18 | RypPn | sip, I'd get other phones tbh |
| 18:15.43 | RypPn | you know the new 9951 and 9971's are sip-only? |
| 18:15.47 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:15.57 | RypPn | they look very hot too |
| 18:16.01 | mpdavis73 | cisco is ditching their own stuff? |
| 18:16.33 | kerframil | RypPn: they do (I doubt the price tag is quite so thrilling though ;) |
| 18:17.52 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:18.27 | RypPn | oh yeah, I don't think they're available quite yet either, at least not here |
| 18:18.56 | RypPn | mpdavis73 since ccm 6 they've been building in a lot more sip functionality, particularly for endpoints |
| 18:21.14 | mpdavis73 | the only one in the company who does more than a 3-way conference call is our swicthboard op, who uses the software piece from smoothstone (probanly rebranded cisco) |
| 18:21.25 | kerframil | RypPn: btw, if you have any interest in V2 for your own purposes this is the place to go now: svn://svn.chan-sccp.org/trunk |
| 18:21.40 | kerframil | RypPn: Federico wanted sole control over his branch so he took his ball elsewhere |
| 18:22.05 | mpdavis73 | so chan-skinny and chan-sccp are branched from the same code? |
| 18:22.15 | RypPn | yeah, none of the code is very good atmo, line presence has been shot for quite a while |
| 18:22.43 | RypPn | still running an old v983 from the first svn repo |
| 18:23.13 | mpdavis73 | for basic calls would a sip flash on the 7961 be solid? |
| 18:23.31 | RypPn | do a taste test |
| 18:23.35 | RypPn | run 2 |
| 18:23.45 | [TK]D-Fender | mpdavis73: Yes |
| 18:23.45 | kerframil | RypPn: the original chan_sccp project was started by Sergio Chersovani. I'm not sure what relationship, if any, it had to chan_skinny; I suspect it was based on it. |
| 18:24.14 | RypPn | iirc the sccp-b code comes from that berlios one for *1.2 and 1.4 |
| 18:24.33 | mpdavis73 | k, ill take another phone home and install in a vm |
| 18:25.10 | mpdavis73 | anyone running on ubuntu? i can use either, but prefer synaptic over rpm |
| 18:25.24 | mpdavis73 | our email server is on centos |
| 18:25.26 | pabelanger | yes |
| 18:25.30 | RypPn | gentoo here |
| 18:25.47 | RypPn | and kerframil too unless I'm way off |
| 18:25.53 | kerframil | RypPn: ah, you're the chap from voipcoop - sorry, I forgot! |
| 18:26.06 | kerframil | RypPn: bit groggy today |
| 18:26.11 | RypPn | onceuponatime, nps |
| 18:26.19 | kerframil | RypPn: oh, right |
| 18:26.21 | RypPn | when I worked from home and had a life |
| 18:26.31 | mpdavis73 | what's a life? |
| 18:26.35 | [TK]D-Fender | Katty: http://blogs.discovery.com/animal_oddities/2009/12/foxy-fun.html |
| 18:26.41 | RypPn | I knew once |
| 18:27.00 | mpdavis73 | looks like i have another weekend project |
| 18:27.03 | mpdavis73 | :) |
| 18:27.13 | RypPn | weekend o.O |
| 18:27.15 | mpdavis73 | for maybe a few weekends |
| 18:27.22 | RypPn | gotta luv that optimism |
| 18:27.26 | RypPn | lol |
| 18:27.49 | mpdavis73 | my other is a php reporting script for our requesttracker implementation |
| 18:27.58 | mpdavis73 | mysql |
| 18:29.01 | mpdavis73 | my end goal is no more winderz on our network |
| 18:29.12 | *** join/#asterisk Shaaan (~Un1x@CPE000024cccb7c-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 18:30.52 | mpdavis73 | i saw reference to fax for asterisk - does that include incoming faxes being delivered to email as pdf attachments? |
| 18:31.18 | [TK]D-Fender | mpdavis73: No. that backend stuff can be done by "anything" |
| 18:32.16 | mpdavis73 | well, i could script the email piece, email i'm better with |
| 18:32.57 | mpdavis73 | i would have to set up something to mimic what we have currently |
| 18:34.48 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 18:35.14 | kerframil | mpdavis73: digium offer a free license for Fax for Asterisk to any user (single concurrency only as opposed to their commercial offerings). it can also be done va iaxmodem/hylafax and probably numerous other methods I'm not familiar with. |
| 18:36.16 | mpdavis73 | i wouldn't be opposed to spending money, it's just we are getting bilked by our current provider |
| 18:36.19 | Katty | what kind of chese do you guys pair spinach with? raw spinach for salad. |
| 18:36.34 | mpdavis73 | the service is outstanding, just expensive |
| 18:36.45 | mpdavis73 | feta |
| 18:36.47 | kerframil | Katty: apropos of nothing, but I would say feta |
| 18:37.05 | Katty | it's been awhile since i've had feta. is that relatively sweet and creamy? |
| 18:37.27 | mpdavis73 | salty and crumbly |
| 18:37.32 | Katty | k |
| 18:40.24 | mpdavis73 | any suggestions as far as brand of sip phone? |
| 18:40.46 | Kobaz | god damn grandstream |
| 18:40.54 | WIMPy | Snom |
| 18:40.55 | Kobaz | this is the third power supply that's fried |
| 18:41.12 | Kobaz | mpdavis73: polycom, linksys, snom (stay away from grandsuck) |
| 18:41.29 | Kobaz | aastra is good too |
| 18:42.11 | [TK]D-Fender | Polycom >All |
| 18:42.26 | Nugget | indeed |
| 18:42.39 | Kobaz | anyone get any of the new avaya sip phones going with asterisk? |
| 18:42.41 | Nugget | adds Cisco to the avoid list. |
| 18:42.53 | Kobaz | i don't know if it's actual sip or some propriatary mix |
