| 00:00.52 | aptura | reading sip.cfg it is 1.57.2.1 |
| 00:02.17 | mmartinn | This is off topic, but has anyone seen any USB mass storage "emulators" ever? |
| 00:02.28 | aptura | I had to reformat the systems os and ip500s image copies on my other server but I also have other versions and just perhaps it is the wrong one. |
| 00:03.11 | [TK]D-Fender | aptura, Upgrade your sip config first, then your BR. |
| 00:03.15 | aptura | phones vrtsion is 3.1.0.0269 |
| 00:03.24 | aptura | k |
| 00:03.32 | [TK]D-Fender | aptura, and you are completely mixing up SIP & BR revisions. |
| 00:03.39 | aptura | very possible |
| 00:07.35 | aptura | so the phone reads one files in sequence. |
| 00:07.35 | astranik | have anyone using asterisknow with les.net? |
| 00:07.38 | JT | mavior: i'm not sure if it's actually possible for different channels to have different load zone |
| 00:07.51 | aptura | getting a error loading "macadress".cfg in the display |
| 00:08.42 | mavior | JT:that sound really strange... |
| 00:09.47 | JT | not really, what's really strange is your requirement for multiple tone zones on a zap card, mavior |
| 00:11.02 | mavior | JT: have to really set up a multilanguage system :) |
| 00:11.14 | JT | why? |
| 00:12.20 | mavior | this is not a real question |
| 00:12.35 | mavior | or better a gentle question :) |
| 00:12.39 | [TK]D-Fender | tone != language. |
| 00:12.44 | [TK]D-Fender | hows that? |
| 00:12.49 | mavior | yes i know... |
| 00:12.53 | [TK]D-Fender | words == language |
| 00:13.07 | mavior | i know... |
| 00:13.17 | [TK]D-Fender | mavior, So what psycho reason would you have for internationalizing tones within the same physical server? |
| 00:13.37 | [TK]D-Fender | mavior, How cares that much to feel like they're "back home"? |
| 00:13.57 | aptura | TK do you know what sip versions belong with which bootrom.ld versions? If not then will continue to search for the info. |
| 00:13.59 | [TK]D-Fender | mavior, And if you care enough, get them their own ATA and be done with it. |
| 00:14.00 | mavior | but having even the tones for different phones would be really cool |
| 00:14.12 | [TK]D-Fender | aptura, Just upgrade to the latest of each |
| 00:14.27 | *** part/#asterisk saftsack (n=saftsack@pD9E06E23.dip.t-dialin.net) |
| 00:14.36 | JT | what [TK]D-Fender said |
| 00:14.40 | JT | a pile of ATAs |
| 00:14.43 | JT | or SIP phones |
| 00:14.46 | mavior | and from what i read from the config files....it's probably possible to have such a configuration |
| 00:15.05 | aptura | okay |
| 00:15.15 | [TK]D-Fender | mavior, And what kind of parameters have you attempted so far? |
| 00:18.27 | *** join/#asterisk jovannotti (n=jovannot@69.79.144.206) |
| 00:21.38 | mavior | [TK]D-Fender ehm...mainly i've played a bit with zaptel.conf....what sounds to me good is that on zaptel.conf http://pastebin.ca/442059 i read something(from line 27 in the page to the bottom) that sounds like i can decide which parameters for which channels i can configure...what i dunno is HOW(i mean the syntax) |
| 00:24.03 | aptura | file error is 0x10000 |
| 00:24.14 | aptura | well at least the error is different! |
| 00:24.16 | aptura | :) |
| 00:26.22 | [TK]D-Fender | aptura, corrupt config file. rebuild your configs for your firmware rev |
| 00:26.53 | [TK]D-Fender | mavior, funny I don't see any parameters in there... |
| 00:28.13 | aptura | TK I have some info of the configs but most of it was never configured and did not need to to make the phone run for a long time so short of being apolycom reseller and accessing there site will locate any info on line to change the configs. |
| 00:28.50 | mavior | [TK]D-Fender, i read"We are all done with our channel parameters, so now we specify what channels they apply to"...this sounds like something to decide which parameters to which channel...isn't it? |
| 00:29.11 | *** part/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) |
| 00:29.31 | [TK]D-Fender | mavior, you have *1* meaningful line in that pastebin for your channel definitions, and thats merely to NUMBER them. |
| 00:29.58 | [TK]D-Fender | mavior, you set no callerID, no signalling type, no flash based featues, EC, NOTHING. |
| 00:30.13 | aptura | taking a break |
| 00:30.17 | [TK]D-Fender | mavior, that is not a config by any reasonable definition. |
| 00:30.26 | mavior | yes i have |
| 00:32.03 | mavior | that one is just an extract...i've not copyed all the file zaptel.conf...if i've understood you (sorry for my english) |
| 00:32.03 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 00:32.03 | [TK]D-Fender | mavior, what useful amount of information do YOU see in this pastebin of yours? http://pastebin.ca/442059 |
| 00:32.11 | [TK]D-Fender | mavior, I asked you to show me what you TRIED. This tells me you tried NOTHING. |
| 00:32.13 | mavior | ok gonna post my entire zaptel |
| 00:32.18 | [TK]D-Fender | ZAPATA! |
| 00:32.25 | [TK]D-Fender | *sheesh* |
| 00:39.13 | jovannotti | something know about this message ? Apr 15 19:35:22 WARNING[16532]: chan_zap.c:4931 zt_write: Frame too large |
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| 00:39.35 | mavior | [TK]D-Fender,let me make a question: why is possible to load more than one loadzone, if then you can not configure and have the possible to change in some conf files an option to use them?? |
| 00:39.47 | *** part/#asterisk kvidell (i=id8272@adsl-63-204-157-35.dsl.snfc21.pacbell.net) |
| 00:40.00 | mavior | s/possible/possibility/ |
| 00:40.15 | JT | zaptel indications are different to indications.conf indications |
| 00:41.00 | [TK]D-Fender | mavior, Did I say you couldn't? No I did not. I asked you to show me what you've tried, and I still don't have anything. |
| 00:41.40 | mavior | what do you mean JT ? |
| 00:42.25 | JT | mavior: zaptel channel indications are different to dialplan generated indications, which uses indications.confd |
| 00:42.29 | JT | .conf |
| 00:42.47 | mavior | hey i'm friend...it was just for start to guessing..i will show something fender |
| 00:50.54 | mavior | [TK]D-Fender, http://pastebin.ca/442099, look the last five lines, i've tried to specify something different for the channel 2, following the logic of channel definition in zapata.conf file |
| 00:51.24 | mavior | but it does not work :| |
| 00:52.08 | [TK]D-Fender | mavior, You set loadzone like 5 times in a row overriding it al the time |
| 00:52.25 | [TK]D-Fender | mavior, permanently remove all that commented out junk. |
| 00:52.36 | [TK]D-Fender | s/al/all |
| 00:52.47 | mavior | which lines? |
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| 00:54.04 | [TK]D-Fender | mavior, ..... |
| 00:54.10 | [TK]D-Fender | <[TK]D-Fender> mavior, permanently remove all that commented out junk. |
| 00:56.06 | mavior | do you mean that this pastebin is not good for you? |
| 00:57.01 | [TK]D-Fender | mavior, how much more clear can I be? |
| 00:58.57 | [TK]D-Fender | mavior, remove all of those commented out lines. Permanently. Re-pastebin afterwards |
| 01:01.41 | JT | if not, use osmosis education method |
| 01:05.10 | mavior | in the meantime take this [TK]D-Fender http://pastebin.ca/442119 |
| 01:06.47 | mavior | this is like the default zaptel.conf....the only difference are the last 5 lines |
| 01:06.57 | mavior | it's not so complicated :) |
| 01:07.07 | aptura | Something I did not know is you cannot go back to previos bootrom verions prior to 3.1.x without incuring expensive in house repairs by polycom. |
| 01:08.09 | jovannotti | someone where I can found these 2 diffences: |
| 01:08.10 | [TK]D-Fender | mavior, its 50 lines of crap to filter through where I can see you set values and override them in the SAME CHANNEL DEFINITION. |
| 01:08.20 | jovannotti | 1. between asterisk 1.2 and asterisk 1.4 |
| 01:08.21 | [TK]D-Fender | 500* |
| 01:08.31 | jovannotti | 2. between slin and u-law |
| 01:08.32 | jovannotti | thanks |
| 01:08.50 | [TK]D-Fender | aptura, time to move FORWARD |
| 01:09.01 | mavior | [TK]D-Fender ive ovveride nothing in that file |
| 01:09.22 | mavior | i've set just one time the values i needed |
| 01:09.37 | aptura | Ohh I know. There is a chart on which sip verions belong to with bootrom versions since thay are entirely different numbers sequences? |
| 01:09.50 | [TK]D-Fender | <PROTECTED> |
| 01:10.00 | Mavvie | [TK]D-Fender: wrong addressee |
| 01:10.22 | [TK]D-Fender | Mavvie, hukt on fonix werkt 4 me! |
| 01:10.39 | [TK]D-Fender | mavior, As I said to our poorly targetd friend... |
| 01:11.08 | mavior | ? |
| 01:11.15 | [TK]D-Fender | mavior, in there we can see the last zone you tried to load is IT, which means ALL zones for 1-4 are IT |
| 01:11.34 | [TK]D-Fender | mavior, lines 318 & 324 |
| 01:12.09 | mavior | http://pastebin.ca/442119 max line is 235 in my browser :) |
| 01:12.57 | [TK]D-Fender | mavior, I was reading the first run you gave me : http://pastebin.ca/442099 |
| 01:13.38 | mavior | please use this one http://pastebin.ca/442119 , it's clearer |
| 01:14.14 | [TK]D-Fender | mavior, and where in there do I see you setting zones different for your channels? |
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| 01:17.44 | mavior | really hard to get something here.....fender! like i said before....i've just tried to play a bit with channels definition....you know the syntax to define options for each channel in zapata.conf ? is like [options] and then channel => 1 [options] and then channel=>2 and so on...so i tried to do something like that even in my zaptel , to try out if it works even here....like i said I DONT... |
| 01:17.46 | mavior | ...KNOW THE SYNTAX TO SPECIFY DIFFERENT options for DIFFERENT channels in zaptel.conf |
| 01:18.10 | mavior | is it ok? |
| 01:18.50 | [TK]D-Fender | mavior, you try setting loadzone once, but not twice. that shows me you haven't atucally tried anything at all. |
| 01:19.17 | mavior | after all i made a question...simple: do you know the syntax? |
| 01:19.20 | [TK]D-Fender | Mavvie, You can do : myshitsetting=true , and * won't blow up you know. you haven't even tried based on what you took so long to show. |
| 01:19.40 | mavior | no i load two loadzones |
| 01:20.01 | [TK]D-Fender | Mavvie, You seem to have what I igure is the right syntax "loadzone=[zonename]" but have not even set it differently between your channels. |
| 01:20.10 | [TK]D-Fender | mavior, What line #'s? |
| 01:20.28 | mavior | one for channels 1,3,4 and one for channel 2 (in my supposed working syntax) |
| 01:20.39 | [TK]D-Fender | mavior, WHERE!? |
| 01:21.38 | mavior | oh ok my fault...line 234 -235 are not commented out in my conf |
| 01:22.13 | [TK]D-Fender | mavior, ask again later when you clean out the commented junk in your configs and can actually show me what you're really trying. |
| 01:22.29 | [TK]D-Fender | mavior, because this has beena futile waste of my time... |
| 01:24.06 | *** join/#asterisk SomethingISODD (n=dan@xplr-ts-v10-208-114-188-192.barrettxplore.com) |
| 01:25.14 | SomethingISODD | hello all question has anyone ever used the program called vocal from vovida? |
| 01:28.21 | JT | anyone have some recommendations for some high density channel banks? |
| 01:29.14 | [TK]D-Fender | JT, typical 24-port? |
| 01:29.31 | [TK]D-Fender | JT, And by that do ou specifically require T1? |
| 01:30.13 | JT | E1 would be nice, it'd mean less t1/e1 ports, but i've never seen an E1 c/b |
| 01:30.23 | [TK]D-Fender | Oh God.... that white shits coming down in force again now..... |
| 01:30.34 | JT | high density, something that doesn't take up tonnes of space for 120 ports |
| 01:30.38 | JT | FXS only |
| 01:30.41 | [TK]D-Fender | JT, I typically advise SIP gatways over CB's |
| 01:31.00 | JT | aren't they ridiculously expensive? |
| 01:31.09 | [TK]D-Fender | JT, for which that'd be the MediaTrix 1124 |
| 01:31.17 | JT | hmm |
| 01:31.20 | JT | must handle fax |
| 01:31.23 | JT | also |
| 01:31.33 | [TK]D-Fender | JT, roughly on par, not requiring a T1 card either, lower load and greater functionality. |
| 01:31.46 | [TK]D-Fender | Hrm... not sure on the fax part. |
| 01:31.54 | JT | which maybe a sip disadvantage, if it has no t.38 |
| 01:32.06 | [TK]D-Fender | JT, I've used Rhino's CB, but its 2U for 24 port. |
| 01:32.14 | JT | hmm |
| 01:32.26 | JT | don't like adtran? |
| 01:33.24 | [TK]D-Fender | JT, I respect the brand, just never tried them |
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| 01:34.03 | JT | i was otherwise thinking high capacity analogue pci cards, but that doesn't seem scalable |
| 01:35.23 | [TK]D-Fender | JT, you know Digium highly recommends against more than 2 cards in a system..... |
| 01:35.45 | Snake-Eyes | Any suggestions for what would be a good comand to run in asterisk console to determine that asterisk is fine, (I'm thinking of a sip comand) ? |
| 01:35.55 | [TK]D-Fender | JT, fax req is the killer. check out the MediaTrix 1124 and AudioCodes MP-124 |
| 01:35.59 | JT | yeah, 48 ports a pc means i'd need at least 3 PCs, not very scalable |
| 01:36.39 | JT | [TK]D-Fender: any idea on cost? |
| 01:36.52 | [TK]D-Fender | JT, And an 8-port card would do the job for 192 ports |
| 01:37.00 | [TK]D-Fender | JT, about $1500 US each |
| 01:38.06 | JT | 8 port card, you mean with T1 channel banks? |
| 01:38.18 | [TK]D-Fender | JT, yes, were you to go down that route |
| 01:38.19 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 01:38.24 | Snake-Eyes | hey JT, you know how we were talking about police/crime etc the other day? |
| 01:38.41 | JT | ah, my 48 ports a pc spec was with respect to FXS PCI cards |
| 01:38.48 | JT | Snake-Eyes: umm maybe |
| 01:38.56 | [TK]D-Fender | JT, Adit & Adtran are very respected CB's, and a Sangoma A108(d) would serve them well |
| 01:39.37 | Snake-Eyes | JT, well some ppl broke into my house on friday |
| 01:39.59 | JT | looking at the mediatrix 1124, it might do the job also |
| 01:39.59 | JT | Snake-Eyes: damn |
| 01:40.33 | Snake-Eyes | thought it was kind of ironic, also it being friday 13th .... |
| 01:41.12 | JT | hope they didn't get away with much |
| 01:41.32 | JT | [TK]D-Fender: the mediatrix unit looks fairly price competitive, going off $1500 |
| 01:41.58 | [TK]D-Fender | JT, go read up on those 2 to see if their feature set meets your needs |
| 01:44.19 | JT | [TK]D-Fender: they seem very similar price/feature wise |
| 01:44.29 | aptura | snake-eyes thats because the theives found little of value or thay were spooked by some one in the area who discovered them. |
| 01:44.41 | [TK]D-Fender | JT, Gogle up t.38 etc and see what you get |
| 01:44.53 | JT | heh |
| 01:45.15 | JT | i know a bit about t.38, but i don't see how that will help me decide between the two :) |
| 01:45.38 | JT | seems a better option that T1 channel bank, since customers don't usually like the idea of second hand gear |
| 01:46.27 | Snake-Eyes | they took tv but left alot of other stuff, it came across they only wanted certain kinds of items |
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| 02:43.17 | mklebel | is there a guide somewhere or HOWTO that shows me the light on voice conferencing over http? |
| 02:43.42 | JT | yeah it can't be done |
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| 02:45.13 | mklebel | why? |
| 02:45.32 | hijacked | that's not what it's for? |
| 02:46.08 | JT | http is a tcp based protocol |
| 02:46.21 | JT | tcp is unsuitable for realtime interactive media like VoIP |
| 02:46.24 | JT | if that's what you mean |
| 02:46.38 | mklebel | what about https? |
| 02:47.29 | [TK]D-Fender | mklebel, same |
| 02:50.24 | JT | that'd be worse |
| 02:50.31 | JT | encryption adds more delay |
| 02:54.08 | sbingner | JT, you mean it SHOULDN'T be done... not it can't be done ;) |
| 02:54.37 | JT | when it comes to VoIP, can't is pretty much the right word, with respect to http |
| 02:54.48 | JT | you can use flash or java and a mor suitable protocol in a web page |
| 02:54.57 | JT | but not send voice media in real time over http |
| 02:55.00 | sbingner | you could do it by using one of the silly IP-over-HTTP packages out there ;) |
| 02:56.07 | JT | not if latency matters though |
| 02:56.08 | sbingner | but you'll get no argument from me on the fact that there's no good reason to actually do it |
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| 04:04.36 | DocHolliday | was the trixbox appliance engineered from the ground up as a 'carrier grade voip gateway / appliance'? |
| 04:05.43 | SwK | its engineered as a PBX replacement |
| 04:05.50 | SwK | imho |
| 04:06.18 | DocHolliday | SwK, whats the best option if you have a customer with a TDM PBX and you simply want to hand off SIP them them using their existing PRI ports? |
| 04:06.26 | DocHolliday | *to |
| 04:06.38 | SwK | DocHolliday how many ports? |
| 04:07.07 | SwK | (T1 ports that is) |
| 04:07.16 | DocHolliday | 24-48 + ~ |
| 04:07.25 | SwK | T1s or DS)? |
| 04:07.25 | DocHolliday | ohh :) 1 - 4 |
| 04:07.33 | DocHolliday | T1 |
| 04:07.36 | SwK | heh ok |
| 04:08.13 | SwK | well that depends on what your budget is like... it can be done w/ an asterisk box, but I would lean more toward a Cisco AS5300 or one of their IAD series |
| 04:08.14 | [TK]D-Fender | DocHolliday, Its a friggen PC with a Telecom card. And Trixbox..... |
| 04:08.24 | [TK]D-Fender | ~trixbox |
| 04:08.39 | jbot | Trixbox is a full linux distro that includes *, FreePBX, and other 3rd party add-ons. It is these things on top of * which make it seriously painful to support and hence you will find little help here for it. Try asking in #freepbx , or their forums at http://www.trixbox.org/modules/newbb/ |
| 04:08.57 | [TK]D-Fender | DocHolliday, Its just comes withe a few extra niceties that make it more robust, but its just a Linux box..... |
| 04:08.57 | DocHolliday | [TK]D-Fender, i dont want to give a customer an entire linux box with PCI TDM cards.. thats way too much responsibility / probability of issues |
| 04:10.36 | [TK]D-Fender | DocHolliday, Think of it as a headless server. How different do you think you can make it? |
| 04:10.37 | JT | DocHolliday: what is this trixbox appliance you speak of? |
| 04:10.37 | CpuID2 | hmm, anyone know if theres any digium reps around? |
| 04:10.37 | DocHolliday | [TK]D-Fender, i'd rather have an integrated appliance or purpose built device |
| 04:10.39 | CpuID2 | need to get some g729 codecs relicensed :P |
| 04:10.39 | DocHolliday | JT, go to the trixbox site :) |
| 04:10.39 | CpuID2 | good old useless digium email support so far...gah |
| 04:10.40 | JT | no thanks |
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| 05:29.46 | hoowa | hi |
| 05:29.59 | hoowa | iftime funtion registed from which one module in asterisk? |
| 05:30.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:36.36 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:43.38 | *** join/#asterisk rahail (i=Oh-Ya@12.191.5.194) |