| 18:43.07 | Kobaz | Nugget: i havent used cisco phones, but the linksys ones are nice |
| 18:43.16 | WIMPy | Kobaz: Only an old one. Seems to be ok. |
| 18:43.41 | mpdavis73 | heh - cisco is what we have now - 7961G |
| 18:43.42 | KavanS | mpdavis73, check out hylafax/iaxmodem for fax to email/pdf |
| 18:43.44 | Nugget | Cisco IP phones are really only viable in a callmanager environment. Trying to keep them working with Asterisk is fraught with frustration. |
| 18:43.51 | KavanS | mpdavis73, works well on our asterisk system |
| 18:44.04 | mpdavis73 | k, thanks |
| 18:44.08 | kerframil | Katty: out of curiosity, do you use hylafax or hyalfax+? |
| 18:44.09 | Nugget | unpredictable, difficult to purchase, completely undocumented, and weird. |
| 18:44.15 | Katty | kerframil: no. |
| 18:44.18 | Katty | hugs on Nugget |
| 18:44.18 | kerframil | Katty: oops, sorry ... redirecto to KavanS |
| 18:44.22 | Nugget | huggles Katty |
| 18:44.24 | Kobaz | Nugget: sounds like cisco |
| 18:44.32 | Kobaz | Nugget: although they do make nice routers and switches |
| 18:44.35 | WIMPy | Linksys look nice, but that old Sipura software is far from great. |
| 18:45.25 | mpdavis73 | i hate cisco pix - complete crap |
| 18:45.46 | mpdavis73 | all of our stuff is cisco though |
| 18:48.14 | *** join/#asterisk hfb (~hfb@pool-98-112-109-237.lsanca.dsl-w.verizon.net) |
| 18:50.09 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 18:55.34 | *** join/#asterisk bsaxon_ (~bsaxon@12.107.149.61) |
| 18:55.45 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 18:57.35 | b14ck | Hey all, is it possible to do dynamic file inclusion? Or glob matching inclusion? EG: #include /path/to/files/*.conf ? |
| 18:57.48 | b14ck | Is there some syntax for that? Or must I hard-code all inclusion files? |
| 18:59.31 | pabelanger | #include <filename> only |
| 18:59.39 | pabelanger | as far as I know |
| 18:59.41 | b14ck | You sure about that? |
| 18:59.58 | pabelanger | not 100%, have you tried the wildcard? |
| 19:00.13 | b14ck | I have, it doesn't work, but I think there is another syntax for it. |
| 19:00.17 | b14ck | I could look at the C code I suppose. |
| 19:00.21 | b14ck | Actually, I'll just do that. |
| 19:00.51 | pabelanger | did your quote your statement? |
| 19:00.58 | pabelanger | s/your/you |
| 19:01.03 | b14ck | Quotes are broken in the latest asterisk release. |
| 19:01.12 | b14ck | There's a parsing bug which prevents you from using them around filenames. |
| 19:01.23 | b14ck | It says it is fixed in the SVN, but it isn't from what I have tested |
| 19:01.25 | pabelanger | issue number? |
| 19:01.31 | b14ck | lemme find it |
| 19:01.57 | Naikrovek | that was includes only i think |
| 19:01.58 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 19:02.01 | b14ck | Yah, includes. |
| 19:02.17 | kerframil | b14ck: ael can |
| 19:02.24 | b14ck | Oh yah? |
| 19:02.31 | b14ck | I haven't messed with AEL, is it worth learning? |
| 19:02.38 | kerframil | b14ck: yeah - not sure about extensions.conf |
| 19:02.44 | b14ck | I wasn't sure if it was a temporary thing, or if it was going to be EOL'ed. |
| 19:02.49 | b14ck | Seems like most people don't use / know about it. |
| 19:03.44 | pabelanger | only recent issue I know of: https://issues.asterisk.org/view.php?id=17472 |
| 19:03.58 | b14ck | pabelanger, that's the one! |
| 19:04.02 | b14ck | I was googling it, but couldn't find it. |
| 19:04.14 | b14ck | It still is broken in SVN though (as of a few days ago) |
| 19:04.21 | pabelanger | should be fixed, if not reopen the issue |
| 19:04.31 | b14ck | this sounds kinda lame |
| 19:04.37 | b14ck | but ive never actually used the asterisk issue tracker |
| 19:04.39 | b14ck | guess i should |
| 19:05.18 | bmoraca_work | what a freakin weird issue...dropping the 7th character and doubling the 16th? |
| 19:07.19 | b14ck | yo, sup bmoraca_work :) |
| 19:07.26 | bmoraca_work | not much, you? |
| 19:07.45 | b14ck | Not much, trying to get dynamic file inclusion working. |
| 19:08.03 | b14ck | Our asterisk platform is huge now, so I'm modularizing it. |
| 19:08.18 | bmoraca_work | nice |
| 19:08.37 | bmoraca_work | i'm building a residential voice billing platform (yuk) |
| 19:08.43 | b14ck | heh |
| 19:08.55 | b14ck | We actually started a SIP company kinda thing too (for sellig to VoIP providers). |
| 19:08.58 | b14ck | But it isn't done yet. |
| 19:08.59 | [TK]D-Fender | AEL is a shell layer that can introduce issues and does nothing that basic dialplan can't, and can't do everything you can do that way either. |
| 19:09.24 | bmoraca_work | yeah, i suggested to my boss we could sell some of our off-peak capacity as residential service...he said "go for it"...so...yeah |
| 19:09.25 | b14ck | [TK]D-Fender, thanks for the perspective. I wasn't sure how it was viewed. |
| 19:09.40 | b14ck | [TK]D-Fender, do you know how to include files dynamically with dialplan by any chance? |
| 19:09.46 | [TK]D-Fender | Looks pretty adds no actual value. Slower to process, harder to debug, and just one more thing that can break |
| 19:09.53 | b14ck | EG: I want to include all *.conf files in some directory, without hardcoding their names. |
| 19:10.13 | b14ck | I'm looking at main/config.c right now |