| 05:43.49 | rahail | is any one using Asteriks to Terminate calls |
| 05:44.31 | JT | quite a few |
| 05:45.53 | rahail | what is the best way to terminate using Gsm network |
| 05:46.07 | rahail | if you have idea JT |
| 05:46.19 | JT | if you must terminate using gsm sims, than a gsm gateway would be best |
| 05:46.36 | rahail | any recomendation on gsm gw |
| 05:46.55 | JT | no, either a T1/E1 unit or SIP, SIP would be ideal |
| 05:46.59 | JT | they're not cheap though |
| 05:47.31 | rahail | well thing is they country i want do termination |
| 05:47.40 | rahail | I am not gone abel to ship that gsm gw |
| 05:47.53 | rahail | coustome will stop |
| 05:48.30 | JT | why, is it illegal? |
| 05:49.08 | rahail | goverment think if people do voip goverment phone line lose money |
| 05:49.08 | rahail | :) |
| 05:49.08 | JT | middle east? |
| 05:49.10 | rahail | close |
| 05:49.12 | rahail | Bangladesh |
| 05:49.39 | rahail | its next to India |
| 05:50.04 | JT | i see |
| 05:50.21 | rahail | I am trying to get way to do that again |
| 06:06.20 | *** join/#asterisk Ifaistos (n=stelios@ipa226.211.tellas.gr) |
| 06:14.20 | Snake-Eyes | i think india has the same/simliar anti voip regulations |
| 06:14.41 | JT | quite a few countries seem to have that sort of thing |
| 06:14.51 | JT | glad i'm not in one of those countries |
| 06:14.54 | rahail | lol |
| 06:15.00 | Snake-Eyes | voip can be used to phone outside the country but not internal or something |
| 06:15.08 | rahail | Me to but I have realtives and friends back there |
| 06:15.15 | rahail | so I provide kind off calling card service |
| 06:15.28 | JT | kind of get arrested |
| 06:15.41 | Snake-Eyes | well if telstra was still 100% gov owned , we might have had something like that here |
| 06:16.07 | JT | nah the government opened up the telco market before they started to sell it off |
| 06:16.40 | rahail | Questiong for you guiess insted of gsm gw |
| 06:16.46 | rahail | is there any solution i can use it with asterisk |
| 06:17.07 | *** join/#asterisk jazzplyer (n=jazzplye@222-154-246-214.adsl.xtra.co.nz) |
| 06:17.11 | *** part/#asterisk jazzplyer (n=jazzplye@222-154-246-214.adsl.xtra.co.nz) |
| 06:17.19 | Snake-Eyes | they opened up companies to resell there products, its not exactly the same thing IMO |
| 06:18.09 | JT | the optus mobile phone network was not a resold telstra product |
| 06:19.13 | Snake-Eyes | true, but no one is going to lay another copper network? which is what voip really threatens |
| 06:20.32 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:20.35 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:20.48 | Snake-Eyes | alot gov's when owning 100% of the copper network want to preserve there profit streams |
| 06:21.05 | JT | they're laying other fibre networks |
| 06:21.13 | JT | and others can use the copper for cheap enough |
| 06:22.01 | Snake-Eyes | in the future, yea |
| 06:22.33 | Snake-Eyes | but alot want to keep the cash cow going for as long as possible :) |
| 06:23.36 | JT | what's in the future? |
| 06:24.09 | Snake-Eyes | fibre to node/home |
| 06:25.33 | Snake-Eyes | any way I got be off, ill carry on tmr :P |
| 06:27.43 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:27.44 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 06:27.48 | cjk | hi |
| 06:28.06 | JT | well the fibre is what costs real money |
| 06:28.11 | JT | copper sharing is already there |
| 06:28.22 | cjk | im looking for a way to play sound form an eagi scrip in php, any idew how to do this, any hint into the right direction would be enough |
| 06:34.11 | *** join/#asterisk ptblank (n=MURDER1@cpe-76-175-68-73.socal.res.rr.com) |
| 06:37.38 | *** join/#asterisk oej (n=olle@cust225-192.dsl.versadsl.be) |
| 06:39.13 | *** join/#asterisk _omer (i=_omer@9-237-154-202.wol.net.pk) |
| 06:42.06 | *** join/#asterisk DrCron (n=rszasz@c-67-174-231-152.hsd1.ca.comcast.net) |
| 06:42.11 | DrCron | anyone live? |
| 06:43.21 | *** join/#asterisk mkl1525 (n=qwertz@107.205.27.217.static.versanetonline.de) |
| 06:46.32 | mkl1525 | Hi, just had a look with tcpdump which ports are used when I use snom - * in one case ports 16060 and 58466 are used in another 14346 and 51896. Are there any port ranges that the packets use? I'd like to setup some bandwidth rules but don't know which to take - any hints? |
| 06:47.24 | JT | udp 5060 10000-20000 |
| 06:48.06 | *** join/#asterisk bmg505 (n=leon@196.209.181.87) |
| 06:58.40 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 06:59.43 | mkl1525 | JT, thanks |
| 07:03.21 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 07:05.05 | *** join/#asterisk zeeesh (i=zeeesh@202.38.55.125) |
| 07:05.07 | zeeesh | hi |
| 07:05.31 | DrCron | hi |
| 07:06.36 | *** join/#asterisk oej_ (n=olle@cust225-164.dsl.versadsl.be) |
| 07:08.18 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 07:08.31 | *** join/#asterisk saftsack (n=saftsack@pD9E0633B.dip.t-dialin.net) |
| 07:10.06 | *** join/#asterisk Jubei (n=jubei@147.27.46.160) |
| 07:10.30 | Jubei | could somebody tell me where I can find an asterisk startup script for ubuntu? (debian) |
| 07:12.55 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 07:13.36 | Supaplex | Jubei: make up your mind. is it ubuntu or is it debian? |
| 07:14.17 | Greek-Boy | lol |
| 07:14.45 | Jubei | Supaplex: I presume a debian script will work with a little tweaking so.. either |
| 07:14.53 | Supaplex | wrong. |
| 07:14.59 | Supaplex | it might. but don't presume. |
| 07:15.03 | DrCron | do any of you run app_rpt? |
| 07:15.22 | Supaplex | does ubuntu have a supported/matianed package for asterisk? |
| 07:15.40 | Jubei | Supaplex: it does |
| 07:16.00 | JT | DrCron: planning to |
| 07:16.05 | Supaplex | Jubei: what's stopping you from using that? |
| 07:16.47 | DrCron | what support is there for terminating a channel to an audio card |
| 07:16.58 | Jubei | Supaplex: nothing really |
| 07:18.16 | JT | DrCron: sorry, what's the question/ |
| 07:19.09 | DrCron | how would i go about making it so that i can call into asterisk and listen to an audio input and give output to a line out |
| 07:20.08 | DrCron | oh, and extension not a channel |
| 07:20.30 | DrCron | <PROTECTED> |
| 07:21.07 | JT | what has this got to do with app_rpt? |
| 07:21.21 | DrCron | oh, as an alternative to |
| 07:22.28 | DrCron | instead of using app_rpt, use an external control link, and just run the audio and voip side from asterisk |
| 07:26.59 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 07:27.16 | tengulre | hi,all |
| 07:27.52 | tengulre | I want to building a VoIP server for my company, contain 20 pstn lines, which card is suit me? |
| 07:28.17 | tengulre | provide CallCenter services. |
| 07:29.02 | DrCron | it sounds like a digital interface would make sense |
| 07:30.14 | DrCron | is this for incoming lines? or to interface phones |
| 07:34.16 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 07:37.04 | *** join/#asterisk tb0301s (n=sowa@brln-4db11d0c.pool.einsundeins.de) |
| 07:38.54 | *** part/#asterisk tb0301s (n=sowa@brln-4db11d0c.pool.einsundeins.de) |
| 07:43.24 | JT | tengulre: still there? |
| 07:44.37 | *** join/#asterisk qdk_ (n=qdk@213.150.62.32) |
| 07:45.57 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 07:45.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 07:53.38 | Jubei | tengulre: I recommend digium E1 card |
| 07:54.45 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 07:54.55 | JT | how specific |
| 07:58.22 | DrCron | um, have any of you set up the oss module in asterisk? |
| 08:10.31 | tengulre | JT: yes |
| 08:11.46 | *** join/#asterisk zwask (n=erkankol@85.105.4.81) |
| 08:13.16 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 08:14.00 | JT | tengulre: like DrCron said, a digital PRI service such as an E1 would be most appropriate |
| 08:15.37 | DrCron | and much much cheaper |
| 08:15.59 | JT | not always |
| 08:16.02 | JT | but probablty |
| 08:16.22 | DrCron | and using ip phones for the call center |
| 08:16.50 | DrCron | the flexibility you gain should be worth any up-front costs |
| 08:18.47 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 08:21.41 | zwask | i want to monitor online sip users, is there any interface? (socket or an api) |
| 08:26.39 | DrCron | iirc there is |
| 08:26.41 | DrCron | give me a sec |
| 08:26.58 | DrCron | oh, and what do you mean by monitor |
| 08:28.16 | thinko | get a list of them? |
| 08:28.23 | DrCron | or tap the calls |
| 08:29.00 | mkl1525 | (* 1.4.0, snom 300|360) I've got problems using DTMF over CAPI. Using SIP it works. Has anybody ever experienced this problem? |
| 08:29.55 | zwask | yes |
| 08:30.04 | zwask | i want to get list of online users |
| 08:30.28 | zwask | is it possible with an AMI command? |
| 08:37.53 | *** join/#asterisk yidiyuehan (n=yidiyueh@58.185.253.70) |
| 08:37.58 | yidiyuehan | hi everyone, |
| 08:38.16 | yidiyuehan | is it possible to let asterisk server said "please key in the numbers you want to dial" |
| 08:38.54 | yidiyuehan | or "pls enter the number you want to dial and hold while we try to connect your call" |
| 08:39.22 | DrCron | yes |
| 08:39.33 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 08:40.00 | zwask | how? |
| 08:41.11 | yidiyuehan | i reviwed the sounds file in the server and not sure which one is proper for the above need |
| 08:41.35 | DrCron | oh, sorry that was for yidiyuehan |
| 08:41.45 | zwask | yidiyuehan: it is so easy with a playback application. |
| 08:41.47 | DrCron | zwask, i think so but i dont know how |
| 08:42.24 | yidiyuehan | zwask, so i just use playback (pls dial your number and hold for a while), is it fine? |
| 08:42.50 | zwask | alo you should upload an WAV file says that words |
| 08:43.24 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 08:43.35 | yidiyuehan | lol...yes that way i know, WAV file or gsm file? |
| 08:43.48 | zwask | record a WAV file for example named as pls-dial.wav, then add Playback(pls-dial) to exntensions |
| 08:44.06 | zwask | doesnt matter, asterisk supports both of them |
| 08:44.40 | yidiyuehan | ok, i thought there are some existing sound files there ;-), thanks zwask |
| 08:45.50 | zwask | www.asterisk.org download the sound files. there are somethink like this but not same exactly |
| 08:47.16 | *** join/#asterisk nMosila (n=nmosila@218.17.224.130) |
| 08:48.01 | yidiyuehan | ok, i will try so, thanks ;-) |
| 08:51.58 | DrCron | if someone could do me a small favor, and tell my what ops are on the #openbsd chan at the moment it would be most helpfull |
| 08:52.04 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 08:52.58 | *** join/#asterisk friedrich| (n=friedric@e177255145.adsl.alicedsl.de) |
| 08:54.26 | yidiyuehan | Drcron, what do u mean? |
| 08:55.20 | DrCron | if you could join the channel #openbsd and tell me who is currently listed as a channel operator |
| 09:03.07 | *** join/#asterisk Zefk (n=Zefk@wsc-fo.b.astral.ro) |
| 09:09.56 | yidiyuehan | hi, DrCron, well, so far there is no operator there |
| 09:10.51 | DrCron | ok, thanks |
| 09:11.01 | DrCron | i just got the info from another source |
| 09:11.10 | DrCron | thankyou for the attempt though |
| 09:11.39 | yidiyuehan | and anyone knows is it possible to play a congestion tone and then hang up the call? i put exten => s,5,Congestion, exten=>s,6,hangup, but it won't execute the hangup command |
| 09:13.24 | *** join/#asterisk marcan (n=marcanso@160.10.220.165) |
| 09:13.58 | DrCron | if you just want to play the tone use the playback command |
| 09:14.26 | kaldemar | or the playtones application. |
| 09:14.36 | file | yay airport |
| 09:16.02 | yidiyuehan | Drcron, no i want to play the tone to the remote ppl and also hangup the channel he is using |
| 09:16.18 | yidiyuehan | as i want to use callback function and write the scripts in extensions_custom.conf |
| 09:16.48 | yidiyuehan | if i just have exten+>s,6 Hangup, the remote party does not actually hangup and need to wait for a long time |
| 09:20.38 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 09:26.36 | *** join/#asterisk bmg505 (n=leon@196.209.181.87) |
| 09:33.24 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 09:35.45 | hi365 | good morning |
| 09:35.54 | tzafrir | hi |
| 09:36.08 | hi365 | my d channel is down (on a pri) is that a problem on my side or bezeq's side? |
| 09:36.22 | hi365 | good morinig tzafrir! |
| 09:36.28 | hi365 | actualy good afternoon :) |
| 09:37.02 | tzafrir | Chances are that it is on your side, I guess. What card do you have? |
| 09:37.32 | hi365 | sangoam a102d-x |
| 09:37.44 | hi365 | and its driving me nuts. |
| 09:38.01 | tzafrir | Can you set the other port of the card as NT and try connecting the TE port to it? |
| 09:39.05 | hi365 | what is nt/te? |
| 09:47.51 | hi365 | tzafrir ^ ^ ^ ^ ^ |
| 09:47.55 | tzafrir | TE what you connect to Bezeq (Terminal Equipment) |
| 09:48.16 | tzafrir | NT: what you connect a TE to (this is what Bezeq has on their side) |
| 09:50.55 | hi365 | got it. so what signaling pri_net ? |
| 09:59.27 | *** join/#asterisk moranil (n=moranil@122.162.67.129) |
| 09:59.55 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 10:05.19 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 10:05.39 | Assid | heya |
| 10:05.50 | Assid | i keep getting this error : [Apr 16 06:03:57] WARNING[19786]: res_odbc.c:513 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Can't open lib '/usr/lib/odbc/psqlodbc.so' : /usr/lib/odbc/psqlodbc.so: c |
| 10:05.54 | Assid | all i did was upgrade debian to etch |
| 10:07.09 | tzafrir | hi365, yes, pri_net |
| 10:07.43 | tzafrir | Assid, hmm... isn't there native support for pgsql? |
| 10:09.28 | Assid | lemme check |
| 10:09.30 | *** join/#asterisk phalacee (n=Sunforge@202.3.110.33) |
| 10:10.08 | zeeesh | i have 2 sip peers at 2 different machines, peer100 at asterisk "A" and peer 200 at asterisk "B".... how can both make connection by using any sip phone ... i would like to initiate this call by using SIP???? not IAX ... ??? |
| 10:10.08 | Assid | i only see res_odbc.conf |
| 10:10.13 | tzafrir | <PROTECTED> |
| 10:10.13 | Assid | no res_pgsql |
| 10:10.36 | tzafrir | Is this from a package? I can't find it in the files search in http://packages.debian.org/ |
| 10:11.32 | Assid | weird.. it used to be there |
| 10:11.34 | Assid | it used to work |
| 10:12.25 | Assid | lemme cvheck if pghsql works |
| 10:13.42 | Assid | yeah i can use plain old pgsql fine |
| 10:15.31 | DrCron | I've been trying to figure out how to make outbound sip calls to sip URIs over iax, can it be done? |
| 10:16.43 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 10:16.51 | skirmisha | anyone here ? |
| 10:18.11 | skirmisha | huh |
| 10:18.16 | skirmisha | guys where are u |
| 10:18.40 | *** join/#asterisk cancerea (n=apnichat@202.59.74.83) |
| 10:19.36 | DrCron | hi |
| 10:20.28 | skirmisha | at least one is here |
| 10:20.40 | skirmisha | i have 1 question about asterisk dialplan |
| 10:21.14 | skirmisha | i have 3-4 ast servers and i want to make when users register on diff server every other ast to know about this |
| 10:21.41 | skirmisha | the problem i face is that all servers has same config about all users |
| 10:22.12 | skirmisha | but when i try to call user which is registered on other ast it starts looping |
| 10:22.42 | skirmisha | it is not so clear |
| 10:22.58 | *** join/#asterisk Plantseeker (n=Plantsek@83.167.161.28) |
| 10:23.44 | skirmisha | but in short user is in config of server 1 and server 2. At the moment of dialing that user - another user dial from server 1 but user is on server 2 |
| 10:24.02 | skirmisha | so if i do nothing server 1 pickup voicemail of that user |
| 10:24.52 | skirmisha | if i forward the call to server 2 , server 2 starts looping because it has same config as server 1 and config points to call itself |
| 10:25.04 | skirmisha | is it clear so far |
| 10:28.02 | Plantseeker | I have 2 networks : 1 windows server and 1 samba server which use 2 separate ip ranges. I need to know how to allow the windows users to access a printer which is connected to the samba server? |
| 10:28.43 | skirmisha | samba server is on linux |
| 10:28.57 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:29.57 | Plantseeker | sorry I just realised I post the question on the wrong board. |
| 10:34.12 | *** join/#asterisk jazzplyer (n=jazzplye@222-154-246-214.adsl.xtra.co.nz) |
| 10:38.56 | cancerea | hello room |
| 10:39.03 | cancerea | how r u all ? |
| 10:39.20 | creativx | great for being a monday |
| 10:39.32 | cancerea | how r u creativx ? |
| 10:40.20 | creativx | i think i just said? |
| 10:40.20 | creativx | ;) |
| 10:40.34 | zeeesh | chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to '"37201" <sip:38201@xxx.xxx.xxx.x>;tag=as020a2a72'????? |
| 10:42.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:43.18 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 10:43.38 | Winkie | so hey guys, a few days ago I was asking about joining an Asterisk box with a digium PRI card to an Inter-tel system via an E1 crossover |
| 10:43.51 | Winkie | i've made 2 of them, tested them, 1-2 > 4-5 etc and I can't seem to get a link working |
| 10:44.07 | Winkie | neither side seems to recognise any connection and I am unaware of if I would have to set any particular flags |
| 10:44.12 | Winkie | does anyone have any experience in this? |
| 10:48.17 | *** join/#asterisk shadebob (n=chatzill@84.16.31.10) |
| 10:48.34 | shadebob | Hi, someone use a S404 from Soundwin with Asterisk? |
| 10:59.18 | cancerea | i want to ask one thing plz |
| 10:59.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 10:59.42 | cancerea | one iax user account can be used simultaneous from 10 different pcs ? |
| 11:00.20 | cancerea | like in sip.conf call_limit=10 is there is anything same in iax.conf ? |
| 11:02.31 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-186401.home.otenet.gr) |
| 11:03.30 | *** join/#asterisk oej (n=olle@cust225-164.dsl.versadsl.be) |
| 11:05.02 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 11:08.15 | *** join/#asterisk Aurs (n=Aurs@ti500720a080-5902.bb.online.no) |
| 11:10.33 | cancerea | one iax user account can be used simultaneously from 10 different pcs ? |
| 11:10.36 | cancerea | like in sip.conf call_limit=10 is there is anything same in iax.conf ? |
| 11:14.21 | cancerea | plz let me know :-( |
| 11:17.48 | Winkie | well I just figured out my e1 problem so if anyone wants to know in the future make sure your t1e1override has the correct settings! |
| 11:24.14 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 11:24.39 | jm|work | hullo |
| 11:25.01 | *** part/#asterisk zwask (n=erkankol@85.105.4.81) |