| 19:10.23 | bmoraca_work | b14ck: realtime extensions might be useful in that case, if your extensions are not pattern-based |
| 19:10.42 | b14ck | I'm trying to avoid realtime (will add another layer of complexity to my setup) |
| 19:11.05 | b14ck | My extensions are pattern based, but what I'm trying to do is create pluggable modules: |
| 19:11.06 | b14ck | EG: |
| 19:11.14 | b14ck | My extensions.conf has somethign like |
| 19:11.19 | bmoraca_work | for our main asterisk-based voice switch, I looked at doing file-based settings, but a combination of FUNC_ODBC and realtime allowed me to do everything I needed without worrying about tons of files...full dialplan is like 120 lines long |
| 19:11.20 | b14ck | #include modules/*.conf |
| 19:11.28 | b14ck | That way, any module I write can be dynamically included. |
| 19:11.39 | b14ck | We make extensive use of modules for features and stuff. |
| 19:11.44 | bmoraca_work | "dialplan reload" isn't really so bad :P |
| 19:11.56 | b14ck | Oh yah, I don't care about reloading. |
| 19:12.05 | b14ck | But I just want to not have to hardcode each file in. |
| 19:12.12 | [TK]D-Fender | b14ck: And have you tried with & without quotes? |
| 19:12.22 | bmoraca_work | how many files are we talking about? |
| 19:12.25 | b14ck | [TK]D-Fender, with the glob? (*) |
| 19:12.31 | b14ck | bmoraca_work, hundreds |
| 19:12.38 | [TK]D-Fender | b14ck: Clearly |
| 19:12.40 | bmoraca_work | really? |
| 19:12.49 | b14ck | [TK]D-Fender, no actually, let me test it out. |
| 19:13.09 | b14ck | [TK]D-Fender, yah, its a complex setup. We have hundreds of services which are all self-contained. |
| 19:13.31 | bmoraca_work | curious |
| 19:15.13 | b14ck | Oh shit. |
| 19:15.22 | b14ck | [TK]D-Fender, I'm a noob. |
| 19:15.30 | b14ck | It does work without the "",s I don't know why I didn't try that before. |
| 19:15.35 | b14ck | Even though I was aware of the " bug. |
| 19:15.36 | b14ck | sigh |
| 19:15.39 | Naikrovek | newb mistake != newb |
| 19:16.06 | *** join/#asterisk philipp64|laptop (~chatzilla@75-92-150-245.war.clearwire-wmx.net) |
| 19:16.09 | b14ck | Nah, I am a noob =) |
| 19:16.19 | Naikrovek | whatever |
| 19:16.27 | b14ck | =p |
| 19:16.32 | b14ck | What are you up to, Naikrovek? |
| 19:16.35 | b14ck | Haven't talked to you in a while. |
| 19:16.48 | Naikrovek | not much |
| 19:17.01 | Naikrovek | if you're a newb, wtf am i |
| 19:17.09 | Naikrovek | because i'm way below you in terms of skill |
| 19:17.12 | b14ck | You probably know more about * than I do. |
| 19:17.17 | b14ck | No, I doubt that. |
| 19:17.22 | b14ck | I am out of practice. |
| 19:17.23 | b14ck | heh |
| 19:17.36 | bmoraca_work | Naikrovek = permanewb |
| 19:17.37 | bmoraca_work | :P |
| 19:18.08 | Naikrovek | ask drmessano; he'll tell anyone who will listen just how much of a jackass i am |
| 19:18.10 | bmoraca_work | hey, Naikrovek, did you ever get your network thing figured out? |
| 19:18.25 | Naikrovek | bmoraca_work: oh the router thing. |
| 19:18.30 | Naikrovek | figured out, yes, i think so |
| 19:18.33 | Naikrovek | implemented, no |
| 19:19.22 | leifmadsen | being a newb is way better than being a nub |
| 19:19.35 | leifmadsen | Qwell: good thing it's been documented you're not a newb |
| 19:20.04 | Qwell | correct |
| 19:20.22 | bmoraca_work | Naikrovek: why not implemented? it's like 4 lines of config! |
| 19:20.27 | Naikrovek | time |
| 19:20.38 | Naikrovek | will have to do it on a weekend, and my weekends are booked |
| 19:20.58 | Naikrovek | boss won't pay for someone to come help so i simulate on GNS3 when I have time for that |
| 19:21.28 | Naikrovek | so i gotta make sure all works before i implement, because when the power goes out on the 1721, there's no going back |
| 19:21.53 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:22.16 | Naikrovek | believe me i want it done |
| 19:22.35 | Naikrovek | but between moving, child visitation, and wife in the hospital my time is spread thin |
| 19:22.58 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 19:23.46 | bmoraca_work | ahhh, sucks |
| 19:23.54 | bmoraca_work | i just moved last week... |
| 19:26.13 | Naikrovek | yeah moving sucks |
| 19:26.15 | bmoraca_work | we looked at the house on monday, got the keys on wednesday, and were moved in by Saturday...all so we could avoid paying another month of rent on our apartment (piece of shit with crappy management) |
| 19:26.36 | Naikrovek | my landlord won't give us our deposit back from teh previous place, either |
| 19:26.50 | bmoraca_work | why not? |
| 19:26.57 | Naikrovek | saw him last night at walmart, he pulled out the envelope we gave him way back when, and he was buying a TV with it |
| 19:27.03 | Naikrovek | because this is how this man works |
| 19:27.19 | Naikrovek | we took before/after pictures, we improved upon the house |
| 19:27.33 | Naikrovek | no stains no drilled holes, no scuffs in the carpet or tile nothing |
| 19:27.34 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 19:27.43 | Naikrovek | $1600 gone |
| 19:27.48 | bmoraca_work | in california, if they don't get it back to you in 30 days, they forfeit any right to keep any of it for repairs and have to give it back in full |
| 19:28.12 | Naikrovek | in illinois, they get to keep it because it costs more to hire a lawyer and go to court over it |