| 11:25.16 | jm|work | what would be the best way to catpure a DMTF PIN? WaitExten? |
| 11:26.04 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 11:27.33 | jm|work | hmm Read |
| 11:29.13 | *** join/#asterisk defswork (n=andy@mailgate2.3gcomms.co.uk) |
| 11:31.18 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:34.14 | *** join/#asterisk kje (n=kje@62-99-209-38.c-vzollerg.xdsl-line.inode.at) |
| 11:34.47 | defswork | can I prefix the incoming CLI (from trunk) with a 0 - purely for cosmetic reasons ? |
| 11:37.52 | tzafrir | maybe: Set(CALLERID(number)=0${CALLERID(number)}) |
| 11:38.09 | tzafrir | Or something along those lines but with a correct syntax |
| 11:46.40 | MaartenB | hey everybody |
| 11:47.03 | MaartenB | I have problems with Asterisk, some people find me hard to understand, while I hear the other party fine |
| 11:47.12 | MaartenB | they tell me that the sound virbrates |
| 11:48.11 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 11:48.11 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:49.35 | creativx | set vibrato=false |
| 11:50.25 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
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| 12:00.47 | JT | hmm |
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| 12:10.38 | tuxick | books isn't too clear, and criteria/info about possible use of modems as fxo on linux? |
| 12:10.44 | tuxick | s/and/any/ |
| 12:11.13 | DrukenLPY | tuxick: look on ebay for x100p |
| 12:11.19 | tuxick | cheers |
| 12:11.47 | tuxick | i don't quite understand why you can't use pretty much any 'voice' modem anyway? |
| 12:11.53 | tuxick | looked like driver issue? |
| 12:12.28 | JT | yes |
| 12:12.31 | JT | you can't |
| 12:12.39 | JT | they're generally designed to act as a modem |
| 12:14.03 | cancerea | one iax user account can be used simultaneously from 10 different pcs ? |
| 12:14.05 | cancerea | like in sip.conf call_limit=10 is there is anything same in iax.conf ? |
| 12:14.07 | tuxick | hmm, duplex comes to mind :) |
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| 12:23.20 | MaartenB_ | is there a way to have a usable conversation with a fast connection (5 ms ttl) and enough bandwith (10 Mbit) but with 5 % packetloss? |
| 12:23.34 | tzafrir | tuxick, you can, if you provide a channel driver for it... |
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| 12:27.45 | JT | MaartenB_: what about jitter? |
| 12:27.57 | zeeesh | i have 2 sip peers, peer100 registered at asterisk server "A" and peer200 registered at asterisk server "B" ??? how can both peer100 and peer200 make sip call through xpro ... ??? |
| 12:28.07 | MaartenB_ | JT, asterisk reports 40 ms jitter |
| 12:29.58 | Vec | Does anyone know how to limit the max number of channels/calls on an IAX trunk ? |
| 12:31.07 | Vec | zeesh : setup an IAX trunk between the 2 asterisk servers |
| 12:31.12 | JT | MaartenB_: i'm led to believe ilbc and speex are best for dodgy connections |
| 12:32.01 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 12:32.44 | Vec | MaartenB_ : JT is correct and in my experiance ilbc should work perfectly with 5% packet loss on a 10mbit line |
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| 12:34.37 | zeeesh | <Vec> : i would like to make this session by using SIP |
| 12:34.37 | Vec | zeesh : why ? |
| 12:34.37 | zeeesh | <Vec> is it possible to make session with SIP ... its my presentation |
| 12:34.37 | uwe | hello, i got my hands on an not-new siemens m20, a box that takes a sim card and gives me an Rj 11 output, the problem is that im not sure how to test this! i tried connecting it to a phone, keeps givving busy signal, so does anyone know if it can be connected to fxo modules ? i cant find a way to connect it to asterisk box, given it has no serial interface what so ever !!! |
| 12:35.01 | Vec | zeeesh : sure its possible, but not sure why, what u mean "its my presentation" ?, are peer100 and peer200 behind NAT ? |
| 12:35.34 | zeeesh | not behind the nat |
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| 12:37.29 | MaartenB_ | ilbc is indeed better, thanks JT and Vec |
| 12:38.08 | Vec | MaartenB_ : just interisted, what connection do u have thats 10mbit with that kind of packetloss ? |
| 12:38.57 | MaartenB_ | Vec, http://www.multikabel.nl, dutch cable company |
| 12:39.10 | Vec | MaartenB_ : oh hehe |
| 12:41.25 | hi365 | anyone using a sangom a102d? |
| 12:42.39 | hi365 | mDuff: ur u around? |
| 12:44.15 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 12:47.51 | zeeesh | : chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to '"37201" <sip:38201@202.154.237.5>;tag=as737e458c' |
| 12:50.04 | JT | uwe: you'd connect it to an FXS port |
| 12:51.05 | uwe | JT , thank you very much, ill try it now |
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| 12:53.05 | lugburg | hi, can someone recommend a good oss sip client for linux (supporting alsa)? |
| 12:58.38 | JT | uwe: actually i might be wrong, it probably connects to an FXO port |
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| 13:09.25 | hi365 | anyone using a sangom a102d? |
| 13:09.31 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:09.41 | RoyK | http://callweaver.org/blog/2 |
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| 13:13.10 | uwe | JT, the thing is that if it connects to and fxo module, then i should be able to connect a phone directly to it as well, right |
| 13:13.12 | uwe | ? |
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| 13:15.10 | JT | uwe: yes |
| 13:15.22 | JT | uwe: the fact it gives a busy signal is evidence enough |
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| 13:16.48 | uwe | i see |
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| 13:18.09 | uwe | JT, as far as you know, how does the later communication happen, using modem signalling ? |
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| 13:18.24 | JT | no |
| 13:18.36 | JT | sounds like just a simple analogue GSM gateway |
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| 13:47.57 | Defraz | I am trying to use the the phpagi does this look right for sending a command? $res = $as->Command('show parkedcalls'); |
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| 13:54.38 | nemski | woot |
| 13:54.42 | nemski | got my IP phone |
| 13:55.53 | gambolputty | which one? |
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| 13:56.45 | nemski | Cisco 7940 |
| 13:56.47 | *** part/#asterisk emiquelito (n=evandro@200-155-185-1.static.spo.ifx.net.br) |
| 13:57.05 | nemski | still having issues with registering it with the asterisk server |
| 13:58.43 | nemski | alright I have the line registered, but not voice |
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| 14:03.47 | pigpen | I am sorry, but what is the dam option in the extensions.conf that when you hit * when entering voicemail it will redirect you to an context/app/priority? |
| 14:08.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:10.15 | pigpen | TK or manx told me one time...I forgot. |
| 14:10.37 | drako | where I can find info about fax protocol for asterisk |
| 14:10.46 | *** join/#asterisk NLok (i=Lok@gunlok.rh.rit.edu) |
| 14:10.58 | drako | s/info/good_info |
| 14:11.33 | NLok | good morning |
| 14:12.28 | NLok | is anyone around that can help a newbie out? :/ |
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| 14:15.26 | coppice | drako: info must be absorbed going forwards, but its validity can only be assessed looking back :-) |
| 14:17.01 | anonymouz666 | drako: i wonder if there is someone in here that knows more than coppice about fax |
| 14:17.06 | anonymouz666 | drako: www.soft-switch.org |
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| 14:21.39 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:21.43 | drako | i heard that t.38 is being implemented on 1.4 but what about 1.2? |
| 14:21.48 | Dr-Linux | anybody has any idea about Jitter buffer for IAX |
| 14:21.50 | Dr-Linux | Apr 16 07:05:56 WARNING[1593]: chan_iax2.c:709 jb_warning_output: Resyncing the jb. last_delay 1, this delay 2175, threshold 1020, new offset -2175 |
| 14:21.50 | Dr-Linux | Apr 16 07:06:06 WARNING[1593]: chan_iax2.c:709 jb_warning_output: Resyncing the jb. last_delay -2175, this delay 5680, threshold 5234, new offset -7855 |
| 14:22.03 | blitzrage | drako: features are NOT backported to releases |
| 14:22.29 | Dr-Linux | i asked the same question a number of time, but no solution |
| 14:22.38 | anonymouz666 | blitzrage: not officially |
| 14:22.53 | tzafrir | NLok, there are some, yes. Provided that the said newbie asks a question... |
| 14:22.59 | blitzrage | anonymouz666: right -- sometimes you can find them on some random website |
| 14:24.20 | *** join/#asterisk Rutro (n=Rutro187@wsip-68-96-31-95.om.om.cox.net) |
| 14:24.42 | Dr-Linux | tzafrir: any idea? |
| 14:25.39 | tzafrir | Dr-Linux, not really, except the information that is already in the message: too much jitter? |
| 14:26.52 | Dr-Linux | tzafrir: it appears a few within a call |
| 14:27.06 | Dr-Linux | also something some additional warnings comes up |
| 14:27.17 | Dr-Linux | Apr 16 07:22:51 WARNING[1582]: channel.c:785 channel_find_locked: Avoided initial deadlock for '0x84e4b60', 10 retries! |
| 14:27.18 | Dr-Linux | Apr 16 07:24:06 WARNING[1593]: chan_iax2.c:709 jb_warning_output: Resyncing the jb. last_delay -1575, this delay -500, threshold 1042, new offset -1075 |
| 14:27.53 | Dr-Linux | tzafrir: it's strange i think i'm the only who is using IAX trunk, |
| 14:28.26 | Dr-Linux | tzafrir: i work with different asterisk setups, and i saw the same everywhere |
| 14:28.50 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:28.53 | tzafrir | sorry, I'm not familiar with that... |
| 14:28.55 | Dr-Linux | tzafrir: do you think, it can affect voice qualify? |
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| 14:32.25 | pigpen | Dr-Linux, try not trunking. |
| 14:33.22 | Rutro | Anyone know a good place to find sample configs? |
| 14:34.16 | Dr-Linux | pigpen: then how can i forward the call to the other asteirsk? |
| 14:34.30 | pigpen | it just wont trunk it. |
| 14:34.39 | pigpen | ie: it wont save bandwidth. |
| 14:34.53 | pigpen | taking it to individual calls saved me allot of issues. |
| 14:35.05 | pigpen | and I heard the iax jitter works better this way. |
| 14:35.16 | pigpen | I tried it on iax trunking...and it was not a good day. |
| 14:35.20 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 14:35.29 | pigpen | sorry...have to leave, just set trunking=no |
| 14:35.37 | Dr-Linux | pigpen: you didn't understand, |
| 14:36.51 | Dr-Linux | pigpen: i'm in pakistan, my all pakistan users are connected with local asterisk server, and i've an asterisk in CA, now i how these PK users will call in the US |
| 14:37.06 | Dr-Linux | ofcos i'll connect both Pakistan and US servers with each other ... |
| 14:37.23 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 14:37.31 | Dr-Linux | so i have only two options, 1) Via SIP 2) IAX |
| 14:37.54 | Dr-Linux | so i choose IAX bcoz it consume low bandwith |
| 14:37.59 | Dr-Linux | pigpen: makes sense? |
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| 14:38.42 | putzz | keep going |
| 14:38.51 | blitzrage | IAX doesn't save THAT much bandwidth unless you have a ton of calls I guess |
| 14:39.05 | blitzrage | choosing a codec with less bandwidth like GSM or G.729 will save more |
| 14:40.13 | Dr-Linux | blitzrage: i'm already using g729 |
| 14:40.24 | Dr-Linux | i tried both, SIP and IAX |
| 14:40.35 | blitzrage | just turn off the jitterbuffer then |
| 14:40.40 | blitzrage | that's where the error is coming from it looks like |
| 14:40.54 | Dr-Linux | blitzrage: actually our bandwidth is very worst here in pakistna |
| 14:41.04 | Dr-Linux | i see |
| 14:41.34 | Dr-Linux | blitzrage: but i'd like to know, why we use often this option, jitterbuffer=yes ? |
| 14:42.13 | blitzrage | go search google for what jitter is and what a jitterbuffer does |
| 14:43.37 | Dr-Linux | ok |
| 14:44.05 | Dr-Linux | blitzrage: i'd like to show you |
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| 14:46.19 | blitzrage | yah, but I gotta work |
| 14:46.42 | blitzrage | Dr-Linux: yes, I know what a jitterbuffer is |
| 14:46.48 | BrokenNoze | Anyone found a DPNSS gateway? |
| 14:46.54 | blitzrage | yer having problems with yours, but I don't have time to help you debug it right now |
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| 15:14.59 | `pariah | when i make a call to a zap card which config file will specify what trunk the call should be placed thru? my problem is everytime a ring any zap card it tries to place the call thru trunk 3 and i dont even have tunk 3 defined. |
| 15:15.16 | anonymouz666 | does the festival server speaks too fast? I am not native english speaker... it's fast to me |
| 15:15.16 | BrokenNoze | seriously has no one found a DPNSS gateway to piggyback a legacy PBX??? there must be one out there? |
| 15:16.23 | [TK]D-Fender | `pariah: When you want to know what you dial out of, thats extensions.conf |
| 15:16.43 | `pariah | [TK]D-Fender: ill show you what it is doing real quick |
| 15:18.07 | `pariah | [TK]D-Fender: http://www.pastebin.ca/442893 |
| 15:19.08 | [TK]D-Fender | `pariah: You clearly don't have the context its looking for. The error is BLATANT and staring you in the face. |
| 15:19.18 | [TK]D-Fender | == Starting Zap/2-1 at DID_trunk_3,s,1 failed so falling back to exten 's' |
| 15:19.33 | [TK]D-Fender | You don't HAVE [DID_trunk_3} =. |
| 15:19.38 | `pariah | [TK]D-Fender: That is where I am confused, why is it looking for DID_Trunk_3? |
| 15:19.40 | ManxPower | `pariah: We really can't support FreePBX here |
| 15:19.54 | `pariah | ManxPower: this isn't freepbx |
| 15:20.04 | [TK]D-Fender | `pariah: because thats where you told zapata to send calls to those channels to. |
| 15:20.12 | [TK]D-Fender | ManxPower: Thats the * GUI actually. |
| 15:20.13 | ManxPower | `pariah: What is it? |
| 15:20.25 | ManxPower | [TK]D-Fender: Ah, OK. Evil thing that it is. |
| 15:20.28 | [TK]D-Fender | `pariah: and we do NOT support GUI's in here. |
| 15:20.42 | `pariah | My question is simple, where is asterisk getting told to go to trunk 3? |
| 15:20.47 | *** join/#asterisk hfb (n=hfb@pool-72-67-156-130.lsanca.dsl-w.verizon.net) |
| 15:20.47 | [TK]D-Fender | `pariah: if you don't know how to configure it, go ask in #asterisk-gui |
| 15:20.53 | [TK]D-Fender | `pariah: zapata.conf |
| 15:21.09 | *** join/#asterisk kumbalae (n=sarisdjk@cm115.omega176.maxonline.com.sg) |
| 15:21.15 | ManxPower | `pariah: zapata.conf of course, like all settings for Zap channels |
| 15:21.30 | [TK]D-Fender | `pariah: ... |
| 15:21.31 | `pariah | [TK]D-Fender: thank you that was my only question...not trying to bug you with GUI problems. |
| 15:21.32 | [TK]D-Fender | ~book |
| 15:21.33 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:21.43 | ManxPower | [TK]D-Fender: Does the Asterisk GUI do things as evil as things like FreePBX? |
| 15:21.56 | `pariah | ManxPower: yes it does, thats why im having this problem |
| 15:22.26 | ManxPower | `pariah: there is a channel totally dedicated to Asterisk GUI, oddly enough it is listed in the /topic of this channel -- #asterisk-gui |
| 15:22.38 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 15:22.40 | [TK]D-Fender | ManxPower: yes & no. I haven't personally used it, but from what I can see it isn't as "all encompassing" as FreePBX and you can do a lot more without getting wiped by a config rebuild |
| 15:22.41 | ManxPower | ~zeeek |
| 15:22.44 | jbot | hmm... zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 15:22.44 | `pariah | ManxPower: in my expirence it is a little better than freepbx |
| 15:23.25 | `pariah | ManxPower: I realize this, my question was relatively simple and obviously had nothing to do with the GUI.. |
| 15:23.32 | [TK]D-Fender | ManxPower: But with * GUI you actually have to do SOME work. |
| 15:23.42 | ManxPower | `pariah: other than the gui causing the problem in the first place.... |
| 15:23.57 | [TK]D-Fender | `pariah: Yes it DOEWS have to do with the GUI. Where do you think that context got SET?! |
| 15:24.11 | ManxPower | [TK]D-Fender: does Asterisk GUI support T-1 configs or only analog? |
| 15:24.17 | *** join/#asterisk illsci (n=illsci@evil.hack3rs.org) |
| 15:24.25 | [TK]D-Fender | ManxPower: All the usuals I'm sure. |
| 15:24.28 | `pariah | [TK]D-Fender: my question was what config file tells zap what trunk to go thru |
| 15:25.15 | ManxPower | `pariah: Um there really isn't any such thing as a trunk in asterisk. |
| 15:25.16 | [TK]D-Fender | `pariah: Yeah, and what do you think filled in those values? THE GUI. I'm sure YOU didn't hand-enter it into the file, since you don't even know WHICH ONE. |
| 15:25.29 | `pariah | that was the only question i had, not why is it broken, not all this other garbage, just that plain and simple. and thank you for answering me |
| 15:25.33 | [TK]D-Fender | ManxPower: unload chan_tree.so |
| 15:26.22 | [TK]D-Fender | `pariah: I'd double-check what you put in the GUI in case it wipes out whatever you think you're going to do manually to your config files. |
| 15:27.12 | *** join/#asterisk Crescendo (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:29.46 | `pariah | also, there is nothing about a trunk 3 in my zapata.conf file =\ |
| 15:30.36 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:30.39 | zdrulio | hello all |
| 15:30.43 | [TK]D-Fender | `pariah: That does not come out of thin air. it was there at the last time Zaptel loaded. If its gone then perhaps you changed more stuff and your changes had simply not been applied yet |
| 15:30.59 | ManxPower | `pariah: It would be context=DID_trunk_3 somewhere before a channel => 3 line |
| 15:31.55 | zdrulio | i have Siemens hicom 300h and i want to connect it to asterisk. can anybody help me in this ... ? |
| 15:31.55 | ManxPower | If you still don't find it than perhaps the GUI put it in some other file and then #includes that file or the entries were removed and you forgot to do a reload or restart |
| 15:32.16 | ManxPower | zdrulio: What is a "Siemens hicom 300h" |
| 15:32.24 | *** join/#asterisk quidpro (n=quid@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 15:32.26 | zdrulio | mm :) |
| 15:32.32 | zdrulio | voice central |
| 15:33.05 | ManxPower | Then I doubt it since most PBX specific devices do not speak any protocol in common with Asterisk. |
| 15:33.27 | zdrulio | hicom300h have ISDN PRI and asterisk hae isdn PRI |
| 15:33.50 | ManxPower | zdrulio: So the Hicomm 300h is a PBX with a PRI interface? |
| 15:33.51 | `pariah | http://www.pastebin.ca/442917 cat zapata.conf | grep -v ";" |
| 15:34.11 | ManxPower | `pariah: put the ENTIRE zapata.conf on pastebin. |
| 15:34.34 | ManxPower | zdrulio: I assume it is a E-1 PRI and not a T-1 PRI? |
| 15:34.48 | zdrulio | E-1 |
| 15:35.07 | *** join/#asterisk Maroderr (n=drago@fanatici.net) |
| 15:35.11 | zdrulio | the main idea is |
| 15:35.35 | zdrulio | i have telecom connection is this PRI interface of the siemens |
| 15:35.52 | zdrulio | but i want wo put asterisk between telecom and siemens |
| 15:35.56 | ManxPower | zdrulio: If the PBX support E-1 PRI then there should be no special config required. Asterisk just knows there is a PRI box on the other end of the cable |
| 15:36.02 | CrazyTux | I'm having a DTMF issue where I'm going from softswitch -> asterisk RFC2833, yet its missing digits, any tips/tricks/ideas? |
| 15:36.14 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:36.21 | Maroderr | hi, how i can get call from other phone |
| 15:37.05 | zdrulio | ManxPower: yes but if have a call outsiide It will come to asterisk and than what ? |
| 15:37.20 | zdrulio | how asterisk comunicate with simens |
| 15:37.22 | ManxPower | `pariah: looks to me like you never restarted asterisk to get it to re-read the config file |
| 15:37.41 | ManxPower | Telco PRI <-> Asterisk <-> PBX |
| 15:37.45 | `pariah | i've restarted * many times :-)\ |
| 15:37.55 | zdrulio | ManxPower: yes |
| 15:38.02 | ManxPower | `pariah: reloaded or restrted |
| 15:38.20 | `pariah | restart now from cli |
| 15:38.27 | zdrulio | but how asterisk comunicate to PBX ? |
| 15:38.46 | zdrulio | i call 150 nuber signal come to asterisk and then what ? |
| 15:38.58 | ManxPower | `pariah: Asterisk gets the config information from /etc/asterisk/zapata.conf and any files #include'd from that file. There are no magical channel faeries telling it it's config. |
| 15:39.05 | zdrulio | how asterisk tell to pbx what number to dial ? |
| 15:39.28 | illsci | why does asterisk listen on 2000 2727 4520 5060 |
| 15:39.35 | ManxPower | zdrulio: Dial(Zap/1/150) |
| 15:39.55 | illsci | all i need asterisk to do is iax on 4569 |
| 15:40.11 | `pariah | ManxPower: it is the faeries i know it, those bastards ruin everything i set up =\ |
| 15:40.17 | illsci | to remove these, assuming they are unneeded is it just a matter of disabling certain modules? |
| 15:40.30 | khronos | Anybody been able to build the Zaptel modules on Centos 5.0? |
| 15:40.37 | ManxPower | illsci: 5060 is SIP shgnalling, 2727 is MGCP, etc |
| 15:40.48 | illsci | right so if im not using sip |
| 15:40.52 | illsci | and i dont know what mgcp is |
| 15:40.56 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:40.59 | illsci | i can disable that some how |
| 15:41.08 | ManxPower | 4520 is dundi |
| 15:41.21 | illsci | i think all i need is iax |
| 15:41.31 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:41.31 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:41.43 | ManxPower | illsci: in /etc/asterisk/modules.conf put in a noload line for those channel drivers (chan_sip.so for example) |
| 15:41.46 | *** join/#asterisk af_ (n=getsmart@81-174-45-50.f5.ngi.it) |
| 15:41.55 | illsci | yeah... cool thats what I though.. |
| 15:42.05 | illsci | is there a list of things you dont need |
| 15:42.08 | ManxPower | `pariah: now is the time to ask on #asterisk-gui |
| 15:42.11 | illsci | like a minimalist setup |
| 15:42.18 | ManxPower | illsci: no, since everyone's needs are different. |
| 15:42.44 | ManxPower | illsci: Asterisk is not a PBX. Asterisk a TOOLKIT that allows you to build a PBX. |
| 15:42.45 | illsci | whats the best way to go about removing the modules I don't need if you're only using iax |
| 15:42.54 | `pariah | ManxPower: thanks for trying to help |
| 15:43.06 | Mercestes | Looking at all the modules. |
| 15:43.15 | illsci | yeah i guess i have to do more reading.. |
| 15:43.21 | illsci | i jsut started playing with this |
| 15:43.22 | quidpro | Hmm... is there a way to test within the dialplan if ${callerid(num)} is actually numbers rather than alphabetic? |
| 15:43.24 | ManxPower | well "show modules" and looking for anything starting with chan_ is a start, then put those as noload lines in /etc/asterisk/modules.conf |
| 15:43.35 | illsci | i got a voip number from voicepulse.com to play with asterisk |
| 15:43.52 | [TK]D-Fender | illsci: ... |
| 15:43.53 | ManxPower | illsci: Trying to build a minimal Asterisk system is NOT something to do until you are fully familiar with Asterisk |
| 15:43.53 | [TK]D-Fender | ~book |
| 15:43.55 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 15:44.06 | Mercestes | for i in $(asterisk -rx 'show modules'); do echo noload => $i >> /etc/asterisk/modules.conf |
| 15:44.09 | illsci | yeah I got it |
| 15:44.11 | Mercestes | ;done |
| 15:44.20 | Mercestes | and then comment what you *do* need. |
| 15:44.27 | Mercestes | I wouldn't suggest donig that btw. |
| 15:44.35 | ManxPower | Mercestes: You have found another way guarnteed to make asterisk not work, I see. |
| 15:44.38 | Mercestes | just...fo rexample. |
| 15:44.48 | Mercestes | ManxPower, ;) we all have to have our talents. :) |
| 15:44.53 | illsci | :) |
| 15:45.09 | [TK]D-Fender | Mine are suited to a more ... intimate environment ;) |
| 15:45.09 | Mercestes | you and TK have the "make it work" job pretty much covered so I figured I would expertise in "hose it." |
| 15:46.42 | ManxPower | My boss at one of my clients sent me a link to the Asterisk Appliance and asked if I had heard of it. |
| 15:46.51 | Mercestes | rofl |
| 15:47.14 | *** join/#asterisk MikHell (n=michel@c-71-56-231-165.hsd1.co.comcast.net) |
| 15:47.16 | ManxPower | Of course I've heard about it. I even looked at the specs. The device is FLASH based has a built in router, 5 ethernet ports and up to eight analog ports. |
| 15:47.35 | ManxPower | Of course, I do not manage any offices with analog ports and we don't ever put in an asterisk system with analog ports. |
| 15:47.44 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:47.54 | Mercestes | Thus the analog ports are wasted cash. |
| 15:47.56 | illsci | why not |
| 15:48.01 | [TK]D-Fender | ManxPower: He clearly wants & needs the new Trixbox "Appliance"! |
| 15:48.11 | Mercestes | I am pretty sure the embedded boxes I play with would support a T1 card |
| 15:48.14 | Qwell[] | k/ickban [TK]D-Fender |
| 15:48.18 | Qwell[] | damn typos |
| 15:48.21 | Qwell[] | :p |
| 15:48.22 | Mercestes | rofl |
| 15:48.27 | ManxPower | What he clearly needs to do is to stop second guessing his consultant |
| 15:48.29 | [TK]D-Fender | </sarcasm> |
| 15:48.37 | MikHell | How do I extract a subpart of a pattern? I want to have that when someone dials 01144anydigits it actually does Dial(0044anydigits) |
| 15:48.47 | [TK]D-Fender | Qwell, Ohhhh I'm feeling the love... |
| 15:48.57 | MikHell | How do I extract 44anydigits from the pattern? |
| 15:49.10 | [TK]D-Fender | MikHell: ${EXTEN:3} |
| 15:49.11 | Mercestes | ${2:EXTEN} IIRC |
| 15:49.19 | ManxPower | MikHell: You need to read README.variables in the asterisk docs/ directort |
| 15:49.30 | Mercestes | Hrm. |
| 15:49.31 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:49.32 | [TK]D-Fender | MikHell: nvm, I missed you took a single char out. |
| 15:50.11 | [TK]D-Fender | MikHell: your sample is not indicative of EXACTLY what goes in/out from the before & after |
| 15:50.24 | MikHell | I figured it would be similar to shell methods :) |
| 15:50.32 | ManxPower | MikHell: you were wrong |
| 15:50.43 | Mercestes | extensions.conf being so similar to shell and all. |
| 15:51.04 | ManxPower | exten => _01144.,1,Dial(Zap/1/00${EXTEN:3)) |
| 15:51.07 | MikHell | Well I am in the US but I want to route int'l calls through a European provider |
| 15:51.42 | ManxPower | MikHell: you NEED to read that readme |
| 15:53.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:54.25 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:54.32 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 15:54.58 | [TK]D-Fender | ~variables |
| 15:55.00 | jbot | Docs on how to use variables in * can be found in doc/README.variables in your * source folder or http://www.voip-info.org/wiki-Asterisk+variables |
| 15:55.03 | [TK]D-Fender | MikHell: ^^^^^ |
| 15:57.50 | NLok | hi does anyone have experience connecting Asterisk to VoiceGenie? |
| 15:58.24 | quidpro | Is there a way to test if ${callerid(number)} is actually numbers rather than alphabetic? |
| 15:58.55 | [TK]D-Fender | quidpro: Sure. A small ton of dialplan to parse it char by car... |
| 15:59.12 | Nugget | sounds like two lines of perl in an AGI. :) |
| 15:59.18 | quidpro | Haha, that's what I was thinking... was hoping to do it without lots of loops. :) |
| 15:59.43 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:59.56 | ManxPower | How about exten => 666/_XXX,1,Noop(Satan calling) |
| 16:00.19 | quidpro | Manx: Right, I forgot about that... thanks. |
| 16:00.20 | [TK]D-Fender | Nugget: Far more I'm sure. Have to set up the AGI environment and prep to set a var on return,e etc... |
| 16:00.29 | *** part/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 16:00.38 | Nugget | I type long lines. ;) |
| 16:01.02 | Qwell[] | bah, that'd be half a line of ruby |
| 16:01.24 | Qwell[] | of course, that line would be about 12 pages long with word wrap on, but that isn't the point |
| 16:01.56 | MrChimpy | I'm using spool call files and need to find out exactly how the outbound call I'm making went. I've created a failed extension in the target context. It works - but DIALSTATUS is helpfully set to OutgoingSpoolFailed rather than the more useful output that Dial() would give. Boo. |
| 16:02.05 | [TK]D-Fender | Qwell : Its lunch-time and I'm about to call out to the west-coast to buy a new "point" :) |
| 16:02.12 | *** join/#asterisk Taadow (n=super@66.119.165.82) |
| 16:02.59 | Taadow | Is there an easy way to configure/disable the Message Waiting Indication service? |
| 16:03.10 | ManxPower | MrChimpy: set the destination to Local/extension@context, then put the Dial in the Dialplan |
| 16:03.22 | quidpro | Is it much quicker to do things in the dialplan with app_MYSQL than to compile a custom AGI C script? |
| 16:03.23 | ManxPower | Taadow: add or remove mailbox= from the config file |
| 16:03.25 | [TK]D-Fender | Taadow: Sure... just don't put "mailbox=" into that channel definition. |
| 16:03.29 | MrChimpy | that's a very good plan manx |
| 16:03.35 | MrChimpy | i was about to resort to AGI |
| 16:03.37 | MrChimpy | thanks lots |
| 16:03.46 | ManxPower | MrChimpy: there is info about chan local in the docs/ dir of the asterisk source |
| 16:04.08 | Taadow | Doh! Freepbx puts that in every extension context. Thank you though. :D |
| 16:04.30 | [TK]D-Fender | ~freepbx |
| 16:04.32 | jbot | [freepbx] unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:04.57 | Taadow | Fender. That last line was a statement, not a question. :P |
| 16:05.31 | [TK]D-Fender | Taadow: And mine was a preemptive strike before you get any "smart" questions coming to mind ;) |
| 16:05.37 | Taadow | Heheh |
| 16:05.39 | Taadow | Touche' |
| 16:06.13 | [TK]D-Fender | On touche pas ici, surtout a cause des pleints du harrassment sexuel ;) |
| 16:06.20 | CrazyTux | is there a way I can extract the From: phone number? |
| 16:06.34 | [TK]D-Fender | CrazyTux: PLIERS |
| 16:06.43 | CrazyTux | [TK]D-Fender, literally? |
| 16:06.58 | CrazyTux | [TK]D-Fender, i.e. CALLERIDNUM, EXTEN, ? |
| 16:07.03 | Taadow | Hmm, you win on that one. <--- only speak-o engle' |
| 16:07.23 | *** join/#asterisk izaak (n=izaak@modemcable097.151-202-24.mc.videotron.ca) |
| 16:09.05 | ManxPower | CrazyTux: generally CALLERID is the From: |
| 16:12.00 | *** join/#asterisk sselby (n=sselby@txplano-nat208.dc.xo.com) |
| 16:13.09 | sselby | needing help with simring on a asterisk box. Can anyone help? |
| 16:13.35 | [TK]D-Fender | sselby: That term does not sounds familiar. Elaborate.... |
| 16:14.08 | sselby | simultaneous ring |
| 16:14.35 | ManxPower | Dial(SIP/12324&Zap/6&MGCP/fred) |
| 16:14.42 | ManxPower | There. Wasn't that easy? |
| 16:14.45 | sselby | trying to get an inbound call to ring several lines(cell phone, home phone, desk phone) at one time |
| 16:15.06 | *** join/#asterisk Corydon76-home (i=pink@pdpc/supporter/sustaining/Corydon76-home) |
| 16:15.06 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:15.10 | ManxPower | Remember: ANALOG FXO ports will not work with that. |
| 16:15.20 | [TK]D-Fender | sselby: Only really doable if you're using PRI or a VoIP provider that does |
| 16:15.32 | [TK]D-Fender | sselby: And even then, Cell VM = real trouble |
| 16:16.01 | sselby | yes i am using a wholesale voip provider |
| 16:16.43 | ManxPower | sselby: Then it should work |
| 16:16.56 | *** join/#asterisk shaft|work (n=shaft@txplano-nat208.dc.xo.com) |
| 16:17.07 | Qwell[] | What would be the opposite of "incredible"? |
| 16:17.53 | kumbalae | not incredible |
| 16:17.54 | ManxPower | Qwell[]: www.m-w.com |
| 16:18.06 | Qwell[] | ManxPower: not really the antonym.. |
| 16:18.07 | toot | crap |
| 16:18.24 | toot | run of the mill :) |
| 16:18.32 | Qwell[] | toot: yeah, what's a word for that? :p |
| 16:18.37 | toot | hehe |
| 16:18.44 | toot | standard :P |
| 16:18.49 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 16:18.56 | Qwell[] | something a bit more...meh |
| 16:19.05 | blitzrage | !incredible |
| 16:19.06 | [TK]D-Fender | Qwell "mundane" |
| 16:19.17 | kumbalae | credible is the right term |
| 16:19.17 | Qwell[] | blitzrage: :p |
| 16:19.31 | kumbalae | credible x incredible |
| 16:20.13 | ManxPower | mundane would be good |
| 16:20.28 | *** join/#asterisk Braxus (n=braxus@66.147.214.164) |
| 16:20.30 | illsci | hey what are these from |
| 16:20.33 | illsci | Apr 16 12:19:02 NOTICE[28684]: chan_iax2.c:7525 socket_read: Registration of 'MY_DEVICE_LOGIN' rejected: 'Registration Refused' from: '64.61.93.109' |
| 16:21.10 | kumbalae | illsci: means, the destination is refusing for registration |
| 16:21.18 | kumbalae | or the registrar |
| 16:21.52 | Qwell[] | mediocre wins |
| 16:22.03 | *** part/#asterisk shaft|work (n=shaft@txplano-nat208.dc.xo.com) |
| 16:22.48 | illsci | wow.. there are a ton of loaded modules |
| 16:22.51 | *** join/#asterisk khronos (n=khronos@duchamp.jurying.net) |
| 16:23.06 | illsci | all look really cool though.. |
| 16:23.18 | illsci | this is going to be fun |
| 16:26.53 | ManxPower | illsci: you do not have a [MY_DEVICE_LOGIN] section of sip.conf or the secret= is wrong |
| 16:27.04 | illsci | i don't even want to use sip |
| 16:27.17 | ManxPower | sorry in iax.conf |
| 16:27.23 | illsci | i think the company i got my number for only has iax going on |
| 16:27.54 | illsci | yeah I havent messed with that yet... as far as configuring it... they actually gave me configs... to download but I didnt want to change anything until I read the book |
| 16:28.41 | illsci | hey... do you happen to know what legal issues surround voicemails? |
| 16:29.28 | illsci | like say you had a bunch of voicemails for people and your box gets owned... and they upload them to cnn.com or something |
| 16:30.17 | *** join/#asterisk _VoiceMeUp_Com (n=_VoiceMe@145-27.mc.cite.net) |
| 16:32.18 | *** join/#asterisk clinthome (n=clinthom@12.167.225.79) |
| 16:33.30 | *** join/#asterisk Corydon76-home (i=three@pdpc/supporter/sustaining/Corydon76-home) |
| 16:33.30 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 16:36.12 | Mercestes | <PROTECTED> |
| 16:36.28 | khronos | Anyone had any problems building the 1.4.1 Zaptel modules on Centos 5? |
| 16:36.38 | khronos | The errors I'm getting when I do a make are at: |
| 16:36.43 | Mercestes | PASTEGBIN |
| 16:36.47 | Mercestes | ~pastebin |
| 16:36.51 | jbot | i heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 16:36.51 | khronos | http://209.124.96.10/make_errors.txt |
| 16:36.52 | Mercestes | oh. |
| 16:36.56 | Mercestes | Good job. |
| 16:37.05 | Qwell[] | tzafrir: ^^ |
| 16:37.26 | Qwell[] | khronos: For now, you can disable the building of the xpp modules to (hopefully) avoid that error. |
| 16:37.32 | Qwell[] | khronos: make menuselect |
| 16:37.57 | tzafrir | khronos, I have some proposed patches, but no build system to test them on at the moment |
| 16:38.31 | khronos | Want to use mine? |
| 16:39.07 | Qwell[] | tzafrir: vmware... it seriously, seriously, SERIOUSLY rocks |
| 16:39.38 | Qwell[] | and hell, your stuff is USB, you could actually use it in vmware |
| 16:39.59 | ManxPower | I don't know why they even bother for weather forecasts for north alabama. It's not like they are EVER correct in the spring/fall |
| 16:40.31 | Strom_M | "Today, a high of sixty-twelve, with a thirty per cent chance of dogballs in the early afternoon" |
| 16:40.46 | blitzrage | Qwell[]: I have yet to figure out how to use USB stuff in VMware server.... |
| 16:40.55 | Qwell[] | blitzrage: plug it in, and it'll yell at you |
| 16:41.02 | blitzrage | Qwell[]: hrmm... I've never seen that |
| 16:41.11 | Qwell[] | don't let the host do anything with it |
| 16:41.14 | ManxPower | Strom_M: the forecast for today changed 10 degrees between last night and now |
| 16:41.21 | blitzrage | Qwell[]: hrmm... how do I do that? :) |
| 16:41.31 | blitzrage | VMware running on Linux |
| 16:41.34 | Qwell[] | dunno, it always "Just Works" for me.. at least in workstation |
| 16:41.40 | Qwell[] | never tried in server/player |
| 16:41.46 | blitzrage | ah |
| 16:41.52 | blitzrage | maybe workstation is better at that |
| 16:42.01 | izaak | hi all, I'm new to *. I'm configuring an outgoing dialplan for 5 phones (each a SIP channel) where each SIP channel has an associated IAX channel. i need some advice to make the configuration short (ie, i don't want to repeat the whole dialplan for each SIP/IAX pair, I'd rather use variables) |
| 16:42.12 | Qwell[] | but, when I plug something in, vmware pops up a window, and is like "Click here to disable this device on your host system" |
| 16:42.23 | *** join/#asterisk eltech (i=G00Ds@ool-457c94a3.dyn.optonline.net) |
| 16:42.26 | Qwell[] | then, when you click it, the guest OS detects it |
| 16:42.30 | blitzrage | Qwell: hrmmm... now you have me curious to try it again |
| 16:42.36 | Qwell[] | it's pretty slick |
| 16:43.01 | Qwell[] | I use the Motorola Phonetools in Windows, and that happens when I plug my phone in |
| 16:43.07 | blitzrage | it'd be wicked if my webcam would get detected in Windows |
| 16:43.07 | Mercestes | izaak, Depends on what your extensions are and waht your phone peer names are, but assuming your extensions are 4001, 4002, 4003, 4004, 4005, and your phone peer name smatch the extensions: |
| 16:43.18 | blitzrage | neato! |
| 16:43.27 | Mercestes | izaak, Then you can exten => _400x,1,Dial(SIP/${EXTEN},180) |
| 16:43.31 | Qwell[] | blitzrage: chan_cellphone! |