| 19:28.24 | bmoraca_work | small claims court |
| 19:28.32 | Naikrovek | wife says that |
| 19:28.33 | bmoraca_work | no need for a lawyer |
| 19:28.36 | Naikrovek | but the mayor is his brother |
| 19:28.43 | Naikrovek | and the judge is a high school friend |
| 19:28.46 | Naikrovek | so i'm fucked |
| 19:28.57 | bmoraca_work | that's pretty fucked up |
| 19:29.01 | Naikrovek | yeah |
| 19:29.04 | bmoraca_work | file a complaint with the state renter's board |
| 19:29.05 | mpdavis73 | can you change venue for small claims? |
| 19:29.26 | Naikrovek | dude there is no way his high school friend will let that happen |
| 19:30.01 | mpdavis73 | if you try and fail, then you can go to state level |
| 19:30.02 | RypPn | kerframil the svn repo you referred to, do hints work for you? |
| 19:30.04 | Naikrovek | why the hell is firefox not following 403's now |
| 19:30.07 | Naikrovek | um 302s i mean |
| 19:30.28 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:30.29 | mpdavis73 | firefox 3.6 hosed my VMware web UI |
| 19:30.41 | kerframil | RypPn: no. I'm still trying to get that sorted out. allegedly, the new behaviour is more CCM-like. check out revision 10 - that should work. |
| 19:31.03 | RypPn | thnx |
| 19:31.42 | mpdavis73 | whats the diff between asterisk and asterisknow? |
| 19:31.59 | bmoraca_work | asterisknow is built on top of asterisk |
| 19:32.13 | Naikrovek | asterisk is a program |
| 19:32.19 | Naikrovek | asterisknow is a linux distribution |
| 19:32.24 | mpdavis73 | ah, ok |
| 19:32.32 | Naikrovek | asterisknow = centos + asterisk + freepbx |
| 19:32.35 | NuclearLucifer | Based on Centos. Good distro. |
| 19:32.39 | [TK]D-Fender | AsteriskNOW is a DISTRO that includes Asterisk, a few GUI tools, and a few other odds & ends. |
| 19:32.45 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:32.59 | mpdavis73 | our spam/email is based on centos - ESVA |
| 19:33.18 | NuclearLucifer | [TK]D-Fender, Baisycly it's Centos with centos stuff. They even use centos repos. |
| 19:33.19 | mpdavis73 | i'll stick to ubuntu |
| 19:33.35 | [TK]D-Fender | NuclearLucifer: Pretty much |
| 19:33.47 | NuclearLucifer | [TK]D-Fender, It's good choice, i think. |
| 19:34.05 | [TK]D-Fender | NuclearLucifer: I prefer to install * direct on my own servers.... |
| 19:34.08 | NuclearLucifer | Long life support and stability of RHEL. |
| 19:34.27 | [TK]D-Fender | NuclearLucifer: Which are usually CentOS anyway, but I go from source. I'm in control, not some packager |
| 19:34.35 | mpdavis73 | i lost my taste for redhat with the whole gcc 2.96 fiasco |
| 19:34.51 | NuclearLucifer | [TK]D-Fender, You can always make your own packages. :) |
| 19:35.07 | NuclearLucifer | As we used to do here. |
| 19:35.28 | NuclearLucifer | Actually, we are not voip provider, but we also use asterisk. |
| 19:35.30 | [TK]D-Fender | NuclearLucifer: No need. I don't redistribute or run multiple large installs. No point. DL. Compile. Enjoy |
| 19:36.07 | NuclearLucifer | [TK]D-Fender, True, it's no need to do that durring one (maybe few) box use. |
| 19:36.56 | NuclearLucifer | [TK]D-Fender, Few days ago, I setup AsteriskNOW on production and it's working quite well. |
| 19:37.07 | NuclearLucifer | No problems noted. |
| 19:37.43 | NuclearLucifer | And fax machines work through t.38. |
| 19:41.22 | *** join/#asterisk pzn (~pzn@pdpc/supporter/active/pzn) |
| 19:42.39 | pzn | I need some help about phone calls... My extension is 200. I intend to dial to extension 201, then put it on music-on-hold. then I dial to extension 202, talk to the people at there, then I connect 201 with 202 and I hang. how can this be made? |
| 19:46.12 | kerframil | pzn: if your device doesn't support it, asterisk itself can provide an attended transfer feature (atxfer in features.conf) |
| 19:49.21 | *** join/#asterisk boynas (~boynas@wsip-68-15-181-232.ph.ph.cox.net) |
| 19:53.37 | boynas | guys, any recomendations for ATAs? here's my deal: I am giving support to a bunch of analog lines E&M through Adtran Channel banks. But I have a problem: If an analog device such as a fax machine is busy, and somebody else calls, it will not play busy signal but will send the call to the channel bank. I need to be able to send back busy signal. |
| 19:53.52 | boynas | I want to install an ATA to test analog using that |
| 19:54.08 | pabelanger | I don't mind the SPA3102 |
| 19:55.02 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:55.25 | boynas | Sorry, can you repeat that.. I just click in "clear" while trying to "copy". I feel like an idiot |
| 19:57.09 | pabelanger | I don't mind the SPA3102 |
| 19:57.22 | pabelanger | Not sure if it will help in your situation however |
| 19:59.53 | boynas | why you say that? |
| 20:01.28 | boynas | wait, spa series come with a router right? do you know any other ATAs without the routing.. is the pap2t the best ATA? there are no commercial ATAs? |
| 20:02.05 | jpmcallister | Hello, I'm gettin this notice on cli: Received SIP subscribe for peer without mailbox: 2308. But that peer has a working mailbox, although it doesn't show with the command "sip show peer 2308" |
| 20:02.48 | *** join/#asterisk gamedna (~Adium@cpe-70-125-155-74.satx.res.rr.com) |