| 16:43.42 | blitzrage | Qwell[]: exactly!!! |
| 16:43.47 | Qwell[] | it's the hotness |
| 16:43.48 | blitzrage | and I can run a SIP client on it too |
| 16:43.56 | Mercestes | izaak, Or more appropriately, exten => _400[1-5],1,Dial(STIP,${EXTEN},180) |
| 16:44.05 | blitzrage | now I need a bluetooth USB adapter for my computer though |
| 16:44.14 | Qwell[] | blitzrage: They can be had for $20ish |
| 16:44.15 | blitzrage | don't think this laptop has bluetooth... |
| 16:44.22 | Qwell[] | meh, I'm an idiot |
| 16:44.23 | blitzrage | yah, I imagine they are fairly inexpensive |
| 16:44.29 | blitzrage | ok :) |
| 16:44.32 | Qwell[] | I decided it wasn't worth the $15 to add bluetooth to my laptop |
| 16:44.43 | Qwell[] | would've been internal and everything, but no... |
| 16:44.47 | izaak | Mercestes: thanks, i think i will use that sort of pattern for incoming. but for outgoing, where i need to dial different IAX channels depending on which SIP channel? |
| 16:44.50 | Qwell[] | now if I want it, it's like $40, heh |
| 16:45.02 | blitzrage | heh |
| 16:45.06 | blitzrage | that was kinda dumb :) |
| 16:45.09 | Qwell[] | totally |
| 16:45.15 | Qwell[] | oh well |
| 16:45.16 | Mercestes | izaak, Are you trying to recreate static lines for you rphones using IAX? |
| 16:45.24 | Qwell[] | I also only got the 40gb hd... |
| 16:45.34 | Qwell[] | which was also an incredibly dumb thing to do |
| 16:45.46 | Qwell[] | now the $40 I saved is gonna cost me like $120 =x |
| 16:46.15 | izaak | Mercestes: i'm not sure what you mean. my VOIP provider has given me an IAX channel per DID for incoming and outgoing. for incoming i understand how to use just one context. but i'm confused about outgoing. |
| 16:46.25 | izaak | Mercestes: each phone on my network has its own DID |
| 16:46.38 | Mercestes | izaak, Yo ushould be able to dial out dynamically. |
| 16:48.38 | NLok | hi does anyone have experience connecting Asterisk to VoiceGenie? I am having problem transfering a call. |
| 16:51.04 | [TK]D-Fender | NLok: just describe the problem you're having. Pastbin CLI output with SIP debug enabled where applicable. |
| 16:51.06 | [TK]D-Fender | ~pb |
| 16:51.12 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 16:52.36 | ManxPower | urpmi /tmp/mpeg4ip-1.5.0.1/player/src/video_sdl.cpp:280: undefined reference to `XMoveWindow' |
| 16:52.39 | ManxPower | drat |
| 16:52.51 | Qwell[] | ManxPower: silly manduck |
| 16:53.13 | *** join/#asterisk zaide (n=zaide@script-kiddy.fr) |
| 16:53.21 | tzafrir | khronos, http://lists.digium.com/pipermail/asterisk-users/2007-April/184970.html |
| 16:53.44 | zaide | hi |
| 16:55.29 | zaide | anyone have an idea to select my second VOIP provider when the first provider is out or timeout (or others errors)? i haven't found how to do it in my extension.conf |
| 16:56.29 | Qwell[] | zaide: after Dial, check the value of the DIALSTATUS variable |
| 17:00.19 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 17:00.23 | LeddyHM | Any thoughts on an os migration through vmware for ~30 users? "It will be fine, should be ok, audio is too choppy, no worries it's fine" for about a week? |
| 17:02.39 | *** join/#asterisk PupenoR (n=pupeno@200.123.183.91) |
| 17:02.41 | NLok | sipphoneA calls sipphoneB(VoiceGenie) through asterisk. Then sipphoneB transfers sipphoneA to sipphoneC through asterisk. |
| 17:02.52 | NLok | I get a maximum retries error when sipphoneB tries to do the transfer. |
| 17:03.29 | ManxPower | NLok: you will want to set aside the rest of the day to work on this problem |
| 17:03.35 | *** part/#asterisk zaide (n=zaide@script-kiddy.fr) |
| 17:03.41 | ManxPower | maximum retries exceeded means asterisk did not get a response from the destination device\ |
| 17:05.44 | *** join/#asterisk dasenjo (n=be185cd4@acuario.unicauca.edu.co) |
| 17:05.56 | *** join/#asterisk HKhan (n=hkhan@sekhmet.hamzahkhan.com) |
| 17:07.23 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 17:07.49 | dasenjo | Hi, I'm trying to compile zaptel in the latest stable kernel (2.6.20-7), but got errors on zaphfc. There is no linux/config.h but autoconf.h instead. I made a symbolic link, but got segfault ... can you help me? |
| 17:08.49 | ManxPower | dasenjo: do you have the kernel SOURCE installed? |
| 17:10.42 | NLok | ManxPower, if sipphoneA calls sipphoneB without going through asterisk, then sipphoneB is able to transfer the call to sipphoneC without a problem |
| 17:13.29 | ManxPower | NLok: That does not change the fact of the error. |
| 17:13.31 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:13.44 | syzygyBSD | Morning beautiful |
| 17:14.11 | ManxPower | are all three phones and asterisk on the same local network with no firwall or NAT between any of the devices |
| 17:14.41 | NLok | they are all on the same local network |
| 17:14.53 | *** join/#asterisk maxdoubt (n=mackstou@169.198.254.6) |
| 17:17.15 | ManxPower | NLok: do the IP addresses look correct in the error message? |
| 17:17.51 | anonymouz666 | it's raining a lot. the world is coming to an end. |
| 17:18.13 | dasenjo | ManxPower: yes .. I installed the kernel form sources |
| 17:19.05 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-178-65.buckeyecom.net) |
| 17:19.09 | ManxPower | dasenjo: if you cd to /usr/src/linux and run the command "make config" does it work? |
| 17:19.43 | dasenjo | yes, I did a make menuconfig |
| 17:19.50 | syzygyBSD | anonymouz666: well just move from earth to mars like we did from venus to earth after we fucked that up |
| 17:19.56 | ManxPower | dasenjo: you forgot to mention what version of zaptel |
| 17:20.06 | dasenjo | I compiled with debian kernel-package without error .. |
| 17:20.12 | anonymouz666 | lol |
| 17:20.19 | maxdoubt | i'm trying to setup gnugk as a method for NAT traversal. I can call clients on the net, but they can't call the client behind the firewall, any clues? |
| 17:20.31 | dasenjo | zaptel 1.2.16 |
| 17:20.49 | *** join/#asterisk Shoeb (n=chatzill@64.34.69.9) |
| 17:21.07 | ManxPower | dasenjo: and what exactly is the error message? |
| 17:21.41 | syzygyBSD | wow, I really should have gotten a count of all the empty directories I am deleting before I started |
| 17:21.50 | ManxPower | dasenjo: also make sure kernel-headers is installed (I don't know what Debian calls the package) |
| 17:21.55 | Hmmhesays | why is that? |
| 17:21.59 | toot | anyone know of a technical writer knocking about? :) |
| 17:22.03 | dasenjo | without th symbolic link, that could not find file linux/config.h, with the symbolic link: |
| 17:22.11 | syzygyBSD | already in the tens of thousands... |
| 17:22.15 | Shoeb | Beginner: Hello, I've configured AsteriskNOW. And now I can't seem to be able to login to it using Xlite, it says "Registration eror - not found" |
| 17:22.30 | ManxPower | dasenjo: you should not need the symbolic link |
| 17:22.42 | maxdoubt | is there a gnugk irc chat room? |
| 17:22.45 | ManxPower | Shoeb: ask on #asterisknow |
| 17:23.00 | *** join/#asterisk JunK-Y (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 17:23.02 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 17:23.09 | Shoeb | ManxPower: It's a dead channel. Besides, it's not a GUI problem as I'd guess. |
| 17:23.16 | JunK-Y | moooo |
| 17:23.26 | ManxPower | Shoeb: did you configure asterisk by hand? |
| 17:23.27 | dasenjo | ManxPower, but what can I do to get rid of it? |
| 17:23.39 | Shoeb | ManxPower: Ofcourse, yes. |
| 17:23.44 | syzygyBSD | dasenjo: uninstall? |
| 17:24.02 | *** part/#asterisk HKhan (n=hkhan@sekhmet.hamzahkhan.com) |
| 17:24.14 | ManxPower | dasenjo: remove the sybolic link, put the error message on pastebin.ca I cannot help you futher until you do that |
| 17:24.19 | syzygyBSD | what version of linux do you have |
| 17:24.21 | ber_ | I am getting one way audio caused by RTP stream trying to terminate at a RFC1918 ip address for my home box behind a nat |
| 17:24.32 | ManxPower | Shoeb: you need to be looking at the CLI |
| 17:24.37 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 17:24.44 | ber_ | what do I change to tell asterisk to translate the RFC space into its external address |
| 17:24.45 | syzygyBSD | dasenjo: and what kernel version... you should install the source for that kernel version |
| 17:24.47 | ManxPower | ber_: check the mailing lists archives |
| 17:24.49 | *** join/#asterisk slmnhq (n=salmanh@denali.asti-usa.com) |
| 17:24.53 | Shoeb | ManxPower: Thanks. One minute. |
| 17:25.09 | ber_ | i have checked it for NAT in sip.conf |
| 17:25.12 | ber_ | that doesnt seem to work |
| 17:25.13 | dasenjo | 2.6.20-7 from a tbz2 from kernel.org |
| 17:25.16 | ber_ | i set nat-yes |
| 17:25.19 | NLok | ManxPower, I think you are right, the address in the error is 4d241449717203de1a273e813f17cb24@192.168.0.10. |
| 17:25.22 | ManxPower | ber_: that depends on if Asterisk is behind the NAT or if the client is behind nat |
| 17:25.28 | dasenjo | I compiled from sources |
| 17:25.33 | ber_ | just client asterisk |
| 17:25.36 | slmnhq | Greetings all.. I'm trying to connect all the pieces of a puzzle related to Voip |
| 17:25.39 | ber_ | my main asterisk box is no nat |
| 17:25.46 | Shoeb | ManxPower: What would I be looking for? |
| 17:25.59 | slmnhq | Is this the right place to be asking newbie questions? |
| 17:26.08 | slmnhq | (regarding Asterisk) |
| 17:26.09 | ManxPower | 12:24:51) ManxPower: ber_: that depends on if Asterisk is behind the NAT or if the client is behind nat |
| 17:26.30 | ManxPower | slmnhq: No place. You need to read The Good Book |
| 17:26.32 | ManxPower | ~book |
| 17:26.50 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:26.50 | ManxPower | ber_: "client asterisk"? |
| 17:27.03 | ber_ | i have an asterisk box not behind nat and an asterisk box behind NAT |
| 17:27.12 | slmnhq | Ok thanks... I have already ordered that book on Amazon |
| 17:27.18 | dasenjo | ManxPower: http://pastebin.ca/443133 |
| 17:27.23 | ManxPower | ber_: Are you using SIP between the two boxes? |
| 17:27.38 | ber_ | Audio from the NATed asterisk box is fine, audio to the natted asterisk box doesnt work because its trying to hit 192.168.1.104 |
| 17:27.41 | ber_ | yes i am |
| 17:28.00 | ber_ | i tried configuring NAT=yes in sip.conf thinking that would translate the RTP from reserved to the external IP |
| 17:28.07 | ber_ | but it doesnt appear to do so |
| 17:28.32 | ManxPower | dasenjo: you need to install glibc-devel |
| 17:28.57 | ManxPower | ber_: on the public IP asterisk you need nat=yes in the sip.conf section for the remote asterisk box. |
| 17:29.07 | ber_ | ok |
| 17:29.17 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:29.23 | ber_ | i will re-review the config, i believe I did that already but i will check |
| 17:29.42 | ManxPower | then on the asterisk behind nat you need in sip.conf [general] you need localnet= and externip= set, then you need to portforward UDP5060 and UDP 10000-20000 to the asterisk box |
| 17:29.55 | slmnhq | is it ok if I come back here with more specific questions based on things I've read in the book? |
| 17:30.15 | ManxPower | slmnhq: yes. Ask questions here if you have questions about parts of the book too. |
| 17:30.33 | dasenjo | ManxPower, for debian, is the package called libc6-dev? |
| 17:30.44 | ManxPower | dasenjo: I don't know what it is in Debian |
| 17:31.25 | ManxPower | [root@fs-1 # rpm -qif /usr/include/linux/config.h |
| 17:31.28 | ManxPower | Name : glibc-devel Relocations: (not relocatable) |
| 17:31.58 | ber_ | can externip= dns name? |
| 17:32.04 | ManxPower | ber_: no |
| 17:32.09 | ber_ | heh |
| 17:32.16 | ber_ | its on DHCP so its extern IP can change |
| 17:32.25 | ber_ | thats good enough to get a hack working for now though, i apprecaite your help |
| 17:32.27 | ManxPower | there is an extern option for name, but I strongly doubt it will work for you |
| 17:32.46 | dasenjo | in debian, that file is in the linux-kernel-headers package |
| 17:33.09 | MikHell | I am not sure if this is a good place to ask, but how do I find the best DID and termination providers to use with asterisk? |
| 17:33.09 | ManxPower | didn't I tell you to install the kernel headers? |
| 17:33.17 | NLok | ManxPower, the ip address in the error is actually correct if the variable before @ is just the session name |
| 17:33.26 | ManxPower | MikHell: they all suck |
| 17:33.43 | dasenjo | does not the tar.bz2 contains the headers? |
| 17:33.47 | MikHell | ManxPower: I am sure :) But some suck less than others, don't they? :D |
| 17:34.18 | ManxPower | dasenjo: I have no idea. I've not needed to build a kernel in YEARS |
| 17:34.27 | *** join/#asterisk voipman (i=distorti@junipero.3sheep.com) |
| 17:34.53 | dasenjo | the headers for my old kernel are installed, but I had to compile, I think source should contains all |
| 17:34.59 | ManxPower | dasenjo: you need to figure out how to get the correct /usr/include/linux |
| 17:35.08 | [TK]D-Fender | ber_: externhost=my.dynamic.dns.host externrefresh=60 |
| 17:35.23 | ManxPower | dasenjo: remove everything for your old kernel |
| 17:35.27 | *** join/#asterisk bmd (n=bmd@72.54.252.34) |
| 17:35.40 | ManxPower | [TK]D-Fender: and soon as there is a DNS failure everything stops working, right? |
| 17:35.57 | [TK]D-Fender | ManxPower: We support only the BEST problems here ;) |
| 17:36.23 | [TK]D-Fender | ManxPower: or... "shit look very good... when compared to CRAP!"\ |
| 17:37.03 | *** join/#asterisk AMUG (n=junky@modemcable140.185-70-69.mc.videotron.ca) |
| 17:37.06 | syzygyBSD | how long should it take to delete a bunch of folders? |
| 17:37.55 | dasenjo | I can't ... linux-kernel-headers is a build-essential dependency :( |
| 17:37.56 | toot | not that long unless you had a space in the rm -rf . / ;) |
| 17:38.16 | syzygyBSD | find /data/*/backupfiles/ -depth -type d -empty -exec rmdir {} \; |
| 17:38.31 | syzygyBSD | but I think there are close to a million |
| 17:38.47 | ManxPower | dasenjo: it sucks to be you |
| 17:38.56 | dasenjo | :p |
| 17:39.32 | ManxPower | syzygyBSD: I always put a -v on my automated rms |
| 17:39.44 | dasenjo | but .. I have a new uname, compiling should not use the old kernel includes ... I think ... |
| 17:40.14 | syzygyBSD | cuz printing out a million folders that no longer exist helps me how? |
| 17:40.43 | ManxPower | also I was not aware that zaphfc was included in zaptel |
| 17:40.45 | syzygyBSD | I understand seeing the progress, but meh... |
| 17:41.04 | ManxPower | syzygyBSD: it helps you make sure you are not deleting the whole system |
| 17:41.57 | [TK]D-Fender | ManxPower: Just curious, what are your 2 preferred distro's for * servers (in order). And anything special about your means of implementing either? |
| 17:42.19 | Shoeb | What does "No matching peer found" mean? |
| 17:42.41 | ManxPower | [TK]D-Fender: I only have 1. Mandrake/Mandriva for any server or workstation. |
| 17:42.44 | [TK]D-Fender | Shoeb: translation, "who the ^%#$ are you?!" |
| 17:42.51 | Shoeb | lol |
| 17:43.03 | Shoeb | Gotcha. |
| 17:43.31 | syzygyBSD | ManxPower: true, also why I am only using rmdir instead of rm -rf |
| 17:43.35 | ManxPower | you need to create a symlink in /lib/modules because of an oddity in the mandrake kernel make file (the add "custom" to the version number. |
| 17:43.46 | [TK]D-Fender | ManxPower: Anything in particular that it facilitates vs CentOS/RH/(screw FC)? |
| 17:43.49 | ber_ | Manx, thanks for your help. That did it |
| 17:43.58 | ber_ | and TK for the externhost command |
| 17:44.04 | [TK]D-Fender | ber_: Quite welcome |
| 17:44.17 | Dr-Linux | anybody is using MultiVoIP (Multitech) gateways with asteirsk? |
| 17:44.21 | ber_ | one of these days I'll answer some questions on this channel instead of just taking pointers :) |
| 17:44.26 | ManxPower | [TK]D-Fender: urpmi resolves RPM dependencies for you and, if correctly set up, installs the required package dependencies |
| 17:45.03 | [TK]D-Fender | ManxPower: I've been well served by YUM so far, but my needs/usage hasn't been esoteric |
| 17:45.05 | ber_ | anyone have just a stock asterisk binary build w/OS out there. Kinda like trixbox but without that web interface/etc |
| 17:45.07 | Defraz | with the phpagi is there a way to just get the calls in queue and what queues they are in. I am using the command: show queues |
| 17:45.22 | Defraz | but it seems like there is a lot of info I don't need in there and it is in a format that isn't usefull. |
| 17:45.29 | [TK]D-Fender | ber_: Pick a common distro and just compile *. Packaged * = ASS |
| 17:45.38 | ber_ | yeah thats what i do now |
| 17:45.45 | ManxPower | [TK]D-Fender: if Yum had been around when I started installing servers I might have used RH |
| 17:45.47 | Dr-Linux | lol |
| 17:45.51 | [TK]D-Fender | ber_: it adds 10-15 mins to CentOS install for me. |
| 17:46.02 | Dr-Linux | ManxPower: RHEL? |
| 17:46.15 | [TK]D-Fender | ManxPower: Yeah, they "inherited" a decent tool. |
| 17:46.17 | ManxPower | Dr-Linux: ?? |
| 17:46.27 | [TK]D-Fender | ManxPower: Ever tried the IAXModem/HylaFAX combo? |
| 17:46.27 | ber_ | i always had issues with the zaptel drivers |
| 17:46.35 | ManxPower | [TK]D-Fender: nope. |
| 17:47.01 | Dr-Linux | Yum works with RHEL? :S |
| 17:47.17 | [TK]D-Fender | ManxPower: I'm getting ready to. Far more robust than rxfax/txfax, and looking to be the only sane way right now.... should be 1.4 compatible as well |
| 17:47.20 | ManxPower | Dr-Linux: I have no idea what YUM works with. I don't use it. |
| 17:47.25 | ber_ | hylaxfax is great |
| 17:47.29 | [TK]D-Fender | Dr-Linux: Yes. |
| 17:47.32 | Mercestes | Yum works????! |
| 17:47.40 | [TK]D-Fender | Mercestes: do YOU?! ;) |
| 17:47.42 | ber_ | i used it for a 8 line email2fax gateway |
| 17:47.47 | ManxPower | [TK]D-Fender: I've had VERY good luck with rxfax |
| 17:47.52 | ber_ | 99.99% reliable |
| 17:47.57 | Mercestes | lol |
| 17:48.14 | ManxPower | [TK]D-Fender: most recent releases of spandsp solved the few lingering problems |
| 17:48.19 | [TK]D-Fender | ManxPower: Have you gotten it to work on 1.2.17 or close version release? |
| 17:48.36 | ManxPower | [TK]D-Fender: Yes. |
| 17:48.40 | [TK]D-Fender | ManxPower: I've had a serious bitch of a time with it. Its crash out on the few times I could even get it to compile |
| 17:49.07 | [TK]D-Fender | ManxPower: If you could spare me a few minutes some night this week to point me in the right direction it'd be greatly appreciated :) |
| 17:49.13 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 17:49.47 | ManxPower | sorry, 1.2.14 is what the main system is on |
| 17:49.47 | Mercestes | Not if I can avoi dit |
| 17:50.26 | ManxPower | 1.2.15 on the other system. |
| 17:50.39 | ManxPower | just remmeber spandsp is not dependent on the asterisk version |
| 17:51.04 | Mercestes | OH |
| 17:51.12 | Mercestes | [TK]D-Fender, I used the iaxmodem hylafax combo |
| 17:51.38 | Mercestes | scary thought, huh? |
| 17:52.49 | Shoeb | I have no idea what I'm doing wrong, when the user/ext is added properly in asterisk and I'm still getting this problem. |
| 17:52.55 | Shoeb | Of no matching peer found. |
| 17:53.13 | [TK]D-Fender | Shoeb: Your phone is not using the right user. |
| 17:53.44 | [TK]D-Fender | Shoeb: pastebin the full CLI output of your error and the pile of lines preceeding it |
| 17:53.46 | PupenoR | Do you know of any other tool to test a PBX/IVR thas SIPp? |
| 17:53.54 | Shoeb | Ok |
| 17:53.57 | *** join/#asterisk type0 (n=type0@216-67-9-25-cdsl-rb2.cwc.acsalaska.net) |
| 17:54.00 | type0 | wassup all |
| 17:54.39 | Mercestes | Wassup? |
| 17:55.03 | Mercestes | PupenoR, a telephone. |
| 17:55.38 | *** join/#asterisk MrParity (n=patrick@dslb-088-076-214-090.pools.arcor-ip.net) |
| 17:55.42 | MrParity | hi ho :-) |
| 17:55.54 | PupenoR | Mercestes: that is a very poor way of testing. Good for developing, but not for testing the setup with a mega calls |
| 17:57.00 | Shoeb | [TK]D-Fender: It worked, I just restarted Asterisk. But now it's giving a registration timeout! |
| 17:57.06 | Shoeb | Request timeout i mean. |
| 17:57.06 | ManxPower | PupenoR: use asterisk to generate calls |
| 17:57.31 | ManxPower | Shoeb: STOP! Provide the requested info or go away. |
| 17:57.58 | PupenoR | ManxPower: running two asterisk in the same computer would be problematic, but I may be able to solve that. Other than that, how do I make Asterisk generate calls, play with the PBX and give me some usefull status of it? |
| 17:58.23 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:58.23 | ManxPower | PupenoR: why would you need to run two instances of Asterisk? |
| 17:59.05 | PupenoR | ManxPower: actually, I want to test an Asterisk module (written in C), so I'd like the tested asterisk to be as isolated as possible (to be able to monitor for memory leaks in my own code). |
| 17:59.09 | Shoeb | ManxPower: After you didn't answer my last question directly to you "<Shoeb>ManxPower: What would I be looking for?" I thought you're off the case. [TK]D-Fender has been helping me since yesterday, and I don't see him freaking out at all? |
| 17:59.17 | ManxPower | PupenoR: like all free tools you would have to write your own testing scripts |
| 18:00.02 | PupenoR | ManxPower: then SIPp is a better tool. |
| 18:00.02 | ManxPower | PupenoR: for your needs, yes. |
| 18:00.02 | *** join/#asterisk dawizard (n=dawizard@mimas.xios.be) |
| 18:00.02 | syzygyBSD | Shoeb: that is because you can't see him |
| 18:00.08 | ManxPower | (12:53:15) [TK]D-Fender: Shoeb: pastebin the full CLI output of your error and the pile of lines preceeding it |
| 18:00.16 | [TK]D-Fender | I'm invisible so long as no one is looking at me! |
| 18:00.20 | PupenoR | ManxPower: ok, thank you. I haven't had to do any testing like this in a long time, so I was checking if anybody know about anything better than sipp. |
| 18:00.30 | ManxPower | Shoeb: your problem should take 5 mins to solve once TK gets the info he requested. |
| 18:01.01 | ManxPower | PupenoR: there is something called "sipsak" or something like that. I don't know if it will be helpful to you |
| 18:02.27 | syzygyBSD | maybe he did pastebin it, but just didn't give the link because we didn't ask? |
| 18:02.27 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:02.28 | MrParity | i have a problem with early dial (gxp2000 + v1.2). i'm not able to type more than 3 digits. i alway get the error 503. does anyoen have an idea how to fix? |
| 18:02.28 | *** join/#asterisk qufk (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 18:02.28 | [TK]D-Fender | MrParity: Stop using early dial. |
| 18:02.28 | MrParity | do i need so set an special option to use this feature? |
| 18:02.28 | [TK]D-Fender | MrParity: This is not really supported in * |
| 18:02.32 | MrParity | [TK]D-Fender: *g* |
| 18:03.08 | MrParity | [TK]D-Fender: ok, thats the reason for this error, thanks :-) |
| 18:03.19 | ManxPower | MrParity: Early Dial is when the phone sends each digit to it's SIP server and the server either responds with a "incomplete number" packet or the server dials the call |
| 18:04.52 | MrParity | ManxPower: i know, but i think early dial is a nice feature and i want to have it :) i didn't know that there is no support in asterisk |
| 18:05.32 | [TK]D-Fender | MrParity: And I want $1,000,000 so nobody is leaving happy today. Live with it :) |
| 18:05.56 | Nivex | [TK]D-Fender: Don't forget the pony! |
| 18:05.58 | syzygyBSD | [TK]D-Fender: well, I was going to give that to you today, but since everyone else isn't happy, I guess I won't |
| 18:06.07 | [TK]D-Fender | Nivex: I can BUY it after if I want it! |
| 18:06.17 | [TK]D-Fender | I'm spending enough this week as it is... |
| 18:06.30 | syzygyBSD | hookers and booze? |
| 18:07.22 | [TK]D-Fender | syzygyBSD: Sword and guitar ($820 +/-, and $900 respectively) |
| 18:07.47 | syzygyBSD | what sword? |
| 18:08.19 | [TK]D-Fender | syzygyBSD: http://www.casiberia.com/product_details.asp?id=SH1018 |
| 18:09.01 | syzygyBSD | don't cut your arm off, any body part come to think of it... |
| 18:09.17 | MrParity | [TK]D-Fender, ManxPower : ok, i've know decreased the keypad timeout - it's not a perfect solution, but it's ok :-) thanks. |
| 18:09.58 | ManxPower | MrParity: BTW, real phones have support for an internal dialplan. |
| 18:10.13 | [TK]D-Fender | syzygyBSD: My Oni Forge Bushi - http://aocomputing.net/bushi |
| 18:10.18 | ManxPower | And by "real phone" I mean "almost anything except Grandstream" |
| 18:10.26 | MrParity | *g* |
| 18:10.50 | MrParity | ManxPower: maybe i will test an snom phone next days |
| 18:11.32 | syzygyBSD | pretty |
| 18:11.56 | ManxPower | I'm a fan of Polycoms |
| 18:14.17 | [TK]D-Fender | syzygyBSD: I'm sure its nicer than the Oriole I just ordered, but the Oriole has a more distinct color and because of tsuka/balde length and the thinner blade on it it'll handle like lightning, and the arc much improved for me. |
| 18:14.50 | bulle | [TK]D-Fender: you bought a sword ? |
| 18:15.04 | [TK]D-Fender | bulle: Another one, yes |
| 18:15.20 | syzygyBSD | the ratio looks a bit odd for me, 2/3 blade, 1/3 handle |
| 18:15.22 | bulle | [TK]D-Fender: oh, japanese one |
| 18:15.29 | syzygyBSD | but I don't really know anything... |
| 18:15.31 | bulle | syzygyBSD: its for two handed use |
| 18:15.56 | [TK]D-Fender | bulle: I have 1 true blade (as linked), a Paul Chen Gorin Iaito ( dulled), and have just ordered the Oriole linked above. |
| 18:16.01 | syzygyBSD | well, ya, but 13" is plenty |
| 18:16.25 | bulle | [TK]D-Fender: http://www.albion-swords.com/swords/johnsson/sword-museum-svante.htm |
| 18:16.35 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:16.56 | syzygyBSD | bulle: dead link? |
| 18:17.04 | bulle | syzygyBSD: dead link ? |
| 18:17.11 | [TK]D-Fender | syzygyBSD: Yeah, you can see the scal difference vs my Bushi, but I am 6'2-3" and longer arms. The added tsuka length is a GOOD thing, and the center of balance will make it a FAST blade. |
| 18:17.32 | bulle | syzygyBSD: the link works for me |
| 18:17.48 | syzygyBSD | can't lookup albion-swords.. dns issues it looks like... |
| 18:18.02 | [TK]D-Fender | bulle: 4lbs? bit of a beast... |
| 18:18.06 | bulle | [TK]D-Fender: i have 0 clues when it comes to japanese katanas, i only train with european ones |
| 18:18.20 | bulle | [TK]D-Fender: yes, its a pretty heavy thingy, one of the heavier |
| 18:18.33 | bulle | [TK]D-Fender: mind, its for use in armour, so you dont use it as you use, say a katana |
| 18:18.46 | [TK]D-Fender | bulle: I currently practice Katori Shinto, and am looking for an MJER Iaido dojo next |
| 18:19.18 | [TK]D-Fender | bulle: No... a katana could poke a nasty woulnd through a joint just the same :) |
| 18:19.50 | bulle | [TK]D-Fender: problem is, there are no joints to chop trough, when the person is standing up, and facing your way |
| 18:20.03 | [TK]D-Fender | bulle: That would do better on heavy leather I would guess and has a greater length. |
| 18:20.30 | bulle | [TK]D-Fender: im one of those arma loons ( www.therama.org ) |
| 18:20.32 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 18:20.34 | [TK]D-Fender | bulle: armour head to toe? humans bleed from all sorts of differnt points :) |
| 18:20.38 | syzygyBSD | I'll just stick with guns |
| 18:21.05 | [TK]D-Fender | syzygyBSD: I used to work at a firearms importer/exporter.... they lost their appeal to me as a weapon long ago. |
| 18:21.13 | falz | oi. I just began using queues this weekend. somehow, a few of the phones in them don't show up in the queue if I do "show queue foo" |
| 18:21.20 | CunningPike | Perhaps, given current events, we could reduce the drooling over weapons for a bit, eh? |
| 18:21.25 | LeddyHM | guns are for sissies |
| 18:21.31 | bulle | swords are for men! |
| 18:21.40 | CunningPike | http://www.cnn.com/2007/US/04/16/vtech.shooting/index.html |
| 18:21.46 | LeddyHM | I was just informed about a virginia tech shooting |
| 18:21.51 | LeddyHM | hah |
| 18:21.51 | [TK]D-Fender | LeddyHM: Agreed. Too "clumsy" and random.... :) |
| 18:21.51 | LeddyHM | ;) |
| 18:22.11 | [TK]D-Fender | bulle: that sword = $$$ |
| 18:23.00 | [TK]D-Fender | bulle: Here's a known American smith with a great variety of Western European arms - http://www.angustrimdirect.com/swordhome.htm |
| 18:23.15 | *** join/#asterisk Waverly360 (n=irc@209.12.249.243) |
| 18:24.02 | type0 | i saw the shooting earlier this morning |
| 18:24.09 | type0 | 22 people implies a massive motivation |
| 18:24.49 | *** join/#asterisk rogerz (i=jon13@cpe-24-195-144-82.nycap.res.rr.com) |
| 18:24.58 | LeddyHM | yup |
| 18:25.52 | bulle | [TK]D-Fender: well, i dont live in america, and i know the guy that does the historical research and design of the albion swords, but thanks anyway |
| 18:26.02 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-51-85.dsl.irvnca.pacbell.net) |
| 18:26.27 | Waverly360 | Hey guys, is it possible to configure a polycom phone to dial another polycom phone directly without using asterisk? |
| 18:26.43 | BSD_Tech | http://www.foxnews.com/story/0,2933,266310,00.html |
| 18:27.09 | mcab | Waverly360: enable url dialing, then press "new call", then "name" and dial the IP of the other phone |
| 18:27.15 | *** join/#asterisk ppyy (n=ppyy@222.66.125.206) |
| 18:27.27 | ManxPower | Waverly360: If you are not using Asterisk then by definition the question does not belong here |
| 18:27.32 | type0 | anyone know of a SNMP platform which uses Adobe Flex? |
| 18:28.02 | Waverly360 | ManxPower: I was just curious. I don't even care how to do it, I was just wondering if it was possible. |
| 18:28.08 | Waverly360 | mcab: Thanks. |
| 18:28.09 | MrChimpy | hmm. i have my spool call file calling a Local/extension@context and doing the dial based on dialplan variables I set in the spool file. The Dial has a G() option so I get a context for the called side - but I need those dialplan variables from the caller. Any way to achieve that? |
| 18:30.47 | ManxPower | MrChimpy: prefix the variable names with two underscores |
| 18:30.56 | ManxPower | you need to read README.variables |
| 18:31.12 | MrChimpy | lots to remember :) |
| 18:32.39 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 18:33.41 | Waverly360 | Does anyone know of a PSTN gateway device that has a sip server built-in to it, so that I can use it to connect to an asterisk server OR have polycoms connect directly to it? |
| 18:35.19 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 18:35.46 | *** join/#asterisk elriah (i=elriah@adsl-074-164-217-189.sip.bhm.bellsouth.net) |
| 18:35.56 | elriah | Hi all. Are MeetMe conference ID's logged anywhere? |
| 18:36.28 | ManxPower | Waverly360: That would be called "Asterisk" |
| 18:37.24 | Waverly360 | ManxPower: I'm looking for a device I can buy that does it all. |
| 18:37.36 | Vec | When using fastagi, if I want to return something back to the caller, does it have to be done by the AGI script or can the AGI script return the variable back to the asterisk dialplan ? |
| 18:38.01 | Supaplex | you can set a var |
| 18:38.23 | Vec | Supaplex : set a channel var in the agi ? |
| 18:39.22 | Supaplex | you only said 'something'. I don't know about setting a channel var. (no idea what that really is atm) |
| 18:39.47 | Supaplex | poke the asterisk wiki how to use vars in dialplans and set/get from agis. |
| 18:41.44 | Vec | Supaplex : I know how to do both just did not make the connection, thanks! |
| 18:41.49 | toot | yay - we finally managed to get a website together :) |
| 18:41.49 | toot | ~ |
| 18:42.00 | toot | eek - soz wrong window :( |
| 18:42.00 | toot | meh |
| 18:43.32 | Supaplex | toot: yay localhost. ;) |
| 18:43.52 | elriah | Hi all. Are MeetMe conference numbers's logged anywhere? |
| 18:43.58 | elriah | is there any meetme logging at all? |
| 18:43.59 | toot | <html><body><p>hello world!</p></body></html> :P |
| 18:44.38 | ManxPower | elriah: should be easy enough to extract the info from /var/log/asterisk/messages |
| 18:44.40 | Supaplex | hehe |
| 18:45.08 | PupenoR | Anyone using SIPp with authentification? it seems it fails to calculate the length of the content in the invite with auth, any ideas? |
| 18:47.06 | *** join/#asterisk izaak (n=izaak@modemcable097.151-202-24.mc.videotron.ca) |
| 18:47.20 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 18:48.07 | izaak | does anyone know how i can force a caller id in a SIP channel configuration? the documentation says 'callerid' is only used when the callerid is "unavailable". |
| 18:51.20 | *** join/#asterisk saftsack (n=saftsack@pD9E0633B.dip.t-dialin.net) |
| 18:51.21 | MrChimpy | manx |
| 18:51.25 | MrChimpy | works now, thanks loads |
| 18:53.10 | _VoiceMeUp_Com | Apr 16 14:52:48 WARNING[15726]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8713fd8 (len 893) to 712.3.12.1.123:0 returned -1: Invalid argument |
| 18:53.13 | _VoiceMeUp_Com | waht this mean ? |
| 18:54.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-136-120.red.bezeqint.net) |
| 18:54.01 | *** join/#asterisk sysreq (n=sysreq@219.47-ppp.3menatwork.com) |
| 18:54.10 | Hmmhesays | ugh, why do we have to keep killing each other |
| 18:54.12 | BSD_Tech | man the httpd serve in asterisk needs fast cgi |
| 18:54.25 | BSD_Tech | because is human nature |
| 18:54.34 | BSD_Tech | man is self destructive |
| 18:54.36 | syzygyBSD | Hmmhesays: if it makes you feel any better you can only die once... unless they bring you back to life |
| 18:54.47 | Hmmhesays | 32 dead at the va tech shooting, so far |
| 18:54.50 | syzygyBSD | So I guess that is why, because people don't stay dead |
| 18:55.32 | BSD_Tech | just give evey man woman and teenager a gun and call it fair play |
| 18:55.40 | BSD_Tech | ope n the boarders |
| 18:55.48 | Supaplex | haha |
| 18:56.09 | CrazyTux | Hmmhesays, recently? |
| 18:56.13 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 18:56.21 | Mercestes | CrazyTux, uh, today |
| 18:56.25 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:56.25 | Hmmhesays | I think it was an hour ago |
| 18:56.37 | BSD_Tech | http://www.foxnews.com/story/0,2933,266310,00.html |
| 18:56.38 | blitzrage | what's a TV? |
| 18:56.41 | CrazyTux | I never watch TV, or rather I need to read CNN more often. |
| 18:56.58 | blitzrage | you shouldn't read American news |
| 18:57.18 | BSD_Tech | 90 american news = propiganda |
| 18:57.21 | Hmmhesays | ahh 8am my time |
| 18:57.22 | BSD_Tech | or death |
| 18:57.26 | Hmmhesays | 32 dead 21 wounded so far |
| 18:57.28 | syzygyBSD | fox news.. isn't that an oxy moron |
| 18:57.43 | Mercestes | syzygyBSD, Clever. |
| 18:57.48 | BSD_Tech | America = to the Middle East |
| 18:57.54 | BSD_Tech | self destructive |
| 18:58.11 | Hmmhesays | the same kid is on fox news and cnn |
| 18:58.28 | Mercestes | That's not allowed in public anymore. |
| 18:58.36 | syzygyBSD | oh, I would have loved to come up with it, but I didn't. I think Rupert Murdoch did |
| 18:58.52 | Hmmhesays | i have a baseball bat with the 10 commandments on it |
| 18:58.55 | [TK]D-Fender | Hmmhesays: Guitar I'm trying out again tonight and likely to buy - http://cachepe.zzounds.com/media/quality,85/brand,zzounds/RGT42BP-be975ffc90113a7700bda886daf61865.jpg |
| 18:58.59 | Hmmhesays | you're going down b1tch |
| 18:59.08 | Mercestes | We have replaced "freedom of speech" with "Thou shalt not be offensive to ANYONE unless your an athiest, a faggot or an extremist, then it's ok." |
| 18:59.23 | BSD_Tech | lol |
| 18:59.33 | Hmmhesays | [TK]D-Fender: what is that? |
| 18:59.33 | syzygyBSD | hmm, what if you are all 3? |
| 18:59.49 | BSD_Tech | down with the goverments |
| 19:00.01 | BSD_Tech | its time for people to take back control |
| 19:00.08 | BSD_Tech | and start fresh |
| 19:00.13 | [TK]D-Fender | Hmmhesays: Ibanez RGT42DX-IBT . 1-piece though-neck double-locking 24 fret... |
| 19:00.18 | BSD_Tech | make drugs legal and tax them |
| 19:00.18 | syzygyBSD | I am an extremist to conseratives nowadays |
| 19:00.23 | Mercestes | I'd rather talk about D-Fender's guitar. |
| 19:00.32 | Hmmhesays | [TK]D-Fender: nice |
| 19:00.33 | Mercestes | Ibanez you say? nice. |
| 19:00.43 | BSD_Tech | make all sex acts legal |
| 19:00.46 | Hmmhesays | I'm sticking with schecters lately |
| 19:00.50 | Mercestes | lol |
| 19:00.52 | BSD_Tech | and taxable |
| 19:00.55 | Mercestes | ROFL |
| 19:00.58 | syzygyBSD | lol BSD_Tech |
| 19:01.01 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 19:01.04 | [TK]D-Fender | Mercestes: I was preparing to shell out around $1200 for a Carvin custom job... so the savings are being pumped into the blade I just ordered ;) |
| 19:01.09 | BSD_Tech | and shoot billgates |
| 19:01.20 | Mercestes | All sex acts are legal. Just....gah, do we have to have an advocacy group around it? |
| 19:01.25 | syzygyBSD | I think sex with fluffy bunnies shouldn't be legal, who will protect them from being exploited? |
| 19:01.26 | BSD_Tech | and burn MS to the ground |
| 19:01.31 | [TK]D-Fender | Hmmhesays: Schecter is pretty much an ESP... definately shred guitars.. |
| 19:01.44 | [TK]D-Fender | Hmmhesays: Not terribly unlike this one I'm looking at :) |
| 19:01.45 | Mercestes | I mean hell, I like to beat my wife and screw her in public, maybe I shoudl whine about my rights. |
| 19:01.55 | syzygyBSD | Mercestes: the only legal way to have sex in idaho is missionary style, |
| 19:01.57 | [TK]D-Fender | Hmmhesays: I'm just completely finished with bolt-on necks.... |
| 19:02.14 | Mercestes | syzygyBSD, Not really enforceable. (and, ps: BORING) |