| 20:02.59 | pabelanger | boynas: that's the only one I've used, maybe somebody else can speak up. |
| 20:03.15 | pabelanger | jpmcallister: is your mailbox context setup properly? |
| 20:04.59 | jpmcallister | pabelanger: I'm using users.conf. I'm not setting the context, so I guess it is the default context, right? |
| 20:05.56 | pabelanger | Yes, I believe so. pb your .conf files, redact any passwords tho |
| 20:07.04 | jpmcallister | pabelanger: how do I set the mailbox context in users.conf? |
| 20:07.05 | [TK]D-Fender | GUESSING... may favouritist part EVAR |
| 20:08.36 | pabelanger | jpmcallister: Hmm, good question. I don't think you can. You should look to voicemail.conf |
| 20:09.19 | [TK]D-Fender | pabelanger: Answer sure won't be there |
| 20:10.00 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 20:10.38 | pabelanger | [TK]D-Fender: Of course, but configuring contexts will |
| 20:10.44 | jpmcallister | pabelanger: I was setting the mailbox to the user as mailbox=2308 and the mailbox field showed empty with sip show peer 2308. I changed it to 2308@default. Now it shows 2308,2308@default |
| 20:11.09 | RypPn | kerframil ahh, cool, that version does the trick :) |
| 20:11.20 | pabelanger | jpmcallister: What version of asterisk? |
| 20:11.30 | kerframil | RypPn: yeah. now if I can just get him to investigate the newer instance ... |
| 20:11.37 | kerframil | RypPn: as the code is otherwise much better |
| 20:12.14 | RypPn | kerframil has he acknowledged its broken state? |
| 20:12.18 | jpmcallister | pabelanger: I figured it out. I was using hasvoicemail and mailbox in the user defintion. I removed the mailbox line and now it works fine. I'm using 1..6.2.10 btw |
| 20:12.43 | pabelanger | congrats :) |
| 20:13.01 | kerframil | RypPn: he's aware that the behaviour is different, yes. he's been predisposed lately, so I'm patiently waiting for him to investigate. |
| 20:13.19 | kerframil | RypPn: I suppose I could backport a few of the bug fixes, but I'd rather just get it sorted out ;) |
| 20:14.25 | RypPn | kerframil for sure, its kinda hard to recommend when one of its main features isn't working, but tbh, he's always been breaking it, its a game of finding the last good version |
| 20:15.50 | kerframil | RypPn: that's not my experience. he generally just fixes things (that's how the V2 trunk came into existence, remember?) but this is one exception. |
| 20:16.50 | kerframil | RypPn: by the way, which device are you using? |
| 20:17.20 | RypPn | kerframil 7940, 7970 and an ata-186 |
| 20:17.23 | kerframil | RypPn: ok |
| 20:18.27 | kerframil | RypPn: I'll keep you posted. are you on freenode regularly? |
| 20:18.48 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:19.13 | RypPn | kerframil pretty much maintain a presence even if I'm not actually around :) |
| 20:19.18 | kerframil | RypPn: good good |
| 20:19.55 | *** join/#asterisk bmg505 (~leon@196-209-163-148.dynamic.isadsl.co.za) |
| 20:20.59 | kerframil | RypPn: do me a favour and see if hintpickup=yes works for you also (pickup inbound calls on hinted lines just by pressing its respective button) |
| 20:21.37 | kerframil | RypPn: actually, don't - that came later |
| 20:21.43 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 20:21.50 | wcselby | anyone know what happened to "Buddy Watch" in polycom firmware 3.2.3? |
| 20:22.04 | wcselby | before I start digging through a bunch of change logs, that is |
| 20:22.29 | RypPn | kerframil damn, sounded great for all of 20 seconds there :) |
| 20:22.34 | kerframil | RypPn: heheh |
| 20:26.17 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 20:31.10 | *** join/#asterisk jmacz (~jmacz@158.239.60.190.host.ifxnetworks.com) |
| 20:50.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:50.14 | mpdavis73 | any recommendations for sip providers? |
| 20:50.55 | mpdavis73 | would physical location of the sip provider matter on any level? |
| 20:51.08 | pabelanger | ~itsp |
| 20:51.09 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:51.29 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:51.38 | mpdavis73 | ~itsplist-us |
| 20:51.39 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:51.46 | mpdavis73 | thanks |
| 21:04.13 | *** join/#asterisk dschuett (~1504@216.229.21.250) |
| 21:05.14 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 21:05.29 | dschuett | Hey everyone, I'm running asterisk version 1.2 and have 2 lines on our credit groups phones. I am trying to get distinctive ringtones set up for each line...anyone done this before? |
| 21:05.40 | dschuett | asterisk is running on centos 5 |
| 21:06.09 | dschuett | i have ringlist.xml working fine with custom tones...but i can't select them within each line it say bad ringlist |
| 21:07.05 | [TK]D-Fender | dschuett: that is not an Asterisk config file |
| 21:08.23 | *** part/#asterisk jpmcallister (~ec06113@200.242.28.231) |
| 21:08.40 | dschuett | but i have it working ... |
| 21:08.55 | dschuett | just can't select for each line |
| 21:09.01 | dschuett | ? |
| 21:09.09 | dschuett | what is the correct file then? |
| 21:11.14 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 21:12.09 | dschuett | anyone? |