| 19:02.19 | Hmmhesays | yeah you get way better sustain from neck through or even set neck |
| 19:02.22 | [TK]D-Fender | syzygyBSD: Yeah..... treat your sheep right! |
| 19:02.25 | Mercestes | [TK]D-Fender, It is a nice blade. |
| 19:02.34 | Hmmhesays | although with bolt on if you have string through body you get good sustain |
| 19:02.36 | Mercestes | save the bunnies. |
| 19:02.38 | Supaplex | Mercestes: not boring. syzygyBSD has 143 tickets for it. ;) |
| 19:02.44 | Mercestes | Supaplex, ROFLMAO |
| 19:02.47 | [TK]D-Fender | Mercestes: it is pretty, and I'm betting on the reviews since I can't expect to find it locally. |
| 19:02.48 | syzygyBSD | lol |
| 19:02.52 | BSD_Tech | Montana where the men are men and the women are to |
| 19:02.59 | [TK]D-Fender | ... |
| 19:03.06 | syzygyBSD | got it wrong BSD_Tech |
| 19:03.16 | BSD_Tech | ? |
| 19:03.17 | Supaplex | so BSD_Tech is a ... man I guess. |
| 19:03.17 | syzygyBSD | Montana, where the men are men and the sheep are scared |
| 19:03.35 | *** join/#asterisk hansin321 (n=hansin32@c-67-190-142-147.hsd1.co.comcast.net) |
| 19:03.39 | Mercestes | Montana, where men are men and women are polyamorous sheep |
| 19:03.57 | BSD_Tech | but make everythign legal and taxable |
| 19:04.17 | BSD_Tech | and anyone in office can only serve 1 term |
| 19:04.22 | Mercestes | California! Where men are women and women are men and women are women doing women and men are women doing men and men are women doing women (etc.) |
| 19:04.22 | syzygyBSD | makes sense, the only true freedom is economic freedome |
| 19:04.26 | BSD_Tech | then a new person has to step in |
| 19:04.39 | syzygyBSD | if you are rich enough you can do anything which is different from now because.... hmm, nm |
| 19:04.45 | Hmmhesays | haha the cnn anchor just said "good hustle" to one of the field reporters |
| 19:04.58 | Mercestes | Smack his ass! Smack his ass! |
| 19:05.01 | Hmmhesays | I like the schecter variaty |
| 19:05.06 | Hmmhesays | *variety even |
| 19:05.35 | MrChimpy | ok, last dull question. what's the "official" dialplan method of do-nothing-until-other-end-hangs-up? |
| 19:05.39 | BSD_Tech | I will try anything sexual and if I like it go back for more |
| 19:05.47 | Mercestes | MrChimpy, macro-stdexten |
| 19:06.06 | Mercestes | MrChimpy, For outbout just Dial(tech/number) with no timeout and no finishing code. |
| 19:06.27 | Mercestes | MrChimpy, You could even call a Congestion() after 1800 seconds if you want to be really nice, and then a hangup just to clean up after yourself. |
| 19:06.31 | MrChimpy | i can't do that as I have to provide dialplan for both legs when dialling with G() |
| 19:06.45 | Mercestes | then the last half should work. |
| 19:07.35 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 19:08.35 | _VoiceMeUp_Com | Apr 16 14:52:48 WARNING[15726]: chan_sip.c:1084 __sip_xmit: sip_xmit of 0x8713fd8 (len 893) to 712.3.12.1.123:0 returned -1: Invalid argument |
| 19:08.37 | _VoiceMeUp_Com | agaiun |
| 19:08.41 | syzygyBSD | MrChimpy: wait(10000000000)? |
| 19:08.43 | _VoiceMeUp_Com | is that from a1.4 ? |
| 19:08.50 | _VoiceMeUp_Com | i think ill block 1.4 from our networks |
| 19:09.01 | *** join/#asterisk amer (n=ame@58.27.148.181) |
| 19:09.10 | MrChimpy | BSD: that's what I have now :) |
| 19:09.32 | BSD_Tech | ? |
| 19:09.38 | syzygyBSD | me not you tech |
| 19:09.49 | BSD_Tech | ahh ok |
| 19:10.08 | Mercestes | _VoiceMeUp_Com, You should. that's what all the othe rcarriers are doing. |
| 19:10.19 | amer | I am using Asterisk as an SBC (SIP trunking) but I am having some problems, will appreciate if someone can help me out |
| 19:10.37 | Mercestes | amer: SBC = session boarder control? |
| 19:10.41 | amer | yes |
| 19:10.49 | Mercestes | s/boarder/border/ |
| 19:11.01 | Mercestes | amer: interesting. |
| 19:11.02 | Supaplex | brodem ;) |
| 19:11.09 | Hmmhesays | yahoo |
| 19:11.21 | amer | ok so now the problem I am facing is codec negotiation |
| 19:11.37 | Mercestes | amer: well, since there is no such thing as sip trunking, and asterisk is not an SBC device, I'm guessing your having problems? |
| 19:12.00 | Mercestes | amer: disallow=all allow=codecs you want |
| 19:12.12 | amer | I have 2 interface on a server, one interface faces SONUS and the 2nd is on public IP facing carriers |
| 19:12.22 | amer | hear me out |
| 19:12.35 | JT | [TK]D-Fender: you about? |
| 19:12.47 | [TK]D-Fender | JT, yup |
| 19:13.05 | *** join/#asterisk dasenjo (n=be185fc7@acuario.unicauca.edu.co) |
| 19:13.30 | Mercestes | amer: Good way to do it |
| 19:13.32 | amer | now if a call comes in with g.729 as preffered codec asterisk negotiates g.729 and sends sonus a list of codecs which is not the same as it gets orignally |
| 19:13.50 | amer | e.g it send g.723...g.729 |
| 19:13.58 | *** join/#asterisk savas^ (n=chatzill@88.243.3.4) |
| 19:14.16 | amer | sonus selects g.723 as its the first in the order |
| 19:15.27 | amer | now on one side I have 729 and on the other 723, I dont want any transcoding on asterisk so the call drops as soon as someone picks up |
| 19:16.23 | savas^ | hi folks, i have a problem with option messages |
| 19:16.28 | savas^ | OPTIONS sip:ffdventures;user=phone SIP/2.0 |
| 19:16.53 | savas^ | in here destination's domain name is missed |
| 19:16.56 | amer | how can I make asterisk send the same preference of codecs as it gets from the orignator |
| 19:17.30 | savas^ | is this a correct message according to asterisk? |
| 19:19.00 | amer | we have been using asterisk as a SBC for the last 2.5 years, aorund 30 million minutes per month :) |
| 19:19.23 | amer | the recent problem is because of whole sale carriers who dont have a single preffered codec |
| 19:20.30 | amer | can anyone help? |
| 19:20.31 | *** join/#asterisk CVirus (n=GoD@82.201.174.72) |
| 19:20.36 | CVirus | Mercestes: there ? |
| 19:20.50 | Mercestes | CVirus, I am now |
| 19:21.20 | amer | Mercestes: any thoughts? |
| 19:21.26 | CVirus | Mercestes: great ..... remember my problem ? I could make direct SIP to SIP calls ... but when I use an asterisk server in the middle .. the call starts but no voice goes through |
| 19:21.28 | Mercestes | Amer: Ok, so you don't know what codec is incoming I'm guessing? |
| 19:21.49 | Mercestes | CVirus, Yea, but I remember none of the troubleshooting steps |
| 19:22.03 | CVirus | Mercestes: lemme re-paste you my sip.conf and extensions.conf |
| 19:22.11 | Mercestes | oh please. :) |
| 19:22.51 | amer | yes sir |
| 19:23.32 | Mercestes | amer: I'm pretty sure you can do disallow=all and allow=g711u, g729 and I'm fairly certain asterisk will "prefer" to use whatever codec is already in place and try to avoid transcoding. |
| 19:24.00 | syzygyBSD | lazy software |
| 19:24.16 | amer | I have tried this, most of the time on both legs different codecs are negotiated |
| 19:24.24 | Mercestes | amer: As long as you specifically allow that codec to the sonus. It sounds like to me that your not specifying allowed codecs so it's doing random stuff. Try specifying disallow=all and allow= codecs you want to allow." |
| 19:24.26 | Supaplex | syzygyBSD: lazy software comes from _______ |
| 19:24.32 | Mercestes | lazy programmers. |
| 19:24.37 | syzygyBSD | effiecint coding? |
| 19:25.22 | BSD_Tech | no such thing |
| 19:25.28 | BSD_Tech | in the real world |
| 19:25.52 | BSD_Tech | thats why apps become bloated with piss poor coding |
| 19:25.56 | syzygyBSD | what does reality tv have to do with this? |
| 19:28.29 | amer | sonus-------(disallow all, allow 723,729)asterisk(disallow all,allow 723,729)----------(729,723)carrier |
| 19:29.03 | amer | sonus---723----asterisk-----729-----carrier |
| 19:29.11 | amer | u see my problem |
| 19:31.17 | *** join/#asterisk beehive (n=michael@pool-71-126-181-126.washdc.fios.verizon.net) |
| 19:31.19 | Mercestes | amer: Your prefering 723, carrier sends you 729, and your stuff picks 723. |
| 19:31.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:31.37 | Mercestes | BSD_Tech, you are such a positive and enlightening person, btw. |
| 19:32.26 | syzygyBSD | just a realist |
| 19:33.16 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-3600465b97da6c9c) |
| 19:35.16 | *** join/#asterisk CVirus (n=GoD@82.201.174.72) |
| 19:35.45 | Mercestes | realist == pessimist |
| 19:36.34 | *** join/#asterisk tmjb (n=tmjb@212.200.241.47) |
| 19:36.55 | JT | if that's what you think reality is... |
| 19:37.01 | *** join/#asterisk ccesario (n=ccesario@201-0-124-218.dsl.telesp.net.br) |
| 19:39.13 | Hmmhesays | anyone ever done any serial communications in perl? |
| 19:39.14 | [TK]D-Fender | No.. a pessimist just thinks everything is bad... a realist KNOWS this to be true ;) |
| 19:39.57 | Mercestes | lol |
| 19:41.24 | techie | wwssdsd |
| 19:41.30 | *** part/#asterisk savas^ (n=chatzill@88.243.3.4) |
| 19:42.46 | Hmmhesays | no one? |
| 19:43.10 | hrmphh | hmm; try #perl |
| 19:43.26 | [TK]D-Fender | Hmmhesays: Code is googlable |
| 19:44.36 | bkruse | [TK]D-Fender: yep |
| 19:44.41 | Hmmhesays | yeah I'm not sure what to google |
| 19:45.02 | [TK]D-Fender | "perl serial communications sample" |
| 19:45.59 | Hmmhesays | if you google that with quotes you get nothing |
| 19:46.09 | Hmmhesays | I have been googling but haven't come up with much |
| 19:46.22 | bkruse | "words specific words" + some words to search |
| 19:46.58 | *** join/#asterisk cspot (i=cspot@ip68-109-8-207.pn.at.cox.net) |
| 19:47.32 | JT | win 74 |
| 19:49.31 | syzygyBSD | Hmmhesays: try google.com/codesearch |
| 19:49.55 | *** join/#asterisk ctooley (n=ctooley@adsl-68-95-129-178.dsl.rcsntx.swbell.net) |
| 19:50.08 | ctooley | Is there a way to get the call-id of an outbound call? |
| 19:50.41 | ctooley | I need to record the call-id that Asterisk sets so I can track it down on the remote end. |
| 19:51.23 | Mercestes | ctooley: it's delivered as the src number, in your CDRS, or as ${EXTEN} in your incoming context. |
| 19:51.36 | Mercestes | ctooley: it would be what your matching under exten => |
| 19:51.55 | syzygyBSD | uh.. it is in ${CALLERID[number]} i believe |
| 19:51.56 | syzygyBSD | or the CDR for the call |
| 19:51.56 | syzygyBSD | or the ... |
| 19:53.09 | ctooley | Not the number that is dialed, the SIP Call-ID |
| 19:53.59 | Mercestes | ctooley: you are correct, ${EXTEN} would be number dialed, but ${CALLERID(number)} would be the src number |
| 19:54.32 | *** join/#asterisk gerphimum (n=trekkie@207.190.62.44) |
| 19:54.40 | ctooley | Mercestes, and I can get that from a bridged all after hangup? |
| 19:55.12 | ctooley | I know how to get the A (inbound) leg's Call-ID, but not the B (ensuing outbound) leg's Call-ID |
| 19:55.47 | amer | is there a variable that the incoming codec value? |
| 19:56.11 | amer | that has * |
| 19:56.42 | amer | if incoming_codec=729 goto 10 |
| 19:57.06 | amer | dial sip_proxy (allow only 729) |
| 19:57.34 | ctooley | 59f66b341c05820a7c7946fe4fe127df@71.42.115.242 |
| 19:57.48 | ctooley | something like that is going to be the outbound SIP Call-ID |
| 19:57.53 | *** join/#asterisk mrdigital (n=mrdigita@207-172-229-15.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 20:01.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:03.21 | *** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/sustaining/Corydon76-home) |
| 20:03.21 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 20:03.45 | *** join/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 20:03.52 | mattwj2005 | hi guys :) |
| 20:04.17 | mattwj2005 | I know it is best to run the same version of asterisk with the same version of zaptel |
| 20:04.32 | mattwj2005 | what do I do....newest version of zaptel isn't out yet |
| 20:04.48 | BSD_Tech | ok asterisk +gui up on 6.2 |
| 20:04.59 | BSD_Tech | now to workon other things |
| 20:05.14 | BSD_Tech | I wish the asterisk httpd supported fastcgi |
| 20:06.44 | syzygyBSD | cgi? or agi? |
| 20:06.57 | JT | cgi |
| 20:07.00 | JT | he said httpd |
| 20:07.17 | BSD_Tech | cgi |
| 20:08.28 | *** join/#asterisk massctrl (n=mlkj@d51A54F17.access.telenet.be) |
| 20:11.23 | *** join/#asterisk toot (n=toot@84.19.255.123) |
| 20:11.39 | *** join/#asterisk CVirus (n=GoD@82.201.174.72) |
| 20:12.03 | CVirus | Mercestes: I'm terribly sorry ... something was wrong with my connection ... here's the conf files http://rafb.net/p/kGlmLg71.html |
| 20:12.49 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:14.12 | CVirus | Mercestes: are you still there ? |
| 20:15.42 | massctrl | yowz, i'm not familiar with voip/asterisk in general. i'm wondering if it's feasable to receive incoming calls with asterisk and route these call based upon the choices they make in the menu to a pool of people who are waiting with a voip client on their computer or incoming calls... |
| 20:16.15 | massctrl | it's for some sort of a helpdesk setup |
| 20:16.35 | Strom_M | massctrl: yes |
| 20:16.36 | [TK]D-Fender | massctrl: Certainly |
| 20:16.46 | Strom_M | IVR menus and queues are what you should look into |
| 20:17.34 | toot | do i get in trouble if i mention a commerical product (link) in here? :) |
| 20:17.39 | massctrl | ok and the one recieving the voip call, does he need to be logged in with a voip client to the asterisk box? |
| 20:18.02 | Mercestes | CVirus, Sorry, world crumbled. |
| 20:18.22 | CVirus | Mercestes: no problem |
| 20:18.36 | toot | not as a sales pitch as such, more out of interest (none of you would need our product i suspect) :) |
| 20:18.43 | Mercestes | ctooley, Oh, the caller id of the outbound leg? Why would the outbound leg even transmit you callerID? |
| 20:19.51 | ctooley | Mercestes, I don't care about callER id. I know what that is. I need to know what the Call-ID is. |
| 20:20.08 | ctooley | I'm not looking for any kind of caller id information |
| 20:20.19 | ctooley | Call-ID is a unique identifier for a call |
| 20:20.30 | *** join/#asterisk saftsack (n=saftsack@pD9E0633B.dip.t-dialin.net) |
| 20:20.49 | Mercestes | ctooley, oh. Hrm. That's a sip debug thing. I don' tthink that's output anywyere. |
| 20:21.27 | Mercestes | CVirus, this isn't helping me much. |
| 20:21.39 | ctooley | Mercestes, yeah, but it's got to "be" somewhere. I can use the GetSIPHeader to get the inbound call-id. But, that's not going to help on the outbound leg. |
| 20:22.06 | Mercestes | ctooley, Might be a sourcecode patch thing. |
| 20:23.25 | ctooley | Mercestes, yeah, I think I'm going to have to do that... was hoping to avoid doing another hack to chan_sip |
| 20:23.26 | ctooley | I've already got several |
| 20:24.18 | falz | anyone have issues with grandstrema phones and the aux/voice/qos vlan setting? I can get it working on a) brand new budgetone 200's, and VERY old firmware budgetone 100's, but none of the newest firmware for handytone's or budgetone 100's |
| 20:24.26 | *** join/#asterisk lee_is_me (n=chatzill@12-227-176-77.client.mchsi.com) |
| 20:24.44 | *** join/#asterisk CVirus (n=GoD@82.201.174.72) |
| 20:25.10 | CVirus | Mercestes: I'm afraid I can't follow you .... My laptop is freezing every 10 minutes .. it seems it's dying :-( |
| 20:25.33 | lee_is_me | Hi all, what is the preferred way to send a caller to an extension's voicemail? Custom extension with parsing? |
| 20:25.46 | toot | well - at any rate - our website is at www.tigercube.net/components/ - not officially launched and will be opening up some interesting parts of our code etc - i welcome all feedback and q's :) |
| 20:26.01 | falz | lee_is_me: just set up an extension that goes direct to voicemail |
| 20:26.28 | falz | we just prefix the normal extention with it, like an asterisk for example |
| 20:26.58 | tmjb | hello i got some wierd callerid form diffrent zapata channels example GSM gateway gives me 00-66-555-555 and ISDN line gives me 66-555-555 but the good nubmer should be 0-66-555-555 any ideas how parse this and send to my phones good caller ID tnx |
| 20:27.01 | lee_is_me | and use the sentinel to indicate a voicemail extension? |
| 20:28.30 | falz | I don't know what "the sentinel" is |
| 20:28.31 | lee_is_me | sorry, an indicator character? |
| 20:28.31 | lee_is_me | like with Credit cards which have starting and ending sentinels to indicate start/stop of track1, track2,etc |
| 20:28.31 | PupenoR | Is there some documentation somewhere explaining how provisioning works in Linksys PAP2? the first file is read but the second and third aren't being read. |
| 20:28.40 | ctooley | Mercestes, thanks for trying to help |
| 20:28.43 | *** part/#asterisk ctooley (n=ctooley@adsl-68-95-129-178.dsl.rcsntx.swbell.net) |
| 20:29.25 | lee_is_me | falz: *111 means to to that extensions VM wherease 111 dials directly? |
| 20:30.49 | falz | lee_is_me: that's what I did, and it worked fine |
| 20:30.57 | falz | unless you have * reserved for something else |
| 20:31.17 | *** join/#asterisk Juggie (n=juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 20:31.22 | lee_is_me | falz: not using * for anything else. I will try that. Thanks a bunch. |
| 20:31.35 | Mercestes | falz: Everyone has that problem on grandstream phones. |
| 20:31.36 | Mercestes | ~gs |
| 20:31.38 | jbot | gs is, like, South Georgia and the South Sandwich islands, or ghostscript. GrandSuck phones are cheap junk which should be avoided with extreme prejudice |
| 20:32.28 | lee_is_me | my first phone as GS budgetone. It's now on my wife's desk... |
| 20:32.38 | Mercestes | lee_is_me, your ex-wife? |
| 20:32.43 | lee_is_me | lol |
| 20:32.55 | falz | I'm not of the opinion that they should be avoided, for the $50 or whatever cost, it's tough to beat. |
| 20:33.10 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 20:33.14 | lee_is_me | falz: it was a great starter phone to learn on |
| 20:33.18 | FuriousGeorge | hey all |
| 20:33.34 | Mercestes | falz: I have two cans and some string that I'll sell you for $15 that outperforms your grandstream. |
| 20:33.43 | falz | meh. |
| 20:33.46 | Mercestes | falz: And a nice bucket of trash for $5. |
| 20:33.58 | Mercestes | falz: hey! for $15 it's hard to beat, eh? |
| 20:34.07 | falz | Mercestes: if they did sip, then I'd say yes. |
| 20:34.18 | Mercestes | falz: Of course they do sip. |
| 20:34.30 | FuriousGeorge | hey all |
| 20:34.37 | falz | obviously a budgetone raped your sister or something. |
| 20:34.46 | Mercestes | falz: You just hook it to an analogue device and run that to an ATA and convert that over to g711, total cost: Only about $60. |
| 20:35.02 | Mercestes | tough to beat for $60.\ |
| 20:35.32 | falz | message waiting inicato? button to auto-route to voicemail? |
| 20:35.41 | falz | transfer/conference buttons? |
| 20:35.43 | Mercestes | falz: No, your rebuttal is falicitious. It's $50 because it's total trash. Trash being sold for $50 is easy to beat. |
| 20:35.47 | Mercestes | however..... |
| 20:36.11 | falz | like I said, what the hell did grandstream do to you? |
| 20:36.24 | falz | we've got 8ish of these, they sit in rooms that get phone calls once a month |
| 20:36.31 | Mercestes | since you are using free software to run a cheap ass phone and lurking in an open support channel asking for free, voluntary help to fix whatever you scraped together with the same amount of money I pay my step son a week to mow my lawn... |
| 20:36.53 | Mercestes | I'm going to say the answer to your question is in the design of the equipment you are trying to use. |
| 20:36.54 | falz | no, I just don't want 7960's in rarely used areas. |
| 20:37.15 | FuriousGeorge | i think chan_zap is causing * to deadlock :( |
| 20:37.16 | falz | the question about voice vlans not working on some firmware versions? |
| 20:37.32 | FuriousGeorge | people call me saying they either cant call out, cant answer incoming calls or both |
| 20:38.03 | Mercestes | falz: Pretty much any question you have about grandstream that is performance related. |
| 20:38.07 | FuriousGeorge | interestingly, they "call out" via voip, but the deadlocks always seem to coincide with incoming analog calls |
| 20:38.30 | FuriousGeorge | all of the servers i install use multiple fxs, but this is the only one using multiple fxo |
| 20:38.46 | FuriousGeorge | the other ones roll calls over to a voip DID |
| 20:38.50 | Mercestes | FuriousGeorge, * version, zap version, which card? |
| 20:38.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:38.52 | falz | Mercestes: seriously, I'm curious, what happened with you and grandstream? |
| 20:39.12 | Mercestes | ... |
| 20:39.21 | Mercestes | it ate my puppy. |
| 20:39.28 | Mercestes | happy now? |
| 20:39.29 | FuriousGeorge | mercestes 1.2.17 on amd_64 |
| 20:39.34 | falz | that would do it. |
| 20:39.46 | FuriousGeorge | zap version is 1.2.16 |
| 20:39.51 | Mercestes | FuriousGeorge, Are you using a 64 bit distro? |
| 20:39.56 | FuriousGeorge | Mercestes: yes |
| 20:40.19 | Mercestes | FuriousGeorge, Could be the 64bit stuff. |
| 20:40.52 | Mercestes | falz: Seriously? Every phone I tried really sucked, and I've heard nothing but "grandsuck" in this channel. |
| 20:40.56 | FuriousGeorge | Mercestes: i suppose. i have another almost identical box running fine on 64-bit. the difference being the other one has only one incoming zap channel |
| 20:42.08 | Mercestes | falz: nothign about the phone really works. Yo ucan't buy the cheapest phone on the market and hope all the bells and whistles work. It's like buying a dlink at walmart and trying to use it as a Cisco 2650. |
| 20:42.28 | FuriousGeorge | i assume they are deadlocks because i always see in messages "avoided initial deadlock" which makes me think that maybe it didnt avoid the subsequent ones. |
| 20:42.29 | Mercestes | falz: It's a $50 phone, you program it as a sip peer, plug it in, pray it works. |
| 20:42.47 | Mercestes | FuriousGeorge, Yea the magical system instability messages. |
| 20:42.50 | falz | heh. |
| 20:42.59 | falz | it works. just fine. fortunately I don't use it to talk on |
| 20:43.09 | Mercestes | falz: ROFL |
| 20:43.13 | Mercestes | yea, they ring great |
| 20:43.22 | lee_is_me | lol |
| 20:43.39 | Mercestes | FuriousGeorge, don't use FoP I suppose. |
| 20:43.45 | lee_is_me | Mine was able to hide the horrible humming noise coming from my verizon line... |
| 20:44.13 | lee_is_me | polycom 301 picked up on it quite well on the other hand |
| 20:45.13 | Mercestes | FuriousGeorge, I hate to point at Sip Debugs. Might wanna watch your console fo rwhat's going on tho. |
| 20:45.23 | Mercestes | FuriousGeorge, Does the other 64bit box run the same version of asterisk/zaptel?? |
| 20:45.26 | *** join/#asterisk Explisit (n=explisit@213.240.243.141) |
| 20:45.47 | PupenoR | Does anybody know what's the purpouse of the different profile rules of provisioning of a PAP2? |
| 20:45.53 | FuriousGeorge | Mercestes: OLDER |
| 20:45.56 | FuriousGeorge | sorry caps |
| 20:46.35 | Mercestes | FuriousGeorge, May want to roll back the versino as a test then. |
| 20:46.43 | Mercestes | FuriousGeorge, mirror the other system as closely as possible. |
| 20:47.02 | Mercestes | FuriousGeorge, And turn up your verbosity and core debug 1 and check your messages for indicators as to what's going wrong |
| 20:47.03 | Explisit | Does anybody has a script similar to this situation - when a call is received the dialing number is check for match in database and if found the sql server return info about the match in browser so the user can see it. |
| 20:47.04 | FuriousGeorge | Mercestes: im thinking of just tarring up the root and kernel of the other server and eliminating software as a problem all together |
| 20:47.05 | *** join/#asterisk zsolt_x (i=1000@cable-87-116-186-114.dynamic.sbb.co.yu) |
| 20:47.17 | FuriousGeorge | the only diff is one has onboard audio which i dont use |
| 20:47.29 | zsolt_x | hello everybody |
| 20:47.39 | zsolt_x | I quick question if someone has time ? |
| 20:47.54 | Mercestes | FuriousGeorge, Coul ddo that if it's identical hardware |
| 20:48.05 | zsolt_x | anyone please ? |
| 20:48.10 | MikHell | Any way to easily install a speaking clock extension on asterisk? |
| 20:48.13 | zsolt_x | will not take long |
| 20:48.14 | FuriousGeorge | Mercestes: this has been a constant problem for this server. the only factor that remains the same is the tdm400p card that is in it |
| 20:48.27 | lee_is_me | zsolt: I'm new to irc myself, but I think you should just ask your question |
| 20:48.29 | FuriousGeorge | err, make that two |
| 20:48.42 | FuriousGeorge | one with 4 fxs and one with 3 fxo |
| 20:48.45 | zsolt_x | thanks lee_is_me |
| 20:48.57 | *** part/#asterisk Explisit (n=explisit@213.240.243.141) |
| 20:49.04 | FuriousGeorge | and like i said, a deadlock always seems to coincide with incoming calls |
| 20:49.11 | zsolt_x | how can I get the ID of the called peer before it picks up the phone, using MAPI ? |
| 20:49.29 | FuriousGeorge | i was thinking of switching to sangoma, but i doubt hardware not developed by digium would work any better with asterisk |
| 20:49.35 | zsolt_x | I need to disconnect the call using MAPI, if the call is taking too long to answer |
| 20:50.54 | Mercestes | FuriousGeorge, I doubt is the card. |
| 20:51.00 | zsolt_x | anyone ? |
| 20:51.48 | lee_is_me | zsolt: MAPI as in message application programming interface? |
| 20:52.05 | zsolt_x | yepp |
| 20:52.34 | zsolt_x | the problem is that I don't have the ID of the called peer |
| 20:52.38 | lee_is_me | zsolt: sorry i do not know. But curious as to why you want to use that |
| 20:52.49 | zsolt_x | so I cannot initiate a hangup message trough MAPI |
| 20:53.00 | lee_is_me | ah, ok |
| 20:53.11 | zsolt_x | I'm working on something and that is what I need :-) |
| 20:54.17 | lee_is_me | zsolt: ok, was just curious |
| 20:54.55 | zsolt_x | sure np |
| 20:55.11 | zsolt_x | if anyone has a clue, or an idea please let me know |
| 20:55.16 | zsolt_x | I will stick around |
| 20:58.30 | *** join/#asterisk _Sam-- (n=sam@fresco.kneedraggers.com) |
| 20:59.18 | _Sam-- | hey i have this parameter set for your voicemail, but most times callers messages are still cut at 1 minute....any ideas? |
| 20:59.19 | _Sam-- | Connection |
| 20:59.21 | _Sam-- | ; Maximum length of a voicemail message in seconds |
| 20:59.21 | _Sam-- | maxmessage=180 |
| 20:59.33 | *** join/#asterisk spatulamaan (n=ggilmore@65-102-118-77.tukw.qwest.net) |
| 21:04.18 | *** join/#asterisk spatulamaan (n=ggilmore@65-102-118-77.tukw.qwest.net) |
| 21:07.59 | *** join/#asterisk naitram (n=ttech@216.77.58.40) |
| 21:08.55 | naitram | how do you use the # key in an exten, such that the #1101 means dial pound then 1101, do you have to escape it with some character? |
| 21:09.14 | Strom_M | naitram: don't start an extension with # |
| 21:09.29 | Strom_M | # tends to mean "I'm finished dialing; put the call through now" |
| 21:09.51 | naitram | Strom_M: ok, thanks |
| 21:13.25 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 21:14.03 | naitram | is there a dial string to just do nothing, i am running scripts based on a series of digits dialed and then want to end but i have to physically push the end key |
| 21:15.12 | irule | is there a good doc online regarding security? I want to make sure I am not vulnerable with a home setup |
| 21:15.33 | Strom_M | naitram: I don't understand what you mean |
| 21:16.56 | naitram | Strom_M: I am going to use asterisk for both voip and also some control functions for opening doors. I have written php scriptst that can be called from the extensions.conf for a dialed string. |
| 21:17.41 | naitram | when i dial now, since there is nothing for asterisk to actually dial, my phone displays dialing even though my scripts are thru? |
| 21:18.12 | *** join/#asterisk bkw_ (i=brian@ppp-70-128-123-137.dsl.tulsok.swbell.net) |
| 21:18.33 | naitram | I wan to end the call since in fact Im not really making one, after the scripts are through running, Just a sec and I will paste to pastebin.ca |
| 21:19.05 | zsolt_x | how can I get the ID of the called peer before it picks up the phone, using MAPI ? |
| 21:19.11 | zsolt_x | I need to disconnect the call using MAPI, if the call is taking too long to answer |
| 21:19.21 | zsolt_x | the problem is that I don't have the ID of the called peer |
| 21:19.35 | naitram | <PROTECTED> |
| 21:20.38 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:22.17 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 21:22.48 | naitram | I thought there was like a dial no op or something |
| 21:24.56 | Strom_M | naitram: so you run the script and then execute a Hangup() |
| 21:24.59 | Strom_M | what's so tough about that? |
| 21:26.54 | naitram | Strom_M: pretty new to this, probably nothing hard, just ignorance. Please enlighten? |
| 21:29.07 | Strom_M | naitram: first, don't start an extension with * either - you conflict with assigned vertical service code format |
| 21:29.35 | Strom_M | second, extension priorities should not be numbered 1-10; the first should be numbered "1" and the rest "n" |
| 21:30.47 | Strom_M | finally, you should have the first priority in your extension be a Progress() and the last priority be Hangup() |
| 21:30.50 | slmnhq | So I've read chapters 1, 2, 5, 7, and 9 of the O'Reilly book.. but I am beginning to think that it may not have the answers I am looking for |
| 21:31.17 | slmnhq | Is it okay if I ask a Off topic question.. not necessarily relating to debugging/configuring Asterisk? |
| 21:31.46 | Strom_M | slmnhq: ask away |
| 21:31.50 | naitram | Strom_M: Ok, will do. thanks |
| 21:33.00 | slmnhq | I am trying to develop an application which will initiate a call between two arbitrary phone users (these parties could be on PSTN, GSM, whatever.. with in a specific country) |
| 21:33.23 | slmnhq | Neither party knows the number of the other party, or is interested in knowing that information... |
| 21:33.46 | *** part/#asterisk naitram (n=ttech@216.77.58.40) |
| 21:33.53 | slmnhq | I think my application will have to initiate a 3-way conference call with the other two parties and disconnect |
| 21:34.06 | *** join/#asterisk paavum (n=Dorphals@200.71.58.39) |
| 21:34.08 | paavum | Hello |
| 21:34.19 | paavum | Im trying to send a fax to a linksys ata |
| 21:34.24 | paavum | which codec should I use? |
| 21:34.26 | slmnhq | I'm trying to get a high-level picture here to understand where Asterisk, my local telephone provider, etc fit in |
| 21:35.29 | voipman | how do you ignore an incoming call / dynamically send to voicemail with cisco 79XX's? |
| 21:35.34 | paavum | (Fax --> pstn --> asterisk --[lan]--> WRT54GP --> Fax ) |
| 21:35.36 | toot | slmnhq home > asterisk > teleco provider :) |
| 21:36.16 | slmnhq | Thanks toot, what kind of an interface should exist between Asterisk and the Telco? |
| 21:36.27 | Strom_M | paavum: unless you're doing T.38 passthrough, don't waste your time |
| 21:37.45 | Strom_M | fax over voice over IP is not a reliable idea :) |
| 21:37.46 | toot | an nice digium card (www.digium.com) |
| 21:40.40 | ChkDigit | T.38 works well in ideal conditions... |
| 21:41.08 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-208f222089074a4e) |
| 21:43.34 | *** join/#asterisk saftsack (n=saftsack@pD9E0633B.dip.t-dialin.net) |
| 21:43.45 | *** join/#asterisk techie (n=gus@voip.routedsystems.com) |
| 21:45.54 | paavum | but t38 is only available in 1.4, right? |
| 21:45.54 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:45.55 | a1fa | anybody know of a sip client for windows mobile 5.1 |
| 21:45.57 | paavum | Strom_M --> its a low traffic lan |
| 21:46.58 | a1fa | i just got a Samsung Blackjack |
| 21:48.11 | thekidrio | what format should I use for touch -d to set a call for some time in the future? for example $ touch -d 02:25:00 PM PDT num1.2.num2.call |
| 21:48.20 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:50.06 | a1fa | anybody know any company that makes sip phones for 5.1 mobile windows |
| 21:51.51 | a1fa | i cant believe there isnt one |
| 21:51.59 | *** part/#asterisk mattwj2005 (n=Matt@user-38q4155.cable.mindspring.com) |
| 21:51.59 | lee_is_me | thekidrio: I tried using touch myself, but for some reason, I couldn't get it to work. I ended up using linux "at" command instead |
| 21:52.14 | Mercestes | <PROTECTED> |
| 21:52.28 | Nugget | Don't touch yourself, you'll go blind. |
| 21:52.30 | lee_is_me | lol |
| 21:52.40 | thekidrio | Lee_is_me i just got it to work, I was being silly and not using a 24 hour clock |
| 21:52.47 | thekidrio | PM does not work hehe |
| 21:53.00 | thekidrio | had to do 14:25:00 |
| 21:53.10 | lee_is_me | what do i do about the hair already on my knuckles? |
| 21:53.45 | lee_is_me | thekid: I tried that as well myself and couldn't get it to work. "at" works great for me... |
| 21:55.29 | lee_is_me | alfa: I tried one on pocketpc 2003 a couple months ago, but it didn't work very well. Can't remember the name of it though |
| 21:56.25 | *** join/#asterisk rvhi3 (n=as@66.175.65.82) |
| 21:56.59 | a1fa | Windows Mobile 5.1 is a differnt thing |
| 21:57.39 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 21:57.42 | xo8ox | guys |
| 21:57.55 | a1fa | what version of windoz? |
| 21:58.18 | *** part/#asterisk sselby (n=sselby@txplano-nat208.dc.xo.com) |
| 21:58.23 | xo8ox | I have cisco 7960 and trying to register it with our asterisk server but it just wont register.. |
| 21:58.31 | xo8ox | I can ping the phone from the server and it pings |
| 21:58.39 | xo8ox | so I don't know what it is I'm missing |
| 21:58.57 | lee_is_me | alfa: were you asking me? |
| 21:59.39 | *** join/#asterisk plantseeker (n=chatzill@host81-154-189-53.range81-154.btcentralplus.com) |
| 22:01.30 | plantseeker | hello |
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| 23:12.45 | i3inary | im looking for a good method of blocking numbers from originates...i dont seem to be able to do that from my extensions.conf am i missing something? |
| 23:13.14 | i3inary | the call seems to process before the extension.conf ....the calls are coming from .call files fyi |
| 23:14.24 | *** join/#asterisk ardor (n=Miranda@las-static-66.18.135.148.mpowercom.net) |
| 23:14.27 | [TK]D-Fender | i3inary, perhaps you should pasten the sample call-file & your extensions.conf |
| 23:15.32 | ardor | test |
| 23:15.38 | ardor | can you guys here me? |
| 23:15.44 | onecentld | anyone having problems with 1.4.2 and dtmf |
| 23:17.28 | a1fa | anybody know of a sip or iax client that will work on Windows Mobile 5.1 |
| 23:20.34 | *** join/#asterisk ardor (n=Miranda@las-static-66.18.135.148.mpowercom.net) |
| 23:20.36 | threat | onecentld, what type of problems? |
| 23:21.05 | i3inary | sure i can paste that stuff but...i was just thinking that im missing a concept regarding the originate call funtion |
| 23:21.19 | threat | onecentld, asterisk continues to ring even though the person has hang up :) that is my problem, don't know if it is dtmf related though |
| 23:21.35 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-219-72.hsd1.il.comcast.net) |
| 23:21.59 | kuku5 | When I do an assisted transfer, is there a way to do a beep so everyone know that the call is transfered ? |
| 23:26.07 | *** join/#asterisk thoughtpolice (n=austin@c75-111-145-138.plaicmtc01.tx.dh.suddenlink.net) |
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| 23:30.20 | i3inary | anyone know what extensions.swo is off the top of their head? mine is uhhh 244 meg |
| 23:31.39 | [TK]D-Fender | i3inary, entirely possible, but until you show us what you're doing we'll never know :) |
| 23:31.48 | [TK]D-Fender | i3inary, and never head of that file |
| 23:32.11 | i3inary | ok ill have to research it i think its growing too |
| 23:32.26 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 23:32.26 | *** mode/#asterisk [+o mog] by ChanServ |
| 23:32.50 | i3inary | actually its .extensions.conf.swo |
| 23:33.15 | *** join/#asterisk mavior (n=chatzill@81-174-8-252.f5.ngi.it) |
| 23:33.28 | [TK]D-Fender | i3inary, sounds like crap to me... |
| 23:35.07 | mavior | hello |
| 23:40.08 | mavior | if i try to dial an extension ,and immediately after that(like if i have dialed some wrong number) I hang up (for approximately 1 sec) and then I re-pick the phone I heard no dial tone, and my console display that http://pastebin.ca/443754 , i'm using a tdm400p on centos 4.4 , ast ver 1.4.0 zap 1.4.0 |
| 23:43.42 | mavior | seems like asterisk is busy with the extension expression matching and can't hangup |
| 23:50.43 | *** join/#asterisk ariel_ (n=ariel_@dsl-20-177.cofs.net) |
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| 23:51.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:53.02 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 23:53.44 | rhombus | Can anyone tell me the real difference between the e-mail and pager parameters in voicemail.conf? |
| 23:54.35 | rhombus | I presume that notifications sent to pager addresses don't include the voice file attachment even if attach=yes |
| 23:54.55 | _VoiceMeUp_Com | pager could be an email to sms maybe |
| 23:54.58 | _VoiceMeUp_Com | not sure |
| 23:55.15 | *** join/#asterisk mrdigital (n=ASFASDF@pool-72-94-124-149.phlapa.east.verizon.net) |
| 23:55.20 | _VoiceMeUp_Com | makes sense |
| 23:56.33 | rhombus | I suppose I'll have to try it -- but I sure wish it were documented somewhere |
| 23:57.09 | DrukenLPY | does someone have a firmware other than 1.4.1 for the aastra 9112i ? |
| 23:57.32 | DrukenLPY | seems i can't jump to 1.4.1 from the current firmware of the phone |
| 23:58.03 | BSD_Tech | 1.4.2 is better update from 1.4.1to 1.4.2 |
| 23:58.10 | BSD_Tech | more stable |
| 23:59.35 | DrukenLPY | and where do i get 1.4.2? since 1.4.1 is the latest posted on aastra's site.... |