| 21:18.16 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 21:25.14 | [TK]D-Fender | BAI BAI |
| 21:27.26 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 21:28.22 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 21:29.48 | *** join/#asterisk endigo-ceo (5710de36@gateway/web/freenode/ip.87.16.222.54) |
| 21:30.36 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 21:31.22 | endigo-ceo | hello, general question. i have a POTS line that I need to connect to my pbx, but it's a laptop so I can't install a pci card. is there something out there that i can put on the network or connect via USB to do this? i dont need anything big, just for one line. |
| 21:31.48 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 21:33.36 | pzn | my music on hold is with problems... it has failures... if plays for a second or two, silence for a small time (half a second), then plays again... what should this be? |
| 21:33.58 | pabelanger | bandwidth issue? |
| 21:34.09 | Chainsaw | Or silence suppression at the other end. |
| 21:34.22 | Chainsaw | Make sure it's off, Asterisk does not support it. It will make it very angry, *especially* during MoH. |
| 21:35.00 | pzn | using asterisk, a 100mbps switch and 5 linksys phone adapters. 1 linksys FXO and 4 FXS. |
| 21:35.09 | pzn | will check about silence suppression |
| 21:37.18 | philipp64|laptop | anyone using app_nv_faxdetect with trunk? |
| 21:37.40 | pzn | wow, "silence compression" was enabled in the FXO adapter... setting to off and testing |
| 21:39.03 | russellb | i'm not sure why anyone would use that, both chan_dahdi and chan_sip have fax detection built in |
| 21:39.04 | pzn | Chainsaw, solved!!! right to the point! thanks!!!! |
| 21:39.12 | Chainsaw | pzn: You're welcome :) |
| 21:40.00 | pzn | Chainsaw, the FXS hardware says "silence supression", the FXO says "silence compression", someone forgot to standardize the names in the factory :-) |
| 21:40.48 | Chainsaw | pzn: Ah yes, bit naughty. |
| 21:40.55 | philipp64|laptop | russellb: I don't know why our users ask for it... they just do. probably supporting legacy dialplans. |
| 21:41.17 | Qwell | or because they don't know better. |
| 21:42.04 | [TK]D-Fender | endigo-ceo: Linksys SPA-3102 |
| 21:42.34 | [TK]D-Fender | endigo-ceo: Networked, not USB however |
| 21:44.21 | pzn | is there any number where I can dial to check the musiconhold ? |
| 21:45.36 | philipp64|laptop | Qwell: if they knew better, they might not be using our distro. :-) |
| 21:45.44 | Qwell | No comment. |
| 21:45.52 | philipp64|laptop | rat bastard |
| 21:46.16 | endigo-ceo | [TK]D-Fender: that would actually be better than USB, it only has USB 1.1 ports |
| 21:46.36 | Chainsaw | pzn: If you've defined a parking lot, the first park extension. |
| 21:47.03 | endigo-ceo | [TK]D-Fender: is this something that can be implemented into FreePBX easily? |
| 21:47.10 | pzn | Chainsaw, I'm new to voip and asterisk. what is a parking lot? can you tell me some hint about how to do it? |
| 21:47.27 | Chainsaw | pzn: It's where you can temporarily transfer a call if you're switching it between phones. |
| 21:47.40 | Chainsaw | pzn: You can also add an extension to your dial plan that does nothing but play hold music. |
| 21:48.08 | pzn | Chainsaw, I'd like to add an extension to play musiconhold. how? |
| 21:48.33 | Chainsaw | pzn: Edit extensions.conf in /etc/asterisk |
| 21:48.36 | Chainsaw | pzn: Unless you have some GUI installed? |
| 21:48.43 | pzn | no gui |
| 21:48.52 | Chainsaw | pzn: Ah, good. |
| 21:48.58 | philipp64|laptop | I'll ask our users if they can cut over to the channel native support. |
| 21:50.02 | Chainsaw | pzn: Try this: http://www.pastebin.ca/1912495 |
| 21:50.14 | Chainsaw | pzn: Put that in, on the console: dialplan reload |
| 21:50.22 | Chainsaw | pzn: Then call 555 |
| 21:51.08 | pzn | Chainsaw, nice! I'll do it. |
| 21:52.34 | Chainsaw | pzn: Add an extra line: exten => 556,1,MusicOnHold |
| 21:52.40 | Chainsaw | pzn: Then you have 555 & 556 to play with. |
| 21:53.40 | pzn | Chainsaw, I can put rolingsstones sympathy for the devil on musiconhold at extension 666 :-) |
| 21:54.01 | Chainsaw | pzn: Yes you can. You may have to pay them royalties for that though. |
| 21:54.39 | pzn | :-D |
| 21:56.49 | pzn | Chainsaw, everithing working as expected now. thanks for the help. |
| 21:56.50 | *** part/#asterisk bsaxon_ (~bsaxon@12.107.149.61) |
| 21:56.58 | Chainsaw | pzn: You're welcome. |
| 21:57.36 | [TK]D-Fender | endigo-ceo: Yes |
| 21:58.26 | endigo-ceo | [TK]D-Fender: generally, what would it take to set up? |
| 21:58.35 | [TK]D-Fender | endigo-ceo: time |
| 21:58.42 | endigo-ceo | lol ok |
| 21:59.23 | endigo-ceo | im pretty technical, but this is a production PBX, not a test system |
| 21:59.41 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 22:00.31 | Qwell | production PBX on a laptop? |
| 22:00.38 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 22:01.01 | citywok | i recently started using g729 instead of g711, and my hold music sounds terrible. currently i have it in .wav, what tools do people use to get it in to g729? |
| 22:01.43 | Qwell | G.729 wasn't made to handle music. |
| 22:01.54 | Qwell | I'm not surprised that it sounds terrible. |
| 22:02.22 | citywok | it cuts out about 50% of the time. is that b/c of the compression cutting out non-human frequencies? |
| 22:02.32 | citywok | (it's piano music) |
| 22:02.35 | Qwell | cuts out, like the audio just stops? |
| 22:02.49 | pzn | citywok, I opened my music file on some audio tool, applied a filter to cut > 8000Hz then saved it in .wav; it got better, but sure it is not as good as desired |
| 22:02.50 | Qwell | turn off silence suppression on your endpoint. I'd tell you to scroll up, but you just joined.. |
| 22:03.03 | *** part/#asterisk mpdavis73 (~marshalld@67.211.235.66) |
| 22:03.14 | citywok | it's like a stutter. in/out/in/out |
| 22:03.32 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 22:03.44 | citywok | what do people do to accomodate, use different music? Not use g729? |
| 22:04.06 | Qwell | How long does it cut out for? |
| 22:04.50 | citywok | 1/4 or an 1/8 of a second. THis is MOH while waiting for other members to join a meetme conference. |
| 22:05.05 | Qwell | yeah, make sure you don't have silence suppression |
| 22:05.08 | citywok | Interestingly i don't get the same effect when i call my cellphone and put the call on hold. It sounds fine on my cell. |
| 22:05.30 | Qwell | easy easy way to test. |
| 22:05.38 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 22:05.42 | Qwell | blow into the mic at a constant rate for about 10s |
| 22:05.51 | Qwell | if it doesn't cut out...it's obviously silence suppression |
| 22:05.58 | citywok | 1sec, will try |
| 22:06.33 | citywok | that's exactly it. it's perfect as long as i blow. |
| 22:06.38 | Qwell | mmhmm |
| 22:06.44 | Qwell | That'll be $1499.99 |
| 22:06.50 | citywok | as soon as i quit blowing into the mic it starts stuttering again. perfect, i'll be looking for that flag now. |
| 22:06.56 | citywok | hahaha. can i pay you in southpark internet monies? |
| 22:06.58 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 22:07.14 | Qwell | can I buy robotic mowers with southpark internet monies? |
| 22:07.21 | citywok | in canada.... |
| 22:07.36 | Qwell | silly canada |
| 22:09.33 | citywok | this option? 'o' -- set talker optimization - treats talkers who aren't speaking as |
| 22:10.05 | Qwell | no. it's on your endpoint. |
| 22:10.12 | Qwell | (ie; not asterisk) |
| 22:10.26 | citywok | ooooooh. aastra manual here i come :P |
| 22:10.59 | citywok | thanks for your help Qwell |
| 22:11.06 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 22:12.51 | citywok | bingo. found the option in my phone and it works. adding that to the rest of my phones now :) -- Muchas Gracias. |
| 22:15.03 | *** join/#asterisk Russ (~russ@206.29.188.229) |
| 22:20.26 | CoffeeIV | I have an asterisk running on the Rackspace cloud (ubuntu). when we do a conference call, often one party has a long delay -- of as much as 15 seconds - but otherwise the sound quality is fine. Is there some timing related setting I can activate in asterisk / ubuntu, or is the Rackspace clound just not going to work for this ? |
| 22:25.21 | pabelanger | CoffeeIV: Could be a bandwidth issue, how much are you provided? |
| 22:25.52 | pabelanger | Or transcoding issue |
| 22:26.02 | pabelanger | if you have multiple codecs |
| 22:27.19 | *** join/#asterisk seanjohn (~admin@gateways.sheltoncomputers.com) |
| 22:27.59 | CoffeeIV | possibly . . . wouldn't I expect more choppy dropouts if it was bandwidth ? Not just a clean sounding delay ? |
| 22:28.40 | CoffeeIV | there is no transcoding issue, both the delayed party and the non-delayed party are reaching the box via the same route, calling a vitelity number |
| 22:29.03 | CoffeeIV | at least, I think that means it would not be a transcoding issue |
| 22:29.18 | CoffeeIV | there is no load on the box, and it is not running out of memory |
| 22:39.04 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 22:41.55 | b14ck | Hey, if I have a digium card plugged in. What command do I need to run to auto-generate chan_dahdi.conf for me? |
| 22:42.03 | b14ck | (using the latest dahdi linux + asterisk)? |
| 22:42.18 | pzn | can you recommend some cheap 4x FXO card? |
| 22:44.40 | chazzam | b14ck: dahdi_genconf will auto generate dahdi-channels.conf, which can then be #include'd in chan_dahdi.conf |
| 22:45.16 | b14ck | Is there anyway to auto generate a chan_dahdi.conf file for my digium card? |
| 22:45.24 | b14ck | I've never used a digium card before, I'm used to the sangoma script setting i tup. |
| 22:45.26 | b14ck | *it up |
| 22:45.52 | chazzam | I think the way dahdi_genconf does it is the closest you will get |
| 22:46.03 | chazzam | it makes it a one line edit to chan_dahdi.conf |
| 22:46.09 | chazzam | '#include dahdi-channels.conf' |
| 22:46.13 | chazzam | at the bottom of the file |
| 22:46.19 | b14ck | ok |
| 22:46.50 | b14ck | Well, let me be more verbose: |
| 22:46.56 | b14ck | I have a sangoma PRI card in this server (configured). |
| 22:47.04 | b14ck | I just plugged in a digium FXS card. |
| 22:47.07 | b14ck | Which I'm trying to configure. |
| 22:47.22 | b14ck | If I just run: dahdi_genconf chandahdi=verbose it will genreate dahdi-channels.conf |
| 22:47.31 | b14ck | So should I then just include that into my chan_dahdi.conf? |
| 22:47.45 | b14ck | I think the way it is detecting it now--it is detecting my sangoma card, not my digium one. |
| 22:47.53 | b14ck | So basically, I can use my PRI card, but not my FXS at all. |
| 22:47.56 | chazzam | oh, I dunno if dahdi_genconf will write configuration for the sangoma or not, if it does you would prolly want to remove the stuff it wrote for the sangoma |
| 22:47.58 | b14ck | genconf doesn't seem to recognize that one |
| 22:48.08 | chazzam | do you have the driver for the FXS card loaded? |
| 22:48.26 | b14ck | I didn't load any custom drivers. |
| 22:48.26 | chazzam | you can run dahdi_hardware I believe to determine which driver you need |
| 22:48.34 | chazzam | and then use lsmod to check if that one is loaded |
| 22:48.51 | b14ck | dahdi_hardware doesnt show anything |
| 22:48.55 | b14ck | it just exits |
| 22:49.17 | chazzam | that could be an indication of a deeper problem then |
| 22:49.22 | chazzam | does lspci show the card? |
| 22:49.30 | *** join/#asterisk russ (~russ@206.29.188.181) |
| 22:49.31 | b14ck | yes, it shows them both |
| 22:49.48 | chazzam | what version of dahdi, and what fxs card? |
| 22:49.58 | b14ck | actually |
| 22:49.59 | b14ck | i lied |
| 22:50.04 | b14ck | dahdi_hardware does detect them both |
| 22:50.04 | b14ck | pci:0000:01:0a.0 wctdm24xxp+ d161:8005 Wildcard TDM410P |
| 22:50.05 | b14ck | pci:0000:01:0c.0 wanpipe- 1923:0040 Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 22:50.19 | chazzam | is wctdm24xxp loaded? |
| 22:50.52 | b14ck | yes |
| 22:50.55 | b14ck | (in lsmod) |
| 22:50.56 | chazzam | either the pci-id name of that card is wrong, or that isn't a PRI card |
| 22:50.59 | b14ck | and here's my chan_dahdi.conf |
| 22:51.03 | b14ck | http://pastie.org/private/a9ktaaesek9fvtwzcxytpw |
| 22:51.14 | b14ck | That is a PRI sangoma card (which currently works) |
| 22:51.24 | b14ck | I'm just trying to get the digium card to work with it |
| 22:51.27 | chazzam | pci-id must be wrong then, weee |
| 22:51.39 | b14ck | lspci shows: |
| 22:51.40 | b14ck | 01:0a.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card (rev 11) |
| 22:51.40 | b14ck | 01:0c.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS Analog AFT card |
| 22:51.47 | chazzam | what is dahdi-channels.conf then? |
| 22:52.05 | b14ck | dahdi-channels.conf: http://pastie.org/1080594 |
| 22:52.21 | b14ck | weird, right? |
| 22:52.29 | b14ck | This is running the latest DAHDI, asterisk, and everything |
| 22:52.32 | b14ck | brand new install |
| 22:52.33 | chazzam | thats from running it just now? |
| 22:52.37 | b14ck | correct |
| 22:52.43 | b14ck | I ran: |
| 22:53.01 | b14ck | dahdi_genconf chandahdi=verbose |
| 22:53.13 | chazzam | what if you leave off the chandahdi=verbose? |
| 22:53.22 | chazzam | I haven't ever tried to use a chandahdi parameter with it |
| 22:53.47 | b14ck | http://pastie.org/1080601 |
| 22:53.52 | b14ck | (ran without the paramater) |
| 22:53.58 | b14ck | looks identical |
| 22:54.20 | b14ck | I'm surprised that it is not picking up the digium, but is picking up the sangoma =p |
| 22:54.25 | b14ck | I figured it would be the opposite problem. |
| 22:54.58 | chazzam | it sounds like there is something keeping the card from being initialized by the driver then |
| 22:55.06 | chazzam | does the tdm410 show up in dmesg? |
| 22:55.20 | chazzam | but that script is a community thing too... |
| 22:56.08 | *** join/#asterisk unspin (~unspin@S010600031d02196a.vc.shawcable.net) |
| 22:56.15 | b14ck | Well, it shows up in dahdi_hardware, the kernel module is loaded (lsmod), and it is detected everywhere |
| 22:56.15 | b14ck | lol |
| 22:56.24 | b14ck | Is there a hard-coded config I can use for it or something? |
| 22:56.38 | chazzam | dahdi_hardware doesn't need the driver loaded to detect the card |
| 22:56.57 | chazzam | and the driver being present in lsmod doesn't mean it initialized the hardware |
| 22:57.03 | b14ck | Oh yah? |
| 22:57.08 | chazzam | if another driver already initialized it, then the new driver can't |
| 22:57.15 | b14ck | Where is the driver for it? |
| 22:57.23 | chazzam | driver for what |
| 22:57.24 | b14ck | The only driver I installed was for my sangoma (wanpipe) |
| 22:57.28 | b14ck | For the TDM410 I have |
| 22:57.34 | *** join/#asterisk [netman] (~netman@83.52.208.225) |
| 22:57.36 | b14ck | Do I need somethign special? Other than dahdi? |
| 22:58.14 | *** part/#asterisk boynas (~boynas@wsip-68-15-181-232.ph.ph.cox.net) |
| 22:58.26 | chazzam | dahdi provides the wctdm24xxp driver |
| 22:58.38 | chazzam | what the answer for dmesg, pastebin that |
| 22:59.49 | *** join/#asterisk b0gatyr (~b0gatyr@adsl-10-117-193.mia.bellsouth.net) |
| 23:00.31 | b14ck | chazzam, http://pastie.org/private/mrhippkk63mtiph0hspupa |
| 23:01.52 | chazzam | the wctdm24xxp isn't in there |
| 23:02.06 | b14ck | Oh. |
| 23:02.11 | b14ck | So what do I need to get it in there, I suppose? |
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| 23:37.49 | Trixboxer | Hi, can somebody tell me about "which agi variable stores the outgoing number in the /var/lib/asterisk/agi-bin/recordingcheck' file ? " |
| 23:38.01 | Trixboxer | I have tried variable "${OUTNUM}" |
| 23:56.31 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |