| 00:00.09 | buzkashi | cable companies that support voip |
| 00:00.10 | *** join/#asterisk [netman] (n=netman@24.Red-88-22-68.staticIP.rima-tde.net) |
| 00:00.53 | jeff_phillips | so these little boxes that plug into the cable coax and give you an RJ11 jack -- is there a similar device that will plug this into asterisk either a PCI card with a coax connector, or a coax to IP/ethernet ??? |
| 00:01.08 | buzkashi | no |
| 00:01.25 | buzkashi | many of these companies are clueless about asterisk or any tdm card |
| 00:01.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 00:01.52 | UQlev | jeff_phillips: you need only ethernet cable |
| 00:02.18 | jeff_phillips | VQlev: yeah but my ethernet connection from the cable company doesn't have their phone service in it, that I can tell |
| 00:02.39 | buzkashi | no I am using rj11 the signal is going over coaxial to pstn |
| 00:03.03 | jeff_phillips | Yeah, see I'd like to get rid of the RJ11 |
| 00:03.06 | UQlev | jeff_phillips: you may establish as many you want sip/iax accounts independently of your ISP |
| 00:03.19 | buzkashi | true |
| 00:03.21 | jeff_phillips | VQlev: yes, but the cable company's phone service isn't SIP |
| 00:03.28 | buzkashi | correct |
| 00:03.31 | *** join/#asterisk ingenius (n=alektro@host253.190-31-41.telecom.net.ar) |
| 00:03.40 | jeff_phillips | It's this stupid little box they give us with a coax connector and an analog RJ11 connector |
| 00:04.00 | buzkashi | depending on what flavor of internet connection you recieve have to make sure ports are open for traffic to support sip |
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| 00:04.33 | UQlev | jeff_phillips: usually ISP will not give you password to their box |
| 00:04.40 | jeff_phillips | well yeah, I use SIP over the IP connection for our own calls -- but I'm wanting to take advantage of this cable company's phone line that's sitting here |
| 00:04.52 | jeff_phillips | The cable phone doesn't use the IP connection |
| 00:05.08 | jeff_phillips | or if it does, it's a completely seperate IP connection than what they give me |
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| 00:05.49 | buzkashi | right interesting |
| 00:06.06 | jeff_phillips | it's essentially the "mystery box" |
| 00:07.15 | lesouvage | I jut found bug 0014167 about callfile generated outbound calls that aren't logged in the cdr database. For those using callfiles this can result in very unpleasant surprises especially when you use the cdr for billing |
| 00:10.40 | pdmmm | 0evening. |
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| 00:17.04 | buzkashi | I will see what I can do about tweaking rxgain |
| 00:17.13 | buzkashi | With this particular setup |
| 00:17.52 | jeff_phillips | i did something similar with using a Tmobile@Home service which uses SIM cards stuck in a Linksys router they send you, with analog RJ11 jacks... |
| 00:18.07 | jeff_phillips | I sure wish I wouldn't have signed up for it. it would have been a lot better to pay more and go with a real SIP account |
| 00:18.20 | jeff_phillips | I really don't like converting to analog to go just a few inches |
| 00:19.08 | buzkashi | true |
| 00:19.57 | jeff_phillips | i had to really crank up the gain levels to get it to sound equal to our SIP trunks, but when doing that I introduced an aweful echo and had to lower it a bit. Still the quality noticably sucks |
| 00:20.28 | buzkashi | interesting |
| 00:21.43 | jeff_phillips | I didn't want the cable company's phone service at all, but they would only sell us IP connectivity if we promise to spend at least $110 /month with them so either I pay for the phone and not get one, or I pay for it and get it. |
| 00:21.59 | jeff_phillips | something about them wanting to get an ROI on building their wires out long enough to reach our building. bah. |
| 00:22.23 | buzkashi | Yes the cable companies are becoming creative but still will not offer an SLA |
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| 00:23.24 | jeff_phillips | that's because they hire people like larry the cable guy to come out and fix stuff |
| 00:23.34 | buzkashi | true |
| 00:26.17 | phix | [TK]D-Fender: you still there? |
| 00:26.56 | phix | I have worked out what was going on, I wasn't waiting enough time after Answer for zap drivers to determine that call was a fax |
| 00:28.17 | phix | I now answer, playtones, wait(4) then after that goto my incoming context, talk goes to incoming context and fax goes to incoming-fax context |
| 00:28.23 | jeff_phillips | "PLEASE NOTE: Although Charter's chat line service is provided for service related assistance, Charter is |
| 00:28.24 | jeff_phillips | unable to accept or process customer complaints via this service. |
| 00:28.27 | jeff_phillips | haha |
| 00:28.30 | phix | although I still don't like how it takes 4 secs to determine if it is a fax or not |
| 00:29.00 | phix | jeff_phillips: wtf |
| 00:29.01 | jeff_phillips | 4 seconds is pretty quick for technology as old as a fax |
| 00:29.12 | phix | jeff_phillips: noted |
| 00:29.27 | jeff_phillips | Charter is our cable company. I clicked to ask them a question and apparently you can't share a complaint |
| 00:30.39 | buzkashi | not surprised |
| 00:33.09 | jeff_phillips | buzkashi: did the cable company give you one box for both your RJ-11 phone & internet/ethernet, or is it two seperate pieces of equipment? They haven't sent me mine yet |
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| 00:34.56 | buzkashi | one box |
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| 00:35.44 | jeff_phillips | what brand / model is it? I can probably get more information out of the manufacturer specs than I can out of these people. The guy doesn't seem to have any idea what a PBX system is. |
| 00:39.44 | buzkashi | motorola sbv5120 |
| 00:40.10 | jeff_phillips | lol this guy in the same conversation has stated that both the IP and phone connections come out of the same box, but also stated that they are two seperate motorola devices |
| 00:40.18 | jeff_phillips | one for internet, one for phone. so which is it? lol |
| 00:40.34 | jeff_phillips | why is it that clueless companies rule the world? |
| 00:40.37 | apeiron | There's a magical box that solves all of your communications problems! |
| 00:42.06 | jeff_phillips | or creates them |
| 00:42.06 | apeiron | heh |
| 00:42.06 | jeff_phillips | buzkashi: thanks, i'll read up on it |
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| 00:45.12 | coppice | jeff_philips: they aren't clueless. they focus on what drives the global economy - BS |
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| 00:53.32 | jeff_phillips | lol |
| 00:54.56 | coppice | don't laugh. its a noble cause. BS is a renewable resource, and extremely plentiful |
| 00:55.02 | buzkashi | jeff we talk later and good luck |
| 00:55.17 | buzkashi | And thanks everyone |
| 00:55.38 | jeff_phillips | good luck buzkashi. |
| 00:55.44 | buzkashi | Yes I am going to need it |
| 01:01.06 | jeff_phillips | omg this fool tells me that he can't say what model of device he's sending me, but I should visit their local office to ask such a question. Then he asks for my zip code and says that they don't have a local office and because of such he can't even tell me where the nearest non-local office is |
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| 01:14.25 | apeiron | ~book |
| 01:14.35 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 01:14.36 | apeiron | pokes jbot |
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| 02:07.58 | jeff_phillips | Has anyone used Asterisk *as* a DOCSIS PacketCable MTA? |
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| 02:13.26 | RichardLynch | Does the SendText() application send MMS text messages? And does that require some kind of special back end connection, or just an open phone line to somebody's hand-set that accepts TXT?... |
| 02:13.31 | m308 | Is there some sort of IVR functionality that will look up data in a db and based on the response state something ? |
| 02:14.22 | RichardLynch | You can definitely tie into a DB with the ODBC module. What you do with it is up to you. |
| 02:18.05 | m308 | What do u mean ? |
| 02:18.15 | m308 | I have no idea how the IVR system works |
| 02:18.20 | m308 | is there a good guide online ? |
| 02:19.18 | jaytee | m308, try the Wiki at www.voip-info.org, just search for Asterisk+IVR+database |
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| 02:23.15 | m308 | jaytee: nothing |
| 02:24.53 | RichardLynch | m308: Google for "asterisk postgresql" There's a dirt-simple "howto" out there. |
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| 02:25.31 | RichardLynch | Or, no, it's in "the asterisk book"! Chapter 12. |
| 02:26.15 | m308 | RichardLynch: http://www.asteriskguru.com/tutorials/realtime_pgsql.html is what you are refering to ? |
| 02:26.22 | m308 | thats just configuring asterix via a db |
| 02:26.37 | m308 | I want the functionality of a user being able to call up, put in there id |
| 02:26.41 | m308 | and get a answer |
| 02:27.24 | m308 | ideally they call in get "authenticated" via 2 selects and then play either of 2 audio files based on the result |
| 02:27.41 | RichardLynch | m308: Then Google for it, as I said. It's out there. I did it, so it can't be too terribly difficult... |
| 02:28.23 | m308 | RichardLynch: I wouldnt be on IRC asking if I didnt spend several days searching for a solution |
| 02:28.26 | m308 | but thanks for your help |
| 02:28.32 | m308 | you troll |
| 02:29.20 | RichardLynch | Okay, m308, let me go google FOR you. |
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| 02:33.10 | RichardLynch | m308: If you read Chapter 12, all of Chapter 12, and go through the HotDesk part, you end up seeing exactly how you can get ANY data in/out of a callplan with PostgreSQL... You're still on your own for the IVR part. |
| 02:33.54 | m308 | where is the asterisk book ? |
| 02:35.02 | dwayne | ~thebook |
| 02:35.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:38.46 | m308 | How is there a free download of it available ? |
| 02:41.41 | m308 | how do I get a inbound voip line in asterisk ? |
| 02:41.58 | m308 | rather a did |
| 02:42.42 | Mog | yes in those links |
| 02:42.53 | Mog | you need a provider to get a did |
| 02:43.01 | Mog | like voicepulse |
| 02:44.16 | m308 | Mog: did u ever hang out on irc.newnet.net ? |
| 02:44.31 | Mog | no |
| 02:46.59 | m308 | k |
| 02:47.04 | m308 | but thanks for that major hint |
| 02:47.52 | Mog | no worries |
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| 03:40.42 | areay | i have asterisk setup with 1 sip trunk... i cant receive more than 2 concurrent incoming calls |
| 03:40.52 | areay | am i doing something wrong, or am i just not paying enough |
| 03:41.57 | [TK]D-Fender | areay: are we supposed to be psychic? |
| 03:42.14 | criten | [TK]D-Fender: yes |
| 03:42.24 | thehar | haha |
| 03:42.27 | [TK]D-Fender | criten: Stop telling me answers I already know! |
| 03:42.28 | areay | [TK]D-Fender, do you think its a limitation of my provider? |
| 03:42.28 | [TK]D-Fender | :p |
| 03:42.35 | criten | see, you are psychic |
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| 03:42.59 | [TK]D-Fender | areay: No I think we know nothing about the actual situation sin you aren't SHOWING US ANYTHING |
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| 03:43.03 | [TK]D-Fender | since* |
| 03:43.11 | jaytee | hehe |
| 03:43.19 | ctooley | What in Asterisk is set up to listen on 14292 and 14293 by default. |
| 03:43.28 | ctooley | both are UDP |
| 03:43.33 | [TK]D-Fender | ctooley: nothing |
| 03:43.54 | ctooley | [TK]D-Fender, That's what I was afraid of... |
| 03:43.58 | ctooley | udp 0 0 0.0.0.0:14292 0.0.0.0:* 3772/asterisk |
| 03:43.58 | ctooley | udp 0 0 0.0.0.0:14293 0.0.0.0:* 3772/asterisk |
| 03:44.06 | thehar | oh I LOVE PASTEIES |
| 03:44.08 | ctooley | from 'netstat -anp | grep asterisk' |
| 03:44.17 | thehar | is that what that neat thing is? |
| 03:44.18 | ctooley | Not good. |
| 03:44.29 | jaytee | well, 10000-20000 is the default range for RTP |
| 03:44.35 | areay | [TK]D-Fender, i've checked sip/iax2 debug... nothing... i dont even think the 3rd caller makes it to asterisk |
| 03:44.53 | jaytee | must be your ITSP then |
| 03:44.55 | [TK]D-Fender | areay: If your eyes were good enough you'd ahve found the answer already |
| 03:45.17 | areay | [TK]D-Fender, why would my provider limit the number of incoming calls? to force me to buy a business contract? |
| 03:45.27 | ctooley | jaytee, Yeah... I think that might of been it. Call shut down and now they aren't there. |
| 03:45.49 | [TK]D-Fender | areay: mayebe you arelady HAVE one. Go look at the terms you signed up under. |
| 03:46.13 | areay | [TK]D-Fender, i don't... i couldn't find an unlimited business contract, so i went for a personal one |
| 03:46.32 | areay | [TK]D-Fender, like i'd be using voip if i wanted to pay per minute... |
| 03:46.46 | jaytee | did you read the fine print (mutters to himself, "they never friggen read the fine print" |
| 03:46.53 | areay | jaytee, of course not :) |
| 03:47.07 | jaytee | SEE!!!! gobble, gobble |
| 03:47.12 | areay | lol |
| 03:47.40 | jaytee | [TK]D-Fender, can I borrow your ClueBat for a second? :-) |
| 03:47.46 | [TK]D-Fender | areay>[TK]D-Fender, like i'd be using voip if i wanted to pay per minute... <---- If you're not using "VoIP", then WTF are you doing? |
| 03:47.50 | areay | can anybody recommend an unlimited business voip contract for the UK? (yes it is 3:48am here) |
| 03:48.01 | areay | [TK]D-Fender, i am |
| 03:48.01 | [TK]D-Fender | jaytee: NO! MPRECIOUS!!!!!!!!! |
| 03:48.06 | [TK]D-Fender | MY* |
| 03:48.07 | jaytee | hahaha |
| 03:48.11 | areay | [TK]D-Fender, i just dont want to pay per minute for outgoing calls |
| 03:48.30 | [TK]D-Fender | areay: So? Go look at what you're PAYING FOR |
| 03:49.44 | areay | [TK]D-Fender, i think its obvious they dont want me using this contract for business.. it just disconnects after 2 concurrent calls, and all their business plans are minute bundles |
| 03:49.58 | jaytee | I want a new Crossfire but I don't want to pay for it, people in Hell want ice water. What are either of our chances? |
| 03:50.06 | areay | [TK]D-Fender, know any unltd uk itsps? |
| 03:50.10 | [TK]D-Fender | "obvious" |
| 03:50.12 | areay | ...that use iax |
| 03:50.22 | [TK]D-Fender | Ok, I'm clearly wasting my time again. |
| 03:50.31 | jaytee | ~itsplist-uk |
| 03:50.53 | jbot | itsplist-uk is, like, UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
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| 03:52.17 | *** mode/#asterisk [+ooo Deeewayne Qwell Corydon76-dig] by irc.freenode.net |
| 03:52.17 | jaytee | whoa!!!! |
| 03:52.17 | [TK]D-Fender | *B00M* |
| 03:52.17 | areay | nooo netsplits are bad |
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| 03:52.20 | jaytee | ~itsplist-uk |
| 03:52.21 | jbot | [itsplist-uk] UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 03:52.48 | areay | thanks jaytee |
| 03:52.48 | jaytee | damn, jbot's slow as hell tonight |
| 03:52.48 | areay | lol |
| 03:52.48 | jaytee | ~botsnack |
| 03:52.48 | jbot | aw, gee, jaytee |
| 03:52.48 | areay | jaytee, in your opinion, do you think i'm being capped by my itsp? |
| 03:53.10 | [TK]D-Fender | areay: WTF... seriously... |
| 03:53.28 | areay | [TK]D-Fender, c'mon dude cut me some slack jesus |
| 03:53.35 | jaytee | in my opinion? 5 to 1, half dozen of the other, sure, what the hell. They're cappin ya! clear as day. Even Ray Charles could see that and he's blind. |
| 03:53.36 | [TK]D-Fender | areay: Some slack? |
| 03:53.49 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:53.49 | areay | [TK]D-Fender, yeah ur brutal man |
| 03:53.53 | [TK]D-Fender | areay: You are asking us the &#^$ING GUESS. BLIND! |
| 03:54.15 | [TK]D-Fender | areay: Open your damn eyes are read your CONTRACT for what you PAID FOR |
| 03:54.32 | [TK]D-Fender | areay: You show us nothing. Make your own guesses |
| 03:54.44 | jaytee | damn brits! this is why we had a revolution, they were always trying to dump the work on us yanks. :-) |
| 03:54.48 | areay | [TK]D-Fender, i have, i just wanted a second opinion ;) |
| 03:54.53 | areay | jaytee, i'm american actually |
| 03:54.55 | areay | i defected |
| 03:55.00 | jaytee | traitor!!! |
| 03:55.04 | areay | lol |
| 03:55.09 | areay | im coming back soon dont worry |
| 03:55.13 | areay | asterisk is gonna make me mobile :) |
| 03:55.18 | [TK]D-Fender | areay: I have a second opinion, its jsut like the first one only LOUDER |
| 03:56.19 | jaytee | brits like foods like blood pudding and jellied eels. that kinda makes them seem like effeminate Klingons to me. |
| 03:56.53 | areay | [TK]D-Fender, i know... i stay away from those kind of brits |
| 03:57.01 | areay | [TK]D-Fender, there are cool ones tho |
| 03:57.05 | areay | [TK]D-Fender, ...some |
| 03:57.14 | areay | sorry tk ifucked that up |
| 03:57.20 | areay | it was meant for jaytee |
| 03:57.24 | [TK]D-Fender | areay: I'm not stereotyping, I'm singling you out uniquely :p |
| 03:57.30 | jaytee | hehe |
| 03:57.48 | areay | [TK]D-Fender, lol... i have to admit if it wasnt for u being a hardass i never would have rtfm'd |
| 03:58.24 | [TK]D-Fender | prints THE BOOK on 120 bond paper and Jason Bourne's areay with it |
| 03:58.37 | jaytee | he's like that, he won't give the pat answer, he'll make you work it out. might throw you a clue here and there but usually tries to make you think it through. |
| 03:58.57 | jaytee | cuz otherwise you'll never learn it |
| 03:59.07 | areay | true dat |
| 03:59.19 | jaytee | [TK]D-Fender, you should have gone into teaching |
| 04:00.06 | [TK]D-Fender | jaytee: there'd be a higher passing rate... and a mysteriously proportionately higher death rate..... |
| 04:01.10 | jaytee | hahaha, I just meant you'd at least make the students try and develop some analytical thinking skills instead of the lazy ass tenured shits that just turn out rote memory retards like an assembly line that we have today. |
| 04:02.06 | jaytee | anyways, I've gotta get some sleep. nite all |
| 04:02.16 | [TK]D-Fender | jayl8r |
| 04:17.09 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
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| 04:21.08 | drmessano | OMG, have I been gone that long? |
| 04:21.13 | drmessano | 1.8 is out!!!! |
| 04:21.16 | drmessano | Wait |
| 04:21.24 | drmessano | Damn illiteracy |
| 04:21.26 | drmessano | :( |
| 04:21.33 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 04:25.06 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-e84f63198bfa0333) |
| 04:30.09 | areay | is there an open-source predictive dialer for asterisk? |
| 04:33.00 | carrar | drmessano, no we are on Asterisk 8.1.2 |
| 04:33.23 | drmessano | Sounds more like some damn Cisco number |
| 04:34.02 | carrar | that would be Asterisk 12.4(4)T |
| 04:34.21 | [TK]D-Fender | areay: http://tinyurl.com/cpqevu |
| 04:35.19 | mchou | anyone have experience with Zulty ip phones? |
| 04:37.15 | prg3 | I'm trying to use a Dial,M() macro to do some call screening, and read out the callerid. If a user hits 1, they should get the caller CID, and then be connected. If they do not, it "should" hangup, but it seems to break out of the macro context, and connect the caller, not hangup. Any thoughts on why? http://pastebin.com/m749b308a |
| 04:38.12 | mchou | prg3: just get google voice :) |
| 04:38.35 | prg3 | mchou: it's more complicated.. there's a queue, and agents and all that mess involved :) |
| 04:38.39 | mchou | prg3: cause it's been deployed :) |
| 04:38.52 | areay | [TK]D-Fender, lol... |
| 04:39.34 | *** join/#asterisk hsv-al (n=pluhngz@75.76.166.94) |
| 04:40.27 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 04:40.36 | prg3 | mchou: you've totally lost me. :) |
| 04:40.49 | [TK]D-Fender | prg3: and I don't see you setting the channel variable the instructions tell you to <- |
| 04:40.58 | mchou | prg3: which part did I lose you? |
| 04:41.17 | hsv-al | Tha killaZ |
| 04:41.21 | hsv-al | hello dfender |
| 04:42.22 | prg3 | [TK]D-Fender: none of the instructions I've found.. instructions for the hangup command? |
| 04:42.22 | [TK]D-Fender | prg3: "core show application dial" <---- |
| 04:46.09 | prg3 | [TK]D-Fender: Thanks, that's exactly what I needed to know. |
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| 04:54.43 | *** join/#asterisk rift0r (i=rift@zeus.barlamenterprises.com) |
| 04:55.14 | rift0r | What would suddenly cause me to get Registration ACL permit/deny errors for my polycom ip550 sip phone? |
| 04:55.18 | rift0r | it worked fine for 9 mos |
| 04:55.23 | rift0r | now suddenly i cant get it to register |
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| 05:22.26 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 05:22.57 | Hanif08 | hi.. anybody can help me.. |
| 05:23.07 | Hanif08 | i am configuring asterisknow |
| 05:23.22 | Hanif08 | but.. can't have access as a root |
| 05:23.31 | Hanif08 | i am only manage as a admin |
| 05:24.01 | hsv-al | hello Hanif |
| 05:24.16 | Hanif08 | yes hsv-al |
| 05:24.22 | hsv-al | :) |
| 05:25.23 | Hanif08 | i m downloading iso image from asterisknow |
| 05:25.54 | Mog | Hanif08, it has no root password by default |
| 05:25.58 | Mog | you can sudo su - |
| 05:26.01 | Mog | and then set one |
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| 05:26.08 | Hanif08 | ic.. |
| 05:26.17 | *** join/#asterisk mizerydearia (n=mizeryde@CPE-65-27-127-37.new.res.rr.com) |
| 05:26.38 | Hanif08 | does anyone tried asterisknow b4 |
| 05:26.52 | mizerydearia | I am currently using Gizmo5 Call In service with a us phone number. At my time of purchase of service more than a year ago, I considered $35/year inexpensive and worthwhile to have the ability to receive voice mails sent directly to my email as well as to answer incoming calls from any device that supported sip. I want to learn more about other similar opportunities or solutions that are available that allow me to use VOIP/SIP |
| 05:26.52 | mizerydearia | <PROTECTED> |
| 05:26.55 | Hanif08 | just want to seek your view |
| 05:27.17 | Hanif08 | interm of stability |
| 05:27.28 | Mog | its good |
| 05:27.52 | hsv-al | awwww |
| 05:27.56 | hsv-al | Mog is up late as well |
| 05:28.02 | Mog | yup |
| 05:28.04 | hsv-al | then up again 8 am for "redbull+work" |
| 05:28.06 | hsv-al | lulz |
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| 05:45.17 | rift0r | What would suddenly cause me to get Registration ACL permit/deny errors for my polycom ip550 sip phone? |
| 05:45.20 | rift0r | it worked fine for 9 mos |
| 05:45.22 | rift0r | now suddenly i cant get it to register |
| 05:46.31 | Mog | are you seeing it talk to your asterisk server |
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| 06:22.00 | jameswf | sits back and waits for the next wave of "Control your monkey" emails to his boss |
| 06:28.40 | MainMax | Hey are there any budget devices to interface Fax Machine with asterisk server |
| 06:29.09 | Mog | there are single port t.38 atas |
| 06:29.11 | jameswf | ummm budget devices? |
| 06:30.22 | MainMax | like around $50 per line preferably with Ethernet connection (not a PCI card) |
| 06:35.06 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
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| 06:35.52 | rvhi | is there a way to show the status of a sip peer |
| 06:38.03 | Mog | sip show peer peer |
| 06:38.07 | Mog | or sip show peers |
| 06:38.49 | rvhi | i don't see where it shows the status |
| 06:40.15 | CunningPike | rvhi: Are you using QUALIFY? |
| 06:40.34 | rvhi | eh... no |
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| 07:25.04 | *** join/#asterisk trymi1 (n=please@203.81.71.107) |
| 07:26.23 | trymi1 | hi |
| 07:26.24 | trymi1 | all |
| 07:26.59 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 07:27.06 | trymi1 | hi |
| 07:27.23 | trymi1 | please check http://forums.digium.com/viewtopic.php?t=67761 and pls reply me |
| 07:29.39 | kaldemar | define "not ok". make an incoming call with sip debug and verbosity and show the cli output. |
| 07:31.41 | mort_gib | trymil: Networking, make sure you have portforwarding (if you are using NAT) |
| 07:31.55 | mort_gib | trymil: and have a close look at your firewall |
| 07:32.05 | trymi1 | yes |
| 07:33.07 | trymi1 | I can use direct access to this provider server. |
| 07:33.07 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:33.11 | trymi1 | this is NAT or not |
| 07:33.45 | trymi1 | ? |
| 07:35.53 | trymi1 | how can I check? |
| 07:36.37 | mort_gib | What is the IP address of your Asterisk server?? |
| 07:37.21 | trymi1 | 192.168.30.1 |
| 07:37.32 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:37.46 | trymi1 | sorry |
| 07:37.53 | trymi1 | my IP is 192.168.15.25 |
| 07:37.58 | mort_gib | trymil: That's a private address, you need port forwarding on your router in order for incoming calls to work... |
| 07:38.35 | trymi1 | I tested with Trixbox |
| 07:38.48 | mort_gib | Trixbox is not supported here |
| 07:38.49 | trymi1 | all OK no need to do for port forwarding |
| 07:39.25 | trymi1 | now I am testing with asterisk |
| 07:39.32 | trymi1 | not OK for Incoming |
| 07:40.04 | trymi1 | If need port forwarding ,I need to forward when testing Trixbox |
| 07:40.39 | trymi1 | you know what I mean? |
| 07:40.47 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 07:41.18 | mort_gib | trymil: Yeah, I do, but you need port forwarding in order to receive incoming calls |
| 07:41.38 | trymi1 | OK |
| 07:42.50 | trymi1 | I have not router for my home ..I am using MY ISP gave IP |
| 07:45.08 | trymi1 | I can use direct login with Xlite to my provider server |
| 07:45.13 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:45.20 | *** join/#asterisk joobie (n=joobie@203-217-82-215.dyn.iinet.net.au) |
| 07:45.27 | trymi1 | and I can use In and Out with Xlite |
| 07:46.38 | mort_gib | Ok |
| 07:47.59 | trymi1 | Please check my conf is right or not |
| 07:51.40 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 07:52.23 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:54.06 | fcois93 | hello |
| 07:55.38 | *** join/#asterisk likwid8888545943 (n=chrislee@dt031n0d.tampabay.res.rr.com) |
| 07:56.54 | fcois93 | how to control if a header is present? is it right to do GotoIf($["${SIP_HEADER(x-client)}"]?1:2) ? |
| 07:59.50 | the_5th_wheel | Hi. Im getting this error in asterisk, and calls stop coming in eventually. Can anyone advise what I can do for this? Im using the latest bristuff1.4 with xorcom updates. It happens on bothe the offficial bristuff and with Xorcom. :chan_dahdi.c: Ring requested on channel 0/1 already in use on span 1. Hanging up owner." |
| 07:59.54 | likwid8888545943 | I am looking for a developer who can make a custom installation for me |
| 08:00.32 | tzafrir_laptop | the_5th_wheel, what type of device? |
| 08:01.32 | the_5th_wheel | tzafrir_laptop: Im using the junghans octobri cards |
| 08:01.54 | the_5th_wheel | Its happening on two different sites. I dont get this issue when running 1.2, but I need 1.4 for some AGI stuff |
| 08:02.20 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 08:03.09 | *** part/#asterisk trymi1 (n=please@203.81.71.107) |
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| 08:11.54 | the_5th_wheel | anyone/ |
| 08:17.36 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7a0aecf76654320e) |
| 08:18.33 | tzafrir_laptop | the_5th_wheel, is it an incoming call or outgoing call? |
| 08:18.42 | tzafrir_laptop | (the new call) |
| 08:26.33 | the_5th_wheel | tzafrir_laptop: Incoming |
| 08:26.43 | the_5th_wheel | tzafrir_laptop: I only do incoming calls |
| 08:35.53 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
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| 08:49.02 | kippi | hey |
| 08:49.06 | kippi | need some advise |
| 08:51.43 | kippi | I am runing openser and 5 asterisk servers, at the moment we are using sipsak to do the registrations to asterisk, is there away that I can use the openser user table for asterisk, so that asterisk is just doing a lookup to openser |
| 08:54.10 | ecret | can someone please try calling my asterisk with a sip phone? address is 99.246.65.73 . I am having issues with transfering the RTP of my DID's. |
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| 09:16.14 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 09:29.45 | *** join/#asterisk fc1979 (n=fc@93-63-253-26.ip30.fastwebnet.it) |
| 09:29.58 | fc1979 | Hi to all, i'm new in this chatroom |
| 09:30.25 | Mog | hi fc1979 |
| 09:31.44 | fc1979 | i join this chatroom because i want to help, and i need help :) |
| 09:31.56 | unk^ | wc |
| 09:31.58 | fc1979 | sorry for my poor english. |
| 09:32.59 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
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| 09:38.48 | angryuser | i am the only person who received mess from haf- about * is a crap and freeswitch better ? |
| 09:39.08 | xrmx__ | angryuser, no |
| 09:40.11 | angryuser | i think he dont understand that he does more harm to the project with that messages |
| 09:40.53 | tzafrir_laptop | angryuser, if you happen to be a LWN subscriber, see http://lwn.net/Articles/322487/ |
| 09:41.00 | tzafrir_laptop | Talks exactly about such things |
| 09:41.43 | tzafrir_laptop | (if not, you'll have to wait till thursday or so to read it, I believe) |
| 09:41.48 | angryuser | tzafrir_laptop, thank you i will look at it |
| 09:43.51 | fc1979 | no, you're no te only person.. |
| 09:45.17 | fc1979 | i've a problem with asterisk realtime configuraton. Can i ask here? |
| 09:45.33 | angryuser | ~question |
| 09:45.34 | jbot | i heard question is If you have a question and want people to give useful answers, make sure you have read this first: http://www.catb.org/~esr/faqs/smart-questions.html |
| 09:45.38 | tzafrir_laptop | angryuser, I consider anybody who does that a spammer. And frankly it makes me far less confident of a project that is advertized through spamming |
| 09:46.17 | tzafrir_laptop | Regardless of the actual quality of the project |
| 09:47.02 | angryuser | tzafrir_laptop, same for me, BUT you need to consider an option where someone who is hating freeswitch doing that, our reaction could be predicted ;) |
| 09:47.32 | tzafrir_laptop | I'm not on #freeswitch. Anybody complained there? |
| 09:47.52 | fc1979 | Ok, i know that document. I configure asterisk with dynamic realtime, backend odbc. The connection is up. In asterisk console i can read correctly the extension. |
| 09:48.57 | tzafrir_laptop | fc1979, err... what do you ask, exactly? |
| 09:49.24 | fc1979 | :) i'm writing now the question.. |
| 09:49.38 | fc1979 | i speak english very bad :) |
| 09:49.54 | fc1979 | For example: i'va a context "test". In the database extension table is set correctly. |
| 09:50.42 | fc1979 | i want to check if the extension is loaded well: |
| 09:50.44 | fc1979 | itpbx01*CLI> realtime load extensions context prova |
| 09:50.53 | fc1979 | sorry: itpbx01*CLI> realtime load extensions context test |
| 09:51.20 | fc1979 | the result is correct. So the connection it's ok and the extension is loaded correctly. |
| 09:51.29 | fc1979 | The quesiotn: |
| 09:52.02 | fc1979 | when i try to call the number, the result is "rejected because extension not found". |
| 09:52.07 | fc1979 | What can i check? |
| 09:52.56 | obruT | hello ! does anybody know is it possible to set teleservice type on the PRA link ? I got asterisk box with digium clone card, it's connected to PSTN via PRA link, everything work ok, I use it for faxing, but I should set teleservice type to "Fax group 2/3", I googled but I can't find any documentation about setting it... |
| 09:54.06 | fc1979 | this is the values of the record that match my extension : context: prova exten: 1000 Priority: 1 app: Playback appdata: tt-weasels |
| 09:54.33 | fc1979 | s/prova/test :) |
| 09:54.57 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:54.59 | SwK | angryuser, still here? |
| 09:55.49 | EmleyMoor | fc1979: what kind of phone? (just the technology will do) |
| 09:56.03 | fc1979 | it's a softphone, X-Lite |
| 09:56.20 | EmleyMoor | OK - pastebin your sip.conf |
| 09:56.23 | fc1979 | on the display appear: 404 not found. |
| 09:56.26 | EmleyMoor | ~pb |
| 09:56.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 09:57.06 | SwK | jbot seen haf- |
| 09:57.08 | jbot | SwK: i haven't seen 'haf-' |
| 09:57.21 | agx | is there some know problem about G711.A generating metallic voice? |
| 09:57.49 | fc1979 | i've users.conf configured for the users, not sip.conf, anyway.. |
| 09:58.19 | fc1979 | .[general] hassip = yes host = dynamic transfer = yes context = internal [1000] fullname = test context = test |
| 09:58.24 | fc1979 | this is users.conf |
| 09:58.29 | fc1979 | the sip.conf contains: |
| 09:58.59 | fc1979 | [general] type = friend realm = mydomain |
| 09:59.52 | fc1979 | with this configuration, without realtime it's ok, with appropriate extensions.conf configuration. |
| 10:00.24 | EmleyMoor | Hmmm... realtime issue then - no idea on that |
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| 10:03.03 | fc1979 | if the context of caller it's set to the same of receiver, asterisk daemon stop! |
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| 10:11.54 | Maliuta_ | did that moron PM everone in channel? |
| 10:12.04 | EmleyMoor | What moron? |
| 10:12.20 | SwK | Maliuta_, he PMd you too? |
| 10:12.33 | Maliuta_ | looks like it was targeted |
| 10:12.47 | SwK | yeah cause I didnt get a PM from him |
| 10:12.53 | SwK | (probably cause I'm on #fs) |
| 10:13.28 | gambler1 | Hi, I does anyone know, how asterisk performs concatenation while using EVAL function? To me it seems that this does not work as I expect: Set(fb=${EVAL($[LCR${BACKUP}])}) produces LCR2 insetead of value of LCR2 |
| 10:14.15 | EmleyMoor | "Thing A" instead of "Thing A"???? |
| 10:14.32 | EmleyMoor | Ah... hold on |
| 10:14.50 | SwK | gambler1, add ${} around LCR${BACKUP} |
| 10:15.12 | EmleyMoor | Yes, that's right |
| 10:15.14 | SwK | that will get you the value of instead of "LCR2" |
| 10:15.23 | gambler1 | SwK: will try that, thank you |
| 10:15.24 | SwK | assuming ${BACKUP} == 2 |
| 10:16.14 | SwK | and I dont think you really need the eval there unless you are doing math or something inside LCR${BACKUP} |
| 10:16.39 | SwK | Set(fb=${LCR${BACKUP}}) is probably all you really need |
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| 10:18.41 | gambler1 | SwK: I just want to perform jump on backup provider, but I am not sure whats the best way to achive this, as in the same manner I would like to keep least cost routing |
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| 10:19.59 | SwK | gambler1, that really depends on how your LCR works... and you would want to probably pastebin that so someone could take a look to better advise if you wanted help there... |
| 10:20.44 | SwK | but usually something like a loop and check the DIALSTATUS after the DIAL() is all you need to do then advance the counter and go to the next route |
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| 10:22.17 | gambler1 | SwK: I do just like that, BACKUP is the counter and LCR1 LCR2 etc hold the named entries in dialplan where to jump in a case of call failure + loop prevention |
| 10:23.11 | c0rnoTa | hello all. can anybody tell me, where i can fined inforation about zttool usage? i've searched in google about zttool, but couldn't find anything about debug information. Only rhino support hase documentation, but it's not full |
| 10:23.31 | gambler1 | SwK: thank you for your support |
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| 10:34.27 | jaap__ | hi! |
| 10:35.05 | jaap__ | trying to detect if an external number is busy after using the flash() command |
| 10:35.21 | jaap__ | has anybody ever tried that before? |
| 10:35.47 | jaap__ | busydetect() works, but it hangs up the flashed line and the incoming connection at the same time |
| 10:36.37 | jaap__ | the idea is to play some music, wait for the extension to be free and when it is, flash the line again to that number |
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| 10:56.26 | tokozedg | hi, i am trying to record calls, with mixmonitor, and for filename i want to have sip clients usernames, which answered the calls, which variable can make so? |
| 10:59.48 | c0rnoTa | if you call to a specific extensions (special for this SIP peer), ${EXTEN} is your variable |
| 11:00.29 | c0rnoTa | if you call to a group or queue, you can use 'post command' option in mixmonitor to rename file |
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| 11:05.44 | Gido-E | sometimes my Asterisk server does not restart if i issue a restart from the CLI. I need todo a killall -9. ANybody a idea? |
| 11:08.06 | Chainsaw | It might be doing a graceful restart Gido-E. |
| 11:08.36 | Chainsaw | That would mean a call is still in progress, and it's waiting for that to complete. Issue a restart now if you don't care about phone calls that are in progress. |
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| 11:09.25 | c0rnoTa | or it may be deadlocked |
| 11:09.25 | Gido-E | Chainsaw nope, I do a restart. |
| 11:09.32 | Gido-E | not a gracefull. |
| 11:10.32 | Gido-E | c0rnoTa maybe a deadlock. But there is not realy something special on the server happening . |
| 11:10.48 | Gido-E | just a lot of sip and a flat dialplan. |
| 11:12.42 | Aurs | Gido-E: "restart now" ? |
| 11:13.20 | Gido-E | Aurs yes |
| 11:13.32 | Gido-E | but, i think i will upgrade to the latest asterisk first. |
| 11:13.39 | Gido-E | now on. 1.4.22.2 |
| 11:13.46 | Aurs | just checking... "restart" won't do anything so.. :) |
| 11:14.02 | c0rnoTa | look in to full log file with debug. if there too much ERRORs - that's the answer :) But i couldn't restart asterisk when i use srvlookup and voip operator like sipnet.ru. while it resolving ip it freezs |
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| 11:32.07 | *** join/#asterisk [psy] (n=psy0rz@lounge.datux.nl) |
| 11:32.54 | [psy] | i get a: "chan_sip.c:5532 process_sdp: No compatible codecs, not accepting this offer.", but the call is eshtablished anyway? i want the call to fail, because i'm writing an automated regression test..how to do that? |
| 11:37.25 | [psy] | i think its because it does packet2packet bridging...how to disable that? |
| 11:38.08 | c0rnoTa | canreinvite =no ? or may be directrtp=no? |
| 11:39.02 | *** join/#asterisk shnootz (n=hanan@bzq-219-113-98.static.bezeqint.net) |
| 11:39.26 | [psy] | i already have them both on no |
| 11:39.49 | shnootz | hi |
| 11:39.55 | [psy] | packet2packet bridging means the packets go though asterisk, but are not processed or transcoded |
| 11:40.23 | [psy] | so its not the same as directrtp=yes anyway |
| 11:40.41 | shnootz | does anyone knows a good workaround for the issue that when the internet drops the asterisk freezes? |
| 11:41.09 | [psy] | freezes in what way? |
| 11:41.39 | shnootz | cann't make calls even between the internal phones |
| 11:42.10 | [psy] | but when you do astersik -r, you still can get into the cli and do stuff? |
| 11:42.15 | c0rnoTa | [psy]: what tells you 'sip set debug on' for peers? are audio codecs combined? |
| 11:42.43 | c0rnoTa | why it tells 'no cmpatible' |
| 11:42.45 | c0rnoTa | ... |
| 11:43.03 | shnootz | [psy] yes |
| 11:44.34 | [psy] | c0rnoTa http://pastebin.com/m1b470b1b |
| 11:44.49 | [psy] | shnootz dunno then |
| 11:45.00 | shnootz | i have seen that it is a known bug but everything i am trying doesn't help |
| 11:45.05 | [psy] | ah k |
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| 11:46.35 | [psy] | oh i think i know..the client accepts all codecs, so asterisk probably concludes it can handle the codec and uses packet2packet |
| 11:47.42 | shnootz | any ideas? does anyone here has the same problem? |
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| 12:10.56 | aiksa[LV] | hello everyone. |
| 12:11.44 | aiksa[LV] | I was wondering - are there eclipse plugin which would feature syntax highlighting and auto-complete for asterisk? |
| 12:13.16 | aiksa[LV] | with the size of the user base for the both I suspect that there should be at least couple of such projects. Yet I am unable to find any except for asterisk config file modelling plugin, which is not exactly what I would like. |
| 12:15.16 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:16.57 | aiksa[LV] | I mean - there are several versions of syntax highlighters for vi and vim, why cant I find any for eclipse ..;) |
| 12:18.59 | mvanbaak | aiksa[LV]: because noone created one |
| 12:21.57 | aiksa[LV] | mvanbaak: old asterisk maillist entry suggests that someone started to work on that |
| 12:22.47 | aiksa[LV] | however i cant find any trace of that project - as if it never existed (perhaps it didnt) |
| 12:27.34 | coppice | the world is full of projects what died without trace when the author found they needed actual work. most of the in "pure Java" :-\ |
| 12:27.35 | mvanbaak | maybe they stopped using eclipse to edit their configs |
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| 12:29.54 | *** join/#asterisk bobsaccamano (n=ymax@203.126.136.142) |
| 12:30.34 | bobsaccamano | how do i send alter_info header in a sip invite |
| 12:30.34 | bobsaccamano | ? |
| 12:30.43 | bobsaccamano | i want to configure it in the dialplan |
| 12:31.30 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 12:32.57 | [TK]D-Fender | bobsaccamano: "core show applications like sip" <- look around a bit, I'm sure it'll come to you |
| 12:33.48 | shnootz | internal sip phones stop communicating while internet connection is loss, any ideas? |
| 12:35.21 | c0rnoTa | shnootz: use tshark to dump pockets on server, and look, why you phones does not send sip pockets on asterisk |
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| 12:38.44 | aiksa[LV] | coppice: sad but true |
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| 12:38.55 | shnootz | c0rnoTa sorry but how do i use tshark? |
| 12:38.56 | *** join/#asterisk ct529 (n=ct529@77-44-78-159.xdsl.murphx.net) |
| 12:39.35 | aiksa[LV] | any other desktop (pref. win and linux) code editor with syntax highlighting for asterisk config files? |
| 12:39.52 | ct529 | hi everybody .... I am trying to start a small asterisk server to do some experiments, to decide whether to adopt it for our company .... my base is kubuntu 8.10, and I am using thirdlane pbx manager for the ste up .... |
| 12:40.05 | ct529 | is that a good combination, or am i starting with the wrong foot? |
| 12:41.25 | c0rnoTa | shnootz, you can use sip debug at first |
| 12:41.44 | c0rnoTa | shnootz, look, does your server receive pockets from phones |
| 12:43.15 | ct529 | or is it better if I use destar to configure? |
| 12:43.46 | russellb | ct529: do you care if it's asterisk, or do you just need a new phone system? |
| 12:43.55 | russellb | typically, if you go the GUI route, you're limited to what the GUI provides |
| 12:44.14 | russellb | where as configuring asterisk directly is more difficult, but provides _much_ more flexibility |
| 12:44.17 | russellb | just depends on your needs. |
| 12:44.25 | ct529 | russellb: as long as it is open source it is ok .... |
| 12:44.42 | russellb | you mean that the GUI is open source? |
| 12:45.13 | *** join/#asterisk WindBack (n=WindBack@201.216.217.129) |
| 12:45.16 | russellb | if you want to go that route, I would recommend trying out FreePBX. Go download AsteriskNOW 1.5 and give that a try. |
| 12:45.22 | ct529 | russellb: I want to "serve" phone calls for our very small company .... a provider provides us a SIP line .... I wanted to send it into our local asterisk and serve the lines to people from there |
| 12:45.28 | russellb | nods |
| 12:45.34 | russellb | give *NOW a try |
| 12:45.52 | ct529 | russellb: what is FeePBX? and alternative to asterisk? |
| 12:46.00 | coppice | I tried now. I'm waiting for the future |
| 12:46.03 | russellb | no, FreePBX is a gui for asterisk. |
| 12:46.16 | russellb | coppice: marketing is silly. :-) |
| 12:46.18 | ct529 | russellb: do you think I can do what I want using asterisk? |
| 12:46.24 | russellb | ct529: yes |
| 12:46.28 | ct529 | russellb: thanks! |
| 12:46.59 | ct529 | russellb: we would also need waiting music and recording some phone calls .... can we also do that? |
| 12:47.24 | c0rnoTa | ct529 ofcourse |
| 12:47.28 | russellb | yup. |
| 12:47.33 | russellb | that's easy stuff :-p |
| 12:47.41 | ct529 | russellb: c0rnoTa: thanks again .... |
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| 12:50.02 | ct529 | russellb: I can install freePBX without installing AsteriskNOW, is that right? |
| 12:50.24 | russellb | you can, but it's just more difficult that way |
| 12:50.56 | ct529 | russellb: oh .... but I do not have a server dedicated to asterisk only .... |
| 12:51.00 | russellb | if you go with something like asterisknow, it is packaged and updates are provided for you |
| 12:51.03 | russellb | ah. |
| 12:51.12 | ct529 | russellb: what is wrong with the Thridlane Webmin module? |
| 12:51.22 | russellb | I do not have experience with it. |
| 12:51.37 | ct529 | russellb: well, if everything works, we will move it to a dedicated online server .... is it possible to do that? |
| 12:52.07 | russellb | also, you should check out switchvox. It's not open source, but if you're looking for a gui, it's quite nice. (my company provides switchvox) |
| 12:52.32 | russellb | it's possible, but could be a pain, depending on the GUI |
| 12:52.43 | russellb | just have to make sure you move all of the right pieces |
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| 12:55.14 | ct529 | russellb: do you think it is possible to run asterisk successfully on an online server, on a virtual machine? |
| 12:56.02 | anonymouz666 | nice morning interview :-) |
| 12:56.11 | russellb | ct529: people do it all the time, yes |
| 12:56.35 | ct529 | russellb: thanks |
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| 12:56.54 | ct529 | russellb: right, I goo and read documentation on freePBX, thanks a lot see you later |
| 12:57.01 | Eduardo_Assis | anonymouz666, Hi |
| 12:57.12 | russellb | ct529: good luck |
| 12:57.20 | anonymouz666 | Eduardo_Assis: are you training your english or what? |
| 12:57.24 | anonymouz666 | he he |
| 12:57.28 | Eduardo_Assis | he he |
| 12:57.30 | ct529 | crosses fingers |
| 12:58.58 | anonymouz666 | Eduardo_Assis: if you turn ON your TV and watch CNN (and understand what they say) then you will be able to talk and understand everyone in here personally :-) |
| 12:59.05 | anonymouz666 | I can't :p |
| 12:59.26 | Eduardo_Assis | anonymouz666, é o que estou tentando ... nem estou no asterisk-br agora ... |
| 12:59.35 | Eduardo_Assis | quero ficar um pouco aqui para praticar |
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| 13:00.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:00.18 | coppice | surely nobody can make sense of what CNN says |
| 13:00.19 | *** join/#asterisk propellerhead (n=yogurt2u@host215.190-138-92.telecom.net.ar) |
| 13:00.44 | Eduardo_Assis | coppice, why not ? |
| 13:01.13 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 13:01.58 | anonymouz666 | 'cause they don't understand T.38 spec (just to be ontopic) :p |
| 13:01.59 | Eduardo_Assis | CNN formers opinions |
| 13:04.07 | coppice | anonymouz666: I wonder what drives that uncontrollable urge to spout rubbish (does that bring us back to CNN?) |
| 13:04.26 | *** join/#asterisk goodjoke (i=4a4359ab@gateway/web/ajax/mibbit.com/x-386cdc85d64a9247) |
| 13:05.22 | c0rnoTa | anonymouz666, excellet phrase! :)) |
| 13:05.48 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:07.10 | *** join/#asterisk shakal__ (n=shakal@80.92.229.58) |
| 13:07.22 | *** join/#asterisk cbullock81 (n=cbullock@adsl-251-32-49.jan.bellsouth.net) |
| 13:10.09 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 13:10.10 | Eduardo_Assis | The Digium Corp cards gsm want to launch? |
| 13:10.36 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 13:10.54 | kaii | i'm trying to handle blind transfer in my dialplan, to do a boomerang effect (callback) when the transfer destination is busy or times out .. |
| 13:12.31 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:12.42 | kaii | but i dont know how to find out which extension initiated the blind transfer .. there is a variable ${BLINDTRANSFER} for this purpose, but it seems to me this is only used when the blindxfer from features.conf is used. in my case, the transfering sip client answers with "302 moved temporarily" |
| 13:13.24 | kaii | asterisk then creates a totally new call with new channels (local channel) and new unique id |
| 13:13.45 | kaii | any suggestions how i could find out which extension sent the 302 redir ? |
| 13:14.06 | angryuser | kaii, try to save the source extension into $var and do whatever you want after |
| 13:16.29 | angryuser | lets say extension 300 pas a call to extension 301, for 301 save the source and if busy goto(300) |
| 13:16.48 | angryuser | goto$(${var}) |
| 13:16.56 | angryuser | goto(${var})* |
| 13:17.50 | *** join/#asterisk ingenius (n=alektro@247.175.73.200-static.serversur.net) |
| 13:18.57 | SuPrSluG | i'm trying to get AMD (answering machine detection) working well. I works pretty reliably with normal extensions phones etc.., but doesn't seem to recognize human voice so well with cell phones. Which parameter should I be looking at? |
| 13:19.00 | angryuser | kali you can use the same macro for everyone, then let's say create ivr where people modify the values for that macro (transfer or not(where?), voicemail etc) and you just need to dud a couple check before actually calling that exten |
| 13:19.24 | angryuser | kali oh, and use DB() for that |
| 13:21.49 | kaii | angryuser: have you thought of the possibility of having multiple active calls and a race condition about $var ? |
| 13:21.57 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 13:22.20 | *** join/#asterisk mphill (n=mphill@174.37.19.92-static.reverse.softlayer.com) |
| 13:22.25 | angryuser | kaii i cant think about everything for you |
| 13:22.31 | angryuser | ;) |
| 13:22.40 | kaii | well fact is, i am already that far. |
| 13:23.12 | kaii | but i need a unique identifier name for the $var which is reproducable .. (in the originating channel and the new channel after transfer) |
| 13:23.28 | kaii | and since the channel is new (new call) i can not use $CHANNEL or $UNIQUEID as identifier |
| 13:23.33 | angryuser | kaii you can evaluate the number of concurrent call with groups and groups count |
| 13:23.51 | kaii | angryuser: to what extend? |
| 13:24.34 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:25.07 | angryuser | kaii to dot whatever you want when your limit is reached, and i dont see why you would need $UNIQUEID |
| 13:25.09 | kaii | angryuser: i have a macro, lets say "docall()" .. it is first initiated by SIP/10 calling SIP/11 .. |
| 13:25.30 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:25.39 | kaii | SIP/11 then sends 302 moved to 20 and asterisk creates a new channel from SIP/10 to Local/20@context |
| 13:26.15 | kaii | angryuser: this is not about limits. this is about identifying the originating extension of that transfer |
| 13:26.41 | *** join/#asterisk ingenius (n=alektro@247.175.73.200-static.serversur.net) |
| 13:27.38 | angryuser | kaii, you have many people/devices who is tellling you moved to ? |
| 13:28.40 | c0rnoTa | it's call forwarding on phone i thinks |
| 13:28.44 | c0rnoTa | think |
| 13:28.50 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:28.54 | kaii | SIP/11 calls SIP/12, SIP/12 answers "302 moved to 20" and asterisk bridges SIP/11 to a new channel Local/20@context |
| 13:28.59 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 13:29.10 | kaii | c0rnoTa: yes thats what i am saying. SIP 302 moved temporarily |
| 13:29.14 | c0rnoTa | yes, it is callforwarding throu phone feate |
| 13:29.15 | kaii | the client redirects, not asterisk. |
| 13:29.55 | c0rnoTa | sorry, i does not read your dialog fully |
| 13:29.55 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 13:29.57 | kaii | but how can i detect in the new channel Local/20 that there was a 302 redir and find out which extension initiated this? |
| 13:30.15 | *** join/#asterisk mosty (n=mosty@213-66-224-163-no22.tbcn.telia.com) |
| 13:30.17 | kaii | i want to route the caller back to the originating ext when the transfer target is busy or timeout |
| 13:30.29 | kaii | c0rnoTa: do you have an idea ? |
| 13:30.54 | *** part/#asterisk shnootz (n=hanan@bzq-219-113-98.static.bezeqint.net) |
| 13:31.01 | c0rnoTa | ofcourse |
| 13:31.06 | kaii | how ? |
| 13:31.21 | kaii | please do not say "save the originating ext in $var" |
| 13:31.22 | kaii | :-) |
| 13:31.34 | c0rnoTa | blindtransfer variable empty? |
| 13:31.38 | c0rnoTa | :)) |
| 13:32.26 | kaii | c0rnoTa: yes, empty. |
| 13:32.36 | c0rnoTa | may be __VARIABLENAME will help you? |
| 13:32.50 | kaii | why should? |
| 13:33.46 | kaii | inheritance does not seem to work in this place, because asterisk _creates a totally new call_ |
| 13:34.18 | c0rnoTa | your operator makes transfer to another person, ex on 20. 20 makes callforward to 12 and 12 is busy now, so client hangs up? |
| 13:34.44 | c0rnoTa | isn't it? |
| 13:35.02 | kaii | yes. |
| 13:35.56 | c0rnoTa | client bridged with local chan or sip chan? |
| 13:36.51 | c0rnoTa | so, first way in my head - using global variable for operator :) |
| 13:37.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 13:37.12 | kaii | SIP/11 then sends 302 moved to 20 and asterisk creates a new channel from SIP/10 to Local/20@context |
| 13:37.35 | WindBack | Hi, I have a TDM800p with 2 Quad FX0 modules. I'm using PAP2t in the extensions. I one of the ATAs I hava a FAX machine. Sometimes the outgoing lines used by this extensions keep unhanguped. Could be FAX machine my problem??? |
| 13:37.38 | kaii | c0rnoTa: yes thats how far i am now. but i need a unique identifier for the variable |
| 13:37.45 | kaii | to avoid race condition |
| 13:38.57 | c0rnoTa | WindBack, no. it's TDM800p problem. Look in tone zones |
| 13:39.33 | c0rnoTa | kaii, very interesting, but i have no idea |
| 13:39.52 | kaii | c0rnoTa: i'll report if i find a way |
| 13:40.05 | c0rnoTa | thx a lot. i'll think about in |
| 13:40.34 | *** join/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
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| 13:43.03 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 13:44.15 | WindBack | c0rnoTa, I have 4 outgoing lines, and 30 extensions. Some extensions go to the PSTN using lines 1, others using line 2, and the last group using lines 3 and 4 together. I have never had this problem in the 1 and 2 lines, But I always have the problem in the group 3 and 4. Also in the group of extensions which uses this group of lines (3 y 4) is the fax machine |
| 13:45.17 | WindBack | c0rnoTa, and the timezone is configured in the same way for all the lines |
| 13:45.45 | WindBack | c0rnoTa, so i dont think it is a problem in the tone zone |
| 13:46.02 | kaii | c0rnoTa: oh! inheritance DOES work! made a mistake while testing |
| 13:46.13 | c0rnoTa | kaii: excellent! ^))) |
| 13:46.31 | c0rnoTa | WindBack, have no idea, sorry |
| 13:48.57 | WindBack | c0rnoTa, thanks |
| 13:53.36 | kaii | c0rnoTa: but there is a bug .. :-( SIP/10 calls SIP/11, the timeout for Dial() is 15 sec. |
| 13:53.50 | kaii | when SIP/11 transfers, asterisk does not reset the timeout |
| 13:54.46 | *** join/#asterisk soylentgreen (n=fgast@missbehave.only640k.net) |
| 13:54.48 | kaii | 15 seconds after call is initiated, the timout occurs and the call is back in the originating channel .. (SIP/10-XY who dialed 11) |
| 13:54.53 | kaii | this sucks. |
| 13:55.22 | kaii | i want a transfered call to ring exactly what is specified in dial .. no what was specified in the dial before |
| 13:55.33 | kaii | (i hope this was somehow logic and understandable) |
| 13:56.18 | kaii | hrmpf. |
| 13:57.18 | *** join/#asterisk zaafouri (i=c4cb33ee@gateway/web/ajax/mibbit.com/x-a23bd732ed813c79) |
| 13:58.49 | *** join/#asterisk soylentgreen (n=fgast@missbehave.only640k.net) |
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| 14:09.36 | *** part/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
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| 14:24.57 | *** part/#asterisk c0rnoTa (n=c0rnoTa@78.24.154.158) |
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| 14:30.36 | BlargMaN00 | anyone in here using 1.6.1.0-rc2??? |
| 14:30.56 | *** part/#asterisk ctooley (n=ctooley@mo-71-0-218-176.dhcp.embarqhsd.net) |
| 14:31.12 | russellb | BlargMaN00: having trouble? |
| 14:33.38 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 14:36.24 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 14:37.31 | BlargMaN00 | russellb: yeah.. it seems as though MWI is broke in 1.6.1.0-rc2 |
| 14:37.58 | BlargMaN00 | it worked fine in rc1, and a straight upgrade to rc2, and now it doesn't work |
| 14:38.09 | BlargMaN00 | no config changes |
| 14:38.58 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 14:39.26 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
| 14:39.27 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:39.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:40.15 | iEatChildren | is there a way to reload asterisk without dropping calls? |
| 14:41.01 | areay | iEatChildren, like anyone's gonna answer you with a nick like that... |
| 14:41.13 | iEatChildren | most people dont care to be honest |
| 14:41.30 | areay | iEatChildren, i was kidding... reload from CLI shouldn't be dropping calls |
| 14:41.35 | iEatChildren | lol |
| 14:41.41 | iEatChildren | okay, wanted to check before i did anything |
| 14:41.43 | MaliutaLap | we all know that children, like kittens, belong in blenders |
| 14:41.44 | iEatChildren | thanks for your help |
| 14:41.47 | iEatChildren | lol |
| 14:41.51 | areay | lol, |
| 14:42.05 | soulclaimer | solves the food crisis and over population in one go |
| 14:42.15 | MaliutaLap | iEatChildren: and not that I know of ... |
| 14:42.28 | MaliutaLap | iEatChildren: unless you just want to reload one thing |
| 14:42.29 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:42.29 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:42.46 | russellb | BlargMaN00: can you open an issue on bug.digium.com with some more details on your configuration? |
| 14:42.53 | iEatChildren | MaliutaLap: thanks |
| 14:42.54 | MaliutaLap | iEatChildren: "dialplan reload" shouldn't drop calls |
| 14:43.02 | iEatChildren | ok |
| 14:43.05 | russellb | BlargMaN00: also, lmadsen is checking to see if there is a bug open for it already |
| 14:43.14 | MaliutaLap | iEatChildren: sip reload _might_ |
| 14:43.40 | areay | the only thing that'll drop calls is a /etc/init.d/asterisk restart |
| 14:43.44 | areay | afaik |
| 14:43.45 | BlargMaN00 | sure thing... |
| 14:43.50 | MaliutaLap | restart gracefully? |
| 14:43.59 | jjshoe | areay and restart now :P |
| 14:44.07 | russellb | BlargMaN00: please ping lmadsen and me after you open it |
| 14:44.08 | areay | jjshoe, ah yea ;) |
| 14:44.39 | BlargMaN00 | ok |
| 14:46.33 | MaliutaLap | russellb: whats a ping packet going to tell you? ;) |
| 14:46.58 | MaliutaLap | russellb: and are you going to notice it? ;) |
| 14:47.07 | russellb | heh |
| 14:47.18 | russellb | i meant like ... a message in here with our names in it :-) |
| 14:47.38 | MaliutaLap | oh, well why didn't you _say_ that? ;P |
| 14:48.00 | *** join/#asterisk ThatKidKel (n=Kelvin@208.110.55.5) |
| 14:48.01 | MaliutaLap | look! my arse is smart _and_ cute! |
| 14:48.17 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 14:48.23 | ThatKidKel | what is the current best solution for fax to email for asterisk? |
| 14:48.59 | lmadsen | ThatKidKel: you might want to look at hylafax for that |
| 14:50.18 | Nasra | !asterisk |
| 14:50.32 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 14:50.32 | MaliutaLap | ~asterisk |
| 14:50.33 | jbot | hmm... asterisk is a free PBX, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 14:50.40 | Nasra | thanks |
| 14:50.52 | MaliutaLap | ~book |
| 14:50.53 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:51.15 | Nasra | thanks |
| 14:51.27 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 14:51.28 | MaliutaLap | loves jbot |
| 14:52.15 | Nasra | question: how do I know what version of asteris I have on my box? |
| 14:52.19 | Nasra | thanks |
| 14:52.44 | BlargMaN00 | russellb & lmadsen: http://bugs.digium.com/view.php?id=14685 I marked it as a feature, cause i think it has more to do with the IMAP storage than anything... |
| 14:52.52 | BlargMaN00 | M14685 |
| 14:53.09 | russellb | ohhhh ... |
| 14:53.17 | russellb | thanks for the report |
| 14:53.20 | BlargMaN00 | yeah... |
| 14:53.28 | BlargMaN00 | np |
| 14:54.05 | BlargMaN00 | i'll be going back to 1.6.1-rc1 until some progress is made on the bug... 8)~ |
| 14:54.39 | lmadsen | BlargMaN00: thanks for the report -- I will attempt to test this afternoon |
| 14:55.00 | *** join/#asterisk henk (n=hank@netwichtig.de) |
| 14:55.01 | BlargMaN00 | cool |
| 14:55.04 | henk | moin |
| 14:55.14 | BlargMaN00 | i'll be around if you need me to do more testing as well... |
| 14:55.17 | henk | is AELv2 usable with asterisk 1.4.21? |
| 14:55.44 | lmadsen | BlargMaN00: sounds good -- I'm just going to test and verify, then will assign back to mmichelson to fix :) |
| 14:56.15 | BlargMaN00 | sounds good |
| 14:56.24 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 14:56.41 | *** join/#asterisk neurosys (n=vinix@173.9.159.182) |
| 14:57.16 | iEatChildren | im having a bit of toruble with the MixMonitor command...i have exten = s,7,MixMonitor(test.wav|av(0)V(0)) but it doesnt seem to append to teh end of the file |
| 14:58.34 | [TK]D-Fender | henk: Yes, AFAIK AEL1 only exists in 1.2 |
| 14:58.49 | iEatChildren | am i doing something wrong? the tutorial im following on asteriskguru.com said the a would make it append |
| 14:59.00 | [TK]D-Fender | iEatChildren: What ver of *? |
| 14:59.44 | iEatChildren | Asterisk 1.4.23.1 |
| 15:00.01 | *** part/#asterisk bob_slacker (n=vncsnvs_@189.114.27.40) |
| 15:00.36 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 15:00.43 | henk | [TK]D-Fender: oh ok, my last info was that AEL is still experimental... yes, i haven't had much to do with asterisk the last few months. is AEL the recommended configuration language? |
| 15:01.03 | [TK]D-Fender | henk: I wouldn't |
| 15:01.07 | lmadsen | prefers dialplan |
| 15:01.15 | [TK]D-Fender | henk: Recommended? Not really |
| 15:01.16 | lmadsen | but lots of others are using AEL2 |
| 15:03.24 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 15:03.46 | [psy] | we're getting this while regression testing Meetme: |
| 15:03.48 | [psy] | [Mar 17 16:03:14] WARNING[29087]: pbx.c:2661 ast_pbx_start: Failed to create new channel thread |
| 15:03.50 | [psy] | [Mar 17 16:03:14] WARNING[29087]: chan_sip.c:14693 handle_request_invite: Failed to start PBX :( |
| 15:03.50 | henk | ok, thanks! |
| 15:04.10 | iEatChildren | [TK]D-Fender: looks like it was just taking some time to append the file |
| 15:04.23 | iEatChildren | [TK]D-Fender: do you know if i can use mp3 instead of wav? |
| 15:04.26 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 15:05.48 | kaii | i love this "avoding initial deadlock" message .. |
| 15:07.30 | *** join/#asterisk stephank (n=shteef@212.178.158.35) |
| 15:08.24 | [TK]D-Fender | iEatChildren: You cannot natively record to MP3, but you can have mixmonitor call a script to convert it after |
| 15:09.37 | kaii | conversion to mp3 via sox is default since asterisk 1.2 AFAIK |
| 15:09.44 | kaii | you need to have the binary soxmix |
| 15:10.39 | stephank | Hello! I have a problem with transferring calls with 1.4 in some situations. It works fine normally, but now I have an application using AMI and FastAGI to control calls that uses the backported app Bridge (feature_bridge.patch, which is apparently default in the Debian package). However, after this bridge, I can no longer transfer, and my Aastras report "Transfer failed". |
| 15:11.06 | iEatChildren | [TK]D-Fender: here is something else i was wanting to try....http://pastebin.ca/index.php |
| 15:11.08 | [TK]D-Fender | kaii: There is no default conversion. |
| 15:11.21 | [TK]D-Fender | iEatChildren: ..... |
| 15:11.31 | iEatChildren | im not sure if thats doable... |
| 15:11.31 | kaii | lol |
| 15:11.41 | kaii | iEatChildren: paste the correct link. please. |
| 15:11.48 | iEatChildren | oh...sorry |
| 15:11.53 | iEatChildren | http://pastebin.ca/1363440 |
| 15:11.57 | iEatChildren | lol...my bad |
| 15:12.44 | iEatChildren | i am not sure if i can do it, but it would help me a ton if i could build the filename of the .wav file based on what the user inputs for the extension they type in |
| 15:12.58 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
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| 15:17.23 | littlerock | what's the method would you recommend to send fax? |
| 15:17.31 | [TK]D-Fender | iEatChildren: Integrate MixMonitor Somewhere else in your dialplan then. |
| 15:18.01 | littlerock | I want asterisk user can send fax outside |
| 15:18.31 | iEatChildren | [TK]D-Fender: sorry for all the questions... do you know of a resource that might help me with some of this programming...perhaps a list of asterisk variables or something? |
| 15:19.21 | [TK]D-Fender | iEatChildren: CHANNELVARIABLES in your DOC folder |
| 15:20.32 | jameswf | ~buybook |
| 15:20.32 | jbot | [~buybook] You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 15:21.29 | iEatChildren | ill check that out |
| 15:21.37 | iEatChildren | i think somewhere around here our last admin had one |
| 15:21.42 | iEatChildren | not sure if he took it... |
| 15:22.03 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:22.08 | kaii | iEatChildren: you should also checkout http://voip-info.org |
| 15:22.20 | iEatChildren | yup, just ran accross http://www.voip-info.org/wiki-Asterisk+variables |
| 15:22.39 | kaii | iEatChildren: huge wiki containing a lot of "best practise" and answers to most of your questions. (if you learn how to use the search feature) |
| 15:26.25 | *** join/#asterisk sack (n=sack@103.Red-79-148-191.dynamicIP.rima-tde.net) |
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| 15:44.49 | *** part/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 15:44.56 | BlargMaN00 | putnopvut: did you guys change the voicemail greeting storage location on 1.6.1.0-rc2???? |
| 15:45.11 | putnopvut | Not by default, no. |
| 15:45.21 | BlargMaN00 | is there an option for it?? |
| 15:45.25 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 15:45.56 | BlargMaN00 | cause i'm getting the following error now: [Mar 17 10:43:43] WARNING[9503]: app_voicemail.c:3686 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/1688/INBOX.WAV: No such file or directory |
| 15:45.59 | putnopvut | Sort of. You can change the astspooldir in asterisk.conf |
| 15:46.47 | putnopvut | It's possible that there is some bug which results in the wrong path used for the file. Did things work properly with 1.6.1.0-rc1? |
| 15:47.26 | BlargMaN00 | yep... no probs in rc1 |
| 15:49.46 | EmleyMoor | Does LCDial.sh intentionally not try a more expensive provider if the cheapest ones give BUSY? |
| 15:57.37 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
| 16:02.29 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 16:04.09 | henk | i tried to slim down the list of loaded modules, now asterisk 'dies' when trying to make a call with ekiga. i get no output concerning that... is there a list of minimally needed modules? or how can i find out what's wrong? |
| 16:07.36 | *** join/#asterisk nima0102 (n=nima@91.98.207.63) |
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| 16:10.48 | carrar | man |
| 16:11.00 | carrar | so many people spamming freebpx in here |
| 16:11.17 | Tommybj | "spamming" freepbx? |
| 16:11.52 | BlargMaN00 | putnopvut: how would I resolve this issue: |
| 16:11.53 | BlargMaN00 | checking for UW IMAP Toolkit c-client library... no |
| 16:12.28 | putnopvut | BlargMaN00: When you ran configure, did you specify --with-imap=/path/to/imap/source ? |
| 16:12.42 | BlargMaN00 | nope... just --with-imap |
| 16:13.03 | BlargMaN00 | it detects the c-client just fine... |
| 16:13.09 | Nugget | was never able to get imap storage to work reliably. asterisk turns into crash city whenever I enable it. |
| 16:13.23 | Qwell | Nugget: tried it lately? |
| 16:13.30 | Nugget | no, probably 8 months ago. |
| 16:13.41 | Qwell | it's gotten a lot better in the last 6 months or so, afaik |
| 16:13.43 | putnopvut | It still has crash issues, but not as many as it did then. |
| 16:13.53 | BlargMaN00 | i'm running it, with minimal issues... I still run into some stuff, but that's because I try to do way more with it than it was designed for... |
| 16:13.57 | Qwell | throws a carp at putnopvut |
| 16:14.05 | Nugget | good to hear, because it's a great solution. |
| 16:14.23 | Nugget | I'll give it another shot when 1.6.1 rolls out |
| 16:15.17 | putnopvut | BlargMaN00: I'm getting a bit confused. You said "it detects the c-client just fine" but you also printed a line from the configure script showing that it wasn't found. |
| 16:15.45 | putnopvut | Plus I can't think of what could have changed in the configure script between 1.6.1.0-rc1 and 1.6.1.0-rc wrt IMAP. |
| 16:15.57 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 16:16.58 | neurosys | l |
| 16:17.07 | BlargMaN00 | checking for UW IMAP Toolkit c-client library... no |
| 16:17.07 | BlargMaN00 | checking for system c-client library...... yes |
| 16:17.29 | BlargMaN00 | not sure what the diff is... |
| 16:17.35 | putnopvut | ah, I see. |
| 16:17.40 | putnopvut | Okay, I can explain. |
| 16:17.49 | BlargMaN00 | is all ears |
| 16:17.55 | putnopvut | First, it checks to see if you have a source checkout of the c-client library. |
| 16:18.20 | putnopvut | Then, if it doesn't find that, it will look in specific locations that are common places where distros will install their c-client packages. |
| 16:18.43 | BlargMaN00 | ahh... so it finds it, just not the checkout version... correct?? |
| 16:18.49 | putnopvut | Correct. |
| 16:18.53 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 16:19.01 | BlargMaN00 | gotcha... it just kinda threw me off... |
| 16:19.32 | BlargMaN00 | i thought maybe it had something to do with why my IMAP was acting funky in rc2 |
| 16:24.53 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 16:26.36 | *** join/#asterisk DavidR2008 (n=chatzill@fw1.safedataisp.net) |
| 16:29.22 | DavidR2008 | I've got a quick Q: I'm looking at phones for a VoIP rollout, when I did some research about a year or two ago, I heard some comments that Aastra supported Asterisk more then other vendors. (as far as investing back into the project, etc.) I would like to invest my hardware dollars in to a vendor that actively supports asterisk if there is one. Are there vendors that support asterisk in this... |
| 16:29.24 | DavidR2008 | ...way now? |
| 16:32.18 | [TK]D-Fender | DavidR2008: Polycom > All |
| 16:32.42 | DavidR2008 | really? I didn't think that used to be true |
| 16:35.21 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:36.47 | CrazyTux | [TK]D-Fender: my only thing with Polycom is that they take a day and half to boot :) and they do not support NAT |
| 16:38.05 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 16:47.13 | kaii | hey oej |
| 16:48.20 | kaii | oej: just 5 seconds ago i wrote a patch to find out who originated a transfer/forward in the next channel .. (sip client sends 302 redirect, asterisk opens new local channel .. in this local channel i needed to know who was originally called, e.g. who initiated the forward) |
| 16:48.46 | kaii | oej: it works fine with Dial() but i would love to know if there is ANY OTHER WAY except patching the code ... |
| 16:48.59 | kaii | i inserted a "FORWARDEDBY" channel variable by app_dial |
| 16:49.22 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
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| 16:54.52 | [TK]D-Fender | CrazyTux: Yeah, they're OK with NAT if you're smart about it, and mine take < 2 minutes, and if you have tor eboot every day then YOU have issues |
| 16:55.35 | CrazyTux | [TK]D-Fender: haha agreed :) -- I just remember when doing some development / testing with them it was a PITA |
| 16:58.47 | DavidR2008 | CrazyTux: I've been testing with an Aastra 9133i and it takes about a minute and a half to boot |
| 17:02.38 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 17:05.05 | rue_mohr | greetings all |
| 17:05.19 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-3923d3818748125d) |
| 17:09.20 | *** join/#asterisk rvhi (n=chatzill@udp100194uds.hawaiiantel.net) |
| 17:10.13 | rue_mohr | haha, I forgot all my sql commands |
| 17:10.16 | rue_mohr | sigh... |
| 17:10.36 | *** join/#asterisk NirS (n=NirS@87.70.157.67) |
| 17:10.38 | NirS | hello all |
| 17:10.43 | rue_mohr | helllo |
| 17:11.00 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:11.04 | rue_mohr | has anyone done postgres integration with asterisk? |
| 17:11.07 | NirS | say, anyone where are the AsteriskNOW 1.5 beta RPM packages are located ? |
| 17:11.14 | rue_mohr | looking at using it for speed dials |
| 17:11.35 | rue_mohr | NirS, your prolly best to download and compile from source :) |
| 17:12.08 | NirS | rue, after spending 6 years of compiling Asterisk from source, I'd like to try some RPMS ;-) |
| 17:12.22 | rue_mohr | if I say more its going to become daftly apparent I hate redhat |
| 17:12.30 | NirS | after all, I did write the PacktPublishing book about Asterisk AGI programming, so I do know how to compile it ;-) |
| 17:12.46 | rue_mohr | ok |
| 17:13.08 | NirS | feels that was too much |
| 17:13.09 | rue_mohr | did your book cover databases? |
| 17:13.15 | NirS | no |
| 17:13.20 | rue_mohr | sigh... |
| 17:13.26 | NirS | well, I did mention how to work with MySQL |
| 17:13.32 | NirS | but other than that, nothing specific |
| 17:13.39 | NirS | and Postgress isn't my forte |
| 17:13.43 | *** join/#asterisk ZeNN (n=abc@ip224-160-173-82.adsl2.static.versatel.nl) |
| 17:13.46 | NirS | I don't really like its management methodology |
| 17:13.47 | rue_mohr | mysql isn't mine |
| 17:13.53 | NirS | Too orcaley for me |
| 17:14.09 | rue_mohr | 7 years ago I looked at both and made a decision to go with postgres |
| 17:14.12 | NirS | is an Oracle 8i DBA - and yet still can't stand Oracle |
| 17:14.33 | rue_mohr | sometimes you have to know someting to truely hate it |
| 17:16.12 | rue_mohr | wait, shoudl I use postgesql for the database or astdb? |
| 17:16.28 | *** join/#asterisk bob_slacker (n=vncsnvs_@189.114.27.40) |
| 17:16.48 | NirS | wow, what are you trying to do exactly ? |
| 17:16.57 | rue_mohr | do I need to use agi to interface to a sql database? |
| 17:17.00 | NirS | AstDB isn't a relational database and isn't meant to be used as one |
| 17:17.07 | rue_mohr | I need to make a speed dial system |
| 17:17.32 | NirS | In general the answer is yes, unless you use MySQL which has an in dialplan connector, although, I don't like using that |
| 17:17.35 | rue_mohr | I wanted to use postgres partly because I can make a php front end to it for programming in and editing numbers |
| 17:17.43 | NirS | you can also use ODBC directly from the dialplan, but that requires some hops |
| 17:17.58 | NirS | SpeedDial? |
| 17:18.00 | rue_mohr | yea |
| 17:18.05 | rue_mohr | #23 |
| 17:18.06 | NirS | How big a directory ? |
| 17:18.22 | rue_mohr | oh well they said I could get away with 10 slots |
| 17:18.24 | rue_mohr | :) |
| 17:18.38 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 17:18.45 | rue_mohr | it stems from a lack of buttons on the polycom phones |
| 17:19.10 | NirS | in that case AstDB will do just fine |
| 17:19.18 | Maliuta | PostgreSQL > MySQL |
| 17:19.19 | NirS | and you can manipulate it from AGI and from AMI as well |
| 17:19.23 | rue_mohr | but then I cant make a php interface toedit it |
| 17:19.28 | NirS | sure you can |
| 17:19.33 | NirS | use AMI or AJAM |
| 17:19.45 | rue_mohr | hmm I'v never used those |
| 17:19.57 | rue_mohr | maliuta :) |
| 17:20.16 | NirS | well, you should |
| 17:20.26 | NirS | these are two of the most usefull tools on Asterisk |
| 17:20.53 | rue_mohr | I need to get it done, if I can stay off learning curves, its good. |
| 17:20.57 | Maliuta | rue_mohr: oh, and I can't write for "X" draws a blank, unbelieving stare from me |
| 17:21.17 | rue_mohr | I just said I havn't |
| 17:21.26 | NirS | well, these are not all that complicated |
| 17:21.30 | rue_mohr | I'v done heaps and heaps of postgres/php pages |
| 17:21.34 | Maliuta | it's a freakin' API, you only need a reference in front of you |
| 17:21.39 | NirS | or, you can write an AGI to do this |
| 17:21.52 | NirS | simply write an AGI that creates the speed dials by entering them via DTMF |
| 17:21.55 | Maliuta | postgres with stored procedures == da shite |
| 17:22.17 | rue_mohr | its just a simple 2 column table with a simple query |
| 17:22.24 | NirS | rue, take a look at PHPAGI, it takes the edge out - or you can probably find my book somewhere on Bittorrent |
| 17:22.33 | NirS | it has a full episode on AMI programming and AGI programming |
| 17:22.33 | rue_mohr | oh hmm |
| 17:22.45 | rue_mohr | I dont have time for episodes |
| 17:22.50 | NirS | unless you can purchase an online copy |
| 17:22.57 | NirS | sorry, no quick fix here |
| 17:23.05 | rue_mohr | I have about 1.5 hours to try to get this working :) |
| 17:23.18 | rue_mohr | I dont think I will, but thats what I have |
| 17:23.56 | NirS | you're trying to sqweeze a task that would take a beginner about 3 days into 1.5 hours - ambitious |
| 17:24.16 | rue_mohr | well this is covered in the asterisk manual, looks like their using odbc |
| 17:24.50 | NirS | well, ODBC will do it |
| 17:25.50 | rue_mohr | trying to look ahead into how asterisk uses the database |
| 17:27.47 | rue_mohr | it looks rather canned |
| 17:27.58 | Maliuta | rue_mohr: for what? cdr? |
| 17:28.13 | rue_mohr | the way it uses the database |
| 17:28.44 | rue_mohr | I dont see that I cn just have it query the database for the dial string for speedial index 4 |
| 17:30.22 | rue_mohr | oh there we go |
| 17:30.41 | Maliuta | rue_mohr: res_pgsql? cdr_pgsql?? |
| 17:31.03 | rue_mohr | er, I'm looking at the hotdesk example in THE BOOK :) |
| 17:31.24 | rue_mohr | I see how i can do it in a dialplan |
| 17:31.28 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 17:31.49 | rue_mohr | looks like its res_ |
| 17:32.01 | rue_mohr | but res_odbc |
| 17:32.30 | Maliuta | right so realtime config mojo |
| 17:32.38 | rue_mohr | yep |
| 17:33.29 | Corydon76-dig | Uh, that's func_odbc |
| 17:33.39 | rue_mohr | shrugs |
| 17:33.39 | Corydon76-dig | relies on res_odbc for connection pooling |
| 17:33.42 | Maliuta | and you've read http://www.asteriskguru.com/tutorials/realtime_pgsql.html ?? |
| 17:33.51 | rue_mohr | no i was reading the book |
| 17:34.11 | Maliuta | all hail The Book! |
| 17:34.26 | Maliuta | but it's just a jumping off point :) |
| 17:34.47 | Maliuta | rue_mohr: that link I pasted might help some |
| 17:34.50 | rue_mohr | wait, shoudl I do that page with the postgres, or what they have in the book? |
| 17:34.56 | [TK]D-Fender | Maliuta: If you stack them high enough.... |
| 17:35.51 | [TK]D-Fender | rueWhichever you feel like, Pick your DB and you means. |
| 17:35.55 | Maliuta | [TK]D-Fender: with some we can all but hope |
| 17:36.19 | jaytee | 4th Dimension on the MAC? |
| 17:36.23 | Maliuta | rue_mohr: well that page might give you an idea of what you can tweek in PG |
| 17:36.31 | rue_mohr | looks like the do the same stuff as the book in a different order |
| 17:36.40 | Maliuta | mmmm safron mash with parmesan and bacon |
| 17:36.47 | jaytee | bacon!!! |
| 17:36.50 | *** join/#asterisk Badrobot- (i=BadRobot@cpe-76-173-233-75.socal.res.rr.com) |
| 17:36.54 | Maliuta | safron!!! |
| 17:37.06 | rue_mohr | so whats a more reliable source, the book or that page? |
| 17:37.08 | jaytee | overpriced but nice spice |
| 17:37.23 | Maliuta | jaytee: blasphemer!\ |
| 17:37.30 | Maliuta | spot the foodie |
| 17:37.37 | outtolunc | $18.53 (this past weekend) |
| 17:37.50 | *** join/#asterisk nima0102 (n=nima@91.98.207.63) |
| 17:37.59 | rue_mohr | I'm gonna follow the book |
| 17:38.14 | jaytee | isn't saffron the most expensive of spices nowadays? just sayin! |
| 17:39.32 | nullable_type | Hello guys, Can anyone tell me the problem with the IF Statement: exten => 100,n,SET(REPORT_RESULT=${IF($[${REPORT_RESULT_LENGTH} = 1]?${REPORT_RESULT}:X)}) |
| 17:39.44 | Maliuta | jaytee: it's more expensive than gold |
| 17:40.02 | nullable_type | Basically i want REPORT_RESULT to be X if the REPORT_RESULT_LENGTH is greater than 1 |
| 17:40.17 | Maliuta | considering ou get 3 stamen per plant per year and it has to be hand picked .... |
| 17:40.56 | *** join/#asterisk Imo (n=Imo@brln-4db83a50.pool.einsundeins.de) |
| 17:41.01 | Imo | hello |
| 17:41.10 | Imo | i want record a phone call |
| 17:41.23 | Imo | how can i do this with asterisk ? |
| 17:41.47 | Maliuta | hold a tape recorder up to the handset? ;P |
| 17:41.50 | Qwell | Imo: show application record |
| 17:41.54 | Maliuta | record() |
| 17:42.22 | Imo | i have searched and i found monitor() |
| 17:42.26 | nullable_type | Qwell >> Can you look at the above IF Statement i posted, it is not working as expected |
| 17:42.31 | Maliuta | Qwell: you need to be a smart arse _first_ |
| 17:42.49 | Imo | but when i have a call, i want put for example 000 and now i want record |
| 17:42.52 | DavidR2008 | imo: http://www.voip-info.org/wiki-Asterisk+record+calls |
| 17:42.54 | DavidR2008 | look at the features.conf & the see also section |
| 17:43.42 | Imo | i have looked but i didnt found it |
| 17:45.07 | Maliuta | rue_mohr: http://hostseries.com/asterisk-realtime-installation-guide/ might also help |
| 17:45.14 | DavidR2008 | imo: if you enable automon features.conf and use the w & W options of Dial |
| 17:45.19 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-36-113.w86-215.abo.wanadoo.fr) |
| 17:45.24 | DavidR2008 | *automon in features.conf |
| 17:45.29 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 17:45.35 | Imo | DavidR2008: ??? |
| 17:45.58 | Imo | when i put * i can tranfser |
| 17:46.01 | Maliuta | that's not so hard to understand |
| 17:46.10 | seb- | Can any pinhead install Asterisk easily and do conference calls? What docs should i start w/? (I think the conf call component is called meetup!?) |
| 17:46.18 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-36-113.w86-215.abo.wanadoo.fr) |
| 17:46.19 | Imo | how the name in the feature.conf |
| 17:46.34 | Maliuta | seb-: judging from some of the ppl we see here ... es |
| 17:46.41 | seb- | heh |
| 17:46.45 | DavidR2008 | imo: what version of *? |
| 17:46.50 | Maliuta | s/es/yes/ |
| 17:47.02 | Imo | Asterisk 1.4.23.2 currently |
| 17:47.19 | Maliuta | seb-: you could start with ~book |
| 17:47.20 | seb- | Maliuta: is there a well written "quick start" doc somewhere? |
| 17:47.28 | Maliuta | ~book |
| 17:47.29 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:47.29 | rue_mohr | the book is completely out of date with postgres |
| 17:47.29 | DavidR2008 | Imo: I'm using 1.4.23.1 so my advise should close |
| 17:47.34 | DavidR2008 | *should be |
| 17:47.58 | seb- | Maliuta: ok...hopefully i don't need to read the whole phonebook to do a quick install |
| 17:48.24 | Maliuta | seb-: reading the book is a good starting point |
| 17:48.38 | Maliuta | seb-: and helps you from getting beaten up |
| 17:48.50 | Imo | DavidR2008: automon => 01 ??? |
| 17:48.51 | DavidR2008 | imo: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 17:49.07 | seb- | Maliuta: yikes...604 pages |
| 17:49.16 | Maliuta | seb-: alternately, several ppl in here might do it for you ... for a price |
| 17:49.27 | Maliuta | seb-: say me ;) |
| 17:49.38 | DavidR2008 | that should be an option, but 01 (in your case) would only work if also put the w or W option in the Dial command |
| 17:49.42 | Maliuta | RTFB |
| 17:50.14 | Imo | DavidR2008: thanks but i have automon and when i put 01 by the call it dosnt record the call |
| 17:50.27 | Maliuta | remind me to beat my ISP over the head |
| 17:52.05 | *** part/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 17:52.22 | DavidR2008 | it won't work on a call unless the call is originated by a dial command that includes the w or W commands. Quote from http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial |
| 17:52.23 | DavidR2008 | # w: Allow the called user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 17:52.24 | DavidR2008 | # W: Allow the calling user to start recording after pressing *1 or what defined in features.conf (Asterisk v1.2.x); requires Set(DYNAMIC_FEATURES=automon) |
| 17:52.26 | DavidR2008 | the DYNAMIC_FEATURES referenced here is not required on 1.4.23 and the *1 is the default, if you change it features.conf it changes here as well |
| 17:52.37 | [TK]D-Fender | seb-: * has a learning curve. Expect to play around with it for a few days |
| 17:53.48 | Maliuta | [TK]D-Fender: I thought that was a minor bump ... you mean to say _that_ was the curve? |
| 17:56.17 | Imo | DafvidR2008: dosnt work |
| 17:56.36 | Imo | i have the default config with *1 |
| 17:58.23 | Imo | it dosnt start the call |
| 17:59.28 | [TK]D-Fender | Imo: PASTEBIN your complete failed attempt |
| 18:01.10 | Imo | what i should pastbin ?? |
| 18:01.28 | Imo | i have only one exten for outgoing call |
| 18:01.43 | Imo | and i have automon => *1 |
| 18:01.46 | Imo | like default config |
| 18:01.57 | [TK]D-Fender | Imo: show us the CALL |
| 18:02.20 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 18:02.35 | *** join/#asterisk cesar_CR (n=cesar@201.199.62.194) |
| 18:02.38 | *** join/#asterisk af_ (n=getsmart@88-149-240-204.dynamic.ngi.it) |
| 18:02.47 | Imo | <PROTECTED> |
| 18:02.48 | Imo | <PROTECTED> |
| 18:02.48 | Imo | <PROTECTED> |
| 18:02.48 | Imo | <PROTECTED> |
| 18:02.48 | Imo | <PROTECTED> |
| 18:03.24 | DavidR2008 | Imo: you're missing the wW option on your Dial command |
| 18:03.41 | Imo | what is the wW option ? |
| 18:04.03 | DavidR2008 | the *four* lines I posted above |
| 18:04.19 | *** join/#asterisk jeff (i=jeff@unaffiliated/jeff) |
| 18:04.42 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.74) |
| 18:04.43 | [TK]D-Fender | Imo: Options you were TOLD you had to sue to ENABLE recording |
| 18:04.46 | [TK]D-Fender | use* |
| 18:04.54 | [TK]D-Fender | Imo: "core show application dial" |
| 18:04.58 | CapriCoRN^80 | hi ! i need some help on asterisk startup script |
| 18:05.01 | CapriCoRN^80 | i am on centos 5 |
| 18:05.08 | [TK]D-Fender | CapriCoRN^80: "make config" |
| 18:05.17 | [TK]D-Fender | CapriCoRN^80: "chkconfig asterisk on" |
| 18:05.21 | [TK]D-Fender | CapriCoRN^80: THE END |
| 18:05.31 | CapriCoRN^80 | [TK]D-Fender: hmm .. i did that |
| 18:06.16 | CapriCoRN^80 | [TK]D-Fender: i did make config when i built it from source |
| 18:06.20 | [TK]D-Fender | CapriCoRN^80: that will set it to load on startup. |
| 18:06.42 | [TK]D-Fender | CapriCoRN^80: "service asterisk start" to start it yourself |
| 18:06.54 | Imo | [TK]D-Fender: sorry i dont unterstand that where must i type the wW ?? |
| 18:07.23 | [TK]D-Fender | Imo: where you have the "tT". read the INSTRUCTIONS. |
| 18:07.23 | CapriCoRN^80 | now when i did chkconfig ... it gave me error . no such file or directory |
| 18:07.39 | [TK]D-Fender | CapriCoRN^80: PASTEBIN. |
| 18:07.51 | CapriCoRN^80 | service asterisk start -> asterisk"unrecognized service |
| 18:08.09 | ecret | can someone please try calling my asterisk with a sip phone? address is 99.246.65.73 . I need to see if its an external issue or related to my DID not working correctly. |
| 18:08.34 | [TK]D-Fender | caprgo do "make config" again |
| 18:08.59 | [TK]D-Fender | ecret: And what are we supposed to dial at your server? |
| 18:09.12 | outtolunc | 911 |
| 18:09.16 | ecret | [TK]D-Fender 1000 i guess |
| 18:09.18 | [TK]D-Fender | ecret: And you haven't described your failure at all either |
| 18:09.38 | [TK]D-Fender | ecret: you GUESS? Go look at your config. |
| 18:09.38 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.74) |
| 18:09.46 | CapriCoRN^80 | sorry i got disconnected |
| 18:09.56 | [TK]D-Fender | CapriCoRN^80: go do "make config" again |
| 18:10.02 | rue_mohr | you dont have to be sorry, we just have to get even |
| 18:10.40 | CapriCoRN^80 | [TK]D-Fender: if i do it again it will not disturb my configuration ? |
| 18:10.51 | [TK]D-Fender | CapriCoRN^80: No |
| 18:11.20 | ecret | [TK]D-Fender: any extension is fine. The issue is I cant properly bridge incoming SIP's from my DID to my MRCP server. It complains about no RPT stream. I was hoping to see if it was any external(non lan) connection that did not work. |
| 18:11.47 | [TK]D-Fender | ecret: then debug an actual call |
| 18:12.49 | rue_mohr | yea the bookneeds an update, their using 7.4.16 and posgres is now at 8.3.6 |
| 18:13.17 | *** join/#asterisk rift0r (i=rift@420nugs.info) |
| 18:14.18 | rift0r | Any of you guys run vodavi phones with asterisk? |
| 18:14.26 | rift0r | i found some for cheap |
| 18:14.30 | rift0r | was curious if anyone had any input or thoughts |
| 18:14.46 | CapriCoRN^80 | [TK]D-Fender: thanks |
| 18:15.05 | [TK]D-Fender | rift0r: link & price please |
| 18:15.49 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 18:17.49 | rue_mohr | anyone know if the books package list "unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel" translates to unixodbc unixodbc-dev libltdl3 libltdl3-dev ? |
| 18:17.52 | carrar | Is there a wav file of the ring sound that is generated over the phone in north america that anyone has handly? |
| 18:18.01 | Qwell | rue_mohr: on? |
| 18:18.08 | rue_mohr | for debian |
| 18:18.16 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 18:18.23 | rue_mohr | I cant find packages called libtool-ltdl |
| 18:18.56 | [TK]D-Fender | carrar: Go make your own. |
| 18:19.22 | rue_mohr | he might have to travel half way aorund the world to do that you know |
| 18:19.42 | carrar | yeah |
| 18:19.54 | carrar | was trying to eliminate that |
| 18:19.56 | carrar | heh |
| 18:20.06 | rue_mohr | I dont have it recorded, but its like drdrdrdrdrdrdr drdrdrdrdrdrrdrd |
| 18:20.08 | rue_mohr | :) |
| 18:20.31 | [TK]D-Fender | carrar: jsut make your own... you could have been finished already by now... |
| 18:20.46 | Maliuta | reminds me to play with ring cadence again |
| 18:20.53 | rue_mohr | he dosn't know how to do it without going to na |
| 18:20.55 | Maliuta | it screws with people |
| 18:21.37 | ZeNN | quick question, the asterisk-gui complain about that it cannot find any spans (No mISDN Cards detected (found 0 spans)) ... but in the cli: misdn show stacks (Port 1 Type TE Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0) ... how does the GUI probe the mISDN card, through misdn-init ? |
| 18:22.08 | [TK]D-Fender | rue_mohr: * can do this all for you... |
| 18:22.21 | rue_mohr | tell him not me |
| 18:22.28 | [TK]D-Fender | rue_mohr: YOU tell him :p |
| 18:22.52 | rue_mohr | I'm knee deep in ... |
| 18:23.04 | [TK]D-Fender | EW! |
| 18:23.05 | [TK]D-Fender | :p |
| 18:23.14 | [TK]D-Fender | carrar: You too! |
| 18:24.34 | Maliuta | ZeNN: Read the topic, and go somewhere that gives a smeg about the gui |
| 18:25.06 | ZeNN | Maliuta: yep, asterisk-gui... thanks |
| 18:25.32 | Maliuta | yeah, not here. Here vim is a gui |
| 18:26.17 | rue_mohr | uh ho, /usr/lib/libodbcpsql.so isn't in any of the debin packages I know of |
| 18:27.35 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 18:33.44 | Maliuta | rue_mohr: odbc-postgresql +ln |
| 18:36.12 | Maliuta | hmm, had calories. Should go to bed ... it being 25 to 5(am) |
| 18:37.23 | rue_mohr | yea, and much has changed, I'm reffering to the webpage for a bit now |
| 18:37.46 | rue_mohr | I'll hopelessly mix the two, get all frustrated and go all crankey |
| 18:37.59 | rue_mohr | but I only have a half hour left, so whatever |
| 18:40.46 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:40.53 | *** join/#asterisk criten (n=criten@216.75.233.220.exetel.com.au) |
| 18:41.11 | criten | err, anyone had experience connecting nokia handsets to *? |
| 18:42.52 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:44.32 | *** join/#asterisk johndbritton (n=johndbri@cpe-74-67-1-88.nycap.res.rr.com) |
| 18:45.09 | johndbritton | can someone recommend a good sip provider for using with asterisk, i need to have around 25 direct dial sip phones |
| 18:45.52 | criten | nuphone is pretty good |
| 18:46.14 | criten | sorry, nufone |
| 18:46.34 | rue_mohr | I feel I'm doing ok, I have the odbc working |
| 18:46.50 | johndbritton | that looks better than nuphone |
| 18:47.19 | *** part/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 18:48.21 | rue_mohr | eeek , I have to recompile asterisk after the odbc stuff... |
| 18:48.41 | *** join/#asterisk nandatebayo (n=nandateb@unaffiliated/killown) |
| 18:48.42 | *** join/#asterisk maagic (i=maagic@fsck.fi) |
| 18:50.26 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:51.08 | johndbritton | their phone doesnt work, just tried calling them |
| 18:51.15 | johndbritton | any other recommendations? |
| 18:51.22 | criten | that sounds about right |
| 18:51.29 | criten | can never reach phone companies on the phone... seems ironic |
| 18:58.04 | rue_mohr | they know their service costs too much |
| 18:58.50 | [TK]D-Fender | johndbritton: ... |
| 18:58.53 | [TK]D-Fender | ~itsplist-us |
| 18:58.54 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 18:59.16 | *** join/#asterisk eric_hill (i=eric_hil@204.94.175.11) |
| 18:59.17 | [TK]D-Fender | johndbritton: Vitelity / Bandwidth / Voicepulse. All pretty decent |
| 19:03.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:03.43 | *** join/#asterisk Swistaczek (n=swistak@dbb200.neoplus.adsl.tpnet.pl) |
| 19:03.46 | Swistaczek | Hello |
| 19:04.56 | Swistaczek | I am bigginer that want to start with asterisk. My main target is to configure asterisk to work with VoIP line. What should I read first? |
| 19:05.15 | rue_mohr | yay I have odbc support in asterisk |
| 19:05.26 | [TK]D-Fender | Swistaczek: ... |
| 19:05.28 | [TK]D-Fender | ~book |
| 19:05.28 | jbot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:06.02 | [TK]D-Fender | Swistaczek: And for some inspiration : jeff_phillips: heh,chump :P .... MP3 is a MIXED-Stereo format where it averages sound into a common pseudo-center-channel. |
| 19:06.04 | [TK]D-Fender | oops |
| 19:06.07 | [TK]D-Fender | lol |
| 19:06.10 | [TK]D-Fender | ~jerjerguide |
| 19:06.11 | jbot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 19:06.15 | [TK]D-Fender | Swistaczek: that :) |
| 19:06.36 | Swistaczek | [TK]D-Fender: thanks you very much |
| 19:07.35 | Swistaczek | [TK]D-Fender: this that cover asterisk 1,4? |
| 19:07.40 | *** join/#asterisk voxter (n=voxter@190.241.15.218) |
| 19:07.45 | [TK]D-Fender | Swistaczek: Yes |
| 19:11.31 | johndbritton | Thanks [TK]D-Fender |
| 19:15.22 | johndbritton | i have a question, how would you go about this... I've got a shared office with 25 phones, each of the 25 phones needs to have its own direct dial number. The phones are in groups of 5, each group corresponds to one company in the shared office. How can I minimize the service cost for everything |
| 19:16.14 | [TK]D-Fender | johndbritton: Depends on the # of minutes required / month, simultaneous channels, etc |
| 19:16.58 | johndbritton | I'm still trying to figure out exactly what I'm looking for, i think it's channels |
| 19:17.08 | johndbritton | the number of calls that can be made at one time |
| 19:23.36 | *** join/#asterisk Aurs (n=Ove_Aurs@85.252.101.201) |
| 19:25.33 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:28.27 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 19:31.23 | eric_hill | I have an Asterisk 1.4 box that has started locking up at random times throughout the day. How do you troubleshoot a hard-locked linux box? |
| 19:32.10 | tzafrir_laptop | eric_hill, hard-locked linux box? |
| 19:32.16 | tzafrir_laptop | is asterisk running with -p ? |
| 19:32.42 | tzafrir_laptop | also: what kernel version? |
| 19:33.32 | eric_hill | Yes, non-responsive on the console, 2.6.24-16, and asterisk is running with -p. I think it's related to the PCI T1 card, but I don't know how to be sure. |
| 19:34.34 | eric_hill | Digium Wildcard TE110P board. Maybe it's too old? Do those go bad? |
| 19:35.24 | eric_hill | The system runs memtest86 for hours without issue, and the drives are new without any sector errors. |
| 19:36.29 | tzafrir_laptop | do you see anything on the console? any panic message or whatever? |
| 19:36.42 | tzafrir_laptop | does the machine respond to pings? |
| 19:37.05 | tzafrir_laptop | do alt-sysrq sequences work on the console? |
| 19:37.39 | eric_hill | No, console is blank (screen saver) and no panics. I haven't tried the alt-sysrq, what does that do? |
| 19:37.52 | eric_hill | Machine DOES respond to pings. |
| 19:38.08 | tzafrir_laptop | you won't see a panic is the console is blank |
| 19:38.55 | tzafrir_laptop | ssh or anything else responds? |
| 19:39.14 | tzafrir_laptop | is asterisk still responsive? |
| 19:39.18 | eric_hill | No, all IP related services are toast. System has SSH and Apache for monitoring some log files. |
| 19:39.24 | eric_hill | Asterisk is non-responsive. |
| 19:39.28 | tzafrir_laptop | e.g. port 5038 (if it is exposed) |
| 19:40.36 | criten | eric_hill: if the machine locks up with asterisk not running... |
| 19:40.46 | criten | then its a hardware/software issue |
| 19:41.23 | eric_hill | Asterisk seems to be part of the issue. System runs fine without it. I still think it's related to the Asterisk/Digium T1 board. |
| 19:41.36 | criten | ahh |
| 19:41.45 | criten | should contact digium, I recon :) |
| 19:41.48 | eric_hill | But all the timing tests work fine, and the system runs great for days on end. |
| 19:42.00 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
| 19:42.07 | criten | crontab a reboot? lulz |
| 19:42.19 | eric_hill | Beware all ye who enter here, for there be dragons, and ye taste like chicken. |
| 19:42.39 | eric_hill | criten: Actually am considering a nightly just for that reason... :) |
| 19:43.08 | criten | contacting digium would be yer best bet... you should get support with that card |
| 19:44.05 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 19:44.36 | eric_hill | Will do. Thanks for verifying my sanity. Well, at least humoring my sanity. |
| 19:45.10 | CapriCoRN^80 | i assigned sip id to hardphone and i can receive call from hard phone to softphone but when softphone try to call hard phone i get error |
| 19:45.41 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 19:47.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:49.38 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 19:50.28 | *** join/#asterisk stoffell (n=kristof@d51A4D629.access.telenet.be) |
| 19:50.49 | DavidR2008 | is there a way to escape a ; in the dialplan? I'm trying to run two linux commands back-to-back from MixMonitor and I need to use a ; to separate the commands |
| 19:56.38 | kaldemar | try \; |
| 19:58.31 | CapriCoRN^80 | kaldemar: i have setup my CISCO ATA 188 hard phone and and my hard phone can call softphone . but when user from softphone try to call hard phone. i get error |
| 19:58.43 | DavidR2008 | kaldemar: thx! that did the trick |
| 19:59.10 | kaldemar | CapriCoRN^80: could you consider telling what the error is? |
| 19:59.18 | CapriCoRN^80 | Unable to create channel of type 'SIP' |
| 19:59.33 | *** join/#asterisk still-lotek (n=still-lo@rrcs-97-77-195-186.sw.biz.rr.com) |
| 19:59.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:00.21 | kaldemar | CapriCoRN^80: what does "sip show peers" say? |
| 20:01.38 | CapriCoRN^80 | nothing .. i mean no ip or username coming in the output of sip show peers |
| 20:02.42 | kaldemar | but it doesn't say "No such command"? |
| 20:03.01 | CapriCoRN^80 | kaldemar: i just did reset the hard phone |
| 20:03.16 | still-lotek | Having an issue with music on hold. I receive the following: |
| 20:03.17 | CapriCoRN^80 | and its working |
| 20:03.17 | still-lotek | Music class default requested but no musiconhold loaded. |
| 20:03.26 | still-lotek | thats in my asterisk log. |
| 20:04.18 | still-lotek | This is my musiconhold.conf |
| 20:04.20 | still-lotek | [default] |
| 20:04.20 | still-lotek | mode=mp3 |
| 20:04.20 | still-lotek | directory=/var/lib/asterisk/mohmp3/main/ |
| 20:04.26 | still-lotek | Any suggestions? |
| 20:04.55 | *** join/#asterisk Eduardo_Assis (n=Eduardo_@200-207-61-133.dsl.telesp.net.br) |
| 20:05.13 | *** join/#asterisk Franz81 (n=franz81@200-207-61-133.dsl.telesp.net.br) |
| 20:07.48 | [TK]D-Fender | still-lotek: Pick a valid mode |
| 20:08.00 | still-lotek | mode=files |
| 20:08.01 | still-lotek | ? |
| 20:11.27 | [TK]D-Fender | still-lotek: better, and if you have MP3's in there be sure to have installed asterisk-addons |
| 20:11.49 | still-lotek | Ok. |
| 20:11.54 | still-lotek | Lets give this a shot. |
| 20:11.59 | still-lotek | Thanks. |
| 20:13.03 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 20:16.52 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:16.53 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:20.35 | *** join/#asterisk jmworx (n=jeval@mail.octasic.com) |
| 20:21.47 | jmworx | Anyone can tell me where I can get a good set of prompts. I don't really care about the contents or language, I'm just looking for good-quality sample material |
| 20:23.25 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 20:23.40 | eric_hill | jmworx: The stuff that you get with asterisk isn't enough? |
| 20:23.44 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 20:25.50 | jmworx | eric_hill: I didn't get Asterisk in the first place, but in general any data would be good. |
| 20:25.51 | [TK]D-Fender | jmworx: Whats the point when the contents have no relevance to you? |
| 20:25.59 | jmworx | This is not related to Asterisk, but to Speex |
| 20:26.21 | jmworx | [TK]D-Fender: The idea is to have test material for Speex (I'm the author) |
| 20:26.26 | [TK]D-Fender | jmworx: Grab a mic and record yourself |
| 20:26.49 | [TK]D-Fender | jmworx: Words from the author's mouth! |
| 20:26.57 | jmworx | [TK]D-Fender: No, that really sucks. I don't have the kind of voice to do clean prompts and I don't have a studio either |
| 20:27.08 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:27.28 | [TK]D-Fender | jmworx: jsut grab anything and transcode it. Go find a "book on tape" or something |
| 20:27.35 | jmworx | Not to mention that I can barely speak with the cold I have |
| 20:28.05 | jmworx | [TK]D-Fender: I need actual (realistic) prompts, not any speech. I have tons of "normal" test sentences |
| 20:29.16 | [TK]D-Fender | jmworx: then call up the telco and record their IVR |
| 20:29.29 | jmworx | Yeah, right |
| 20:30.03 | jmworx | (even if it was convenient to do that, it would still suck because they would have at least been processed with G.711) |
| 20:30.31 | jmworx | So Asterisk doesn't come with any prompts or what? |
| 20:31.35 | dni | can anyone suggest a good provider to terminate IAX/SIP that offers unlimited in/out local/domestic ?? I tried TelaSIP but the phone support sucked. |
| 20:32.21 | [TK]D-Fender | jmworx: Nope. * for YOU to build, not "hi we're a cookie factory, here's our complete IVR just as a sample for that Speex guy" :) |
| 20:32.37 | johndbritton | [TK]D-Fender: do you have a voicepulse referral code? |
| 20:32.37 | [TK]D-Fender | checkout time, BBL |
| 20:32.57 | johndbritton | or anyone else |
| 20:33.07 | kaldemar | jmworx: http://downloads.digium.com/pub/telephony/sounds/ |
| 20:34.13 | kaldemar | those may not match your quality requirements but have a look. |
| 20:34.38 | *** join/#asterisk voxter (n=voxter@190.241.15.218) |
| 20:35.07 | still-lotek | dni I use simplesignal. |
| 20:35.10 | still-lotek | Great company. |
| 20:35.21 | dni | thank you,. i will check it now |
| 20:35.32 | still-lotek | Tell them Digital Street referred you. |
| 20:35.35 | still-lotek | :) |
| 20:35.49 | jmworx | kaldemar: Looks like there's some uncompressed ones. I'll have a closer look. Thanks |
| 20:41.02 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 20:41.19 | dni | still-lotek, do t hey provide unlimited local/domestic ? |
| 20:42.59 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
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| 20:46.32 | *** part/#asterisk guizazeri (n=guilherm@200-207-61-133.dsl.telesp.net.br) |
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| 21:00.52 | *** part/#asterisk lbt (n=david@78.32.229.233) |
| 21:03.01 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 21:07.31 | *** join/#asterisk ScribbleJ (n=nsj@c-67-172-6-141.hsd1.il.comcast.net) |
| 21:10.04 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) |
| 21:15.03 | ScribbleJ | OKay, this is kinda vague, I know, but my coworker admins one of our * servers, 1.4, and says today it stopped answeringf calls, and there were a large number of unreaped zombie processes. Google didn't give me anything obvious, is this somehting that rings any obviousness bells here? |
| 21:16.15 | eric_hill | ScribbleJ, what linux version? |
| 21:16.49 | eric_hill | (Cause I'm fighting a kernel-level crash/halt/failure/lockup/whatever on a 2.6.something kernel) |
| 21:18.42 | *** part/#asterisk lbt (n=david@78.32.229.233) |
| 21:19.31 | ScribbleJ | eric_hill - it's 2.6 something I'm sure, some recent Ubuntu version. |
| 21:24.22 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:26.11 | *** join/#asterisk killown (n=idai@unaffiliated/killown) |
| 21:26.22 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 21:29.44 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:30.15 | *** join/#asterisk gmfm (n=hithere@boise-office.itsatomic.net) |
| 21:30.35 | *** join/#asterisk dlewis (i=c7340d66@about/security/staff/dlewis) |
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| 21:40.09 | *** join/#asterisk still-lotek (n=still-lo@rrcs-97-77-195-186.sw.biz.rr.com) |
| 21:40.15 | still-lotek | Hey guys. |
| 21:40.17 | still-lotek | Quick questions. |
| 21:40.23 | russellb | quick answers! |
| 21:40.27 | still-lotek | yes! |
| 21:40.28 | still-lotek | When someone uses the phonebook directory, how can I allow it so if they press 0 they go back to the IVR or back to a receptionist extension? |
| 21:40.50 | still-lotek | So in case they cant find the entry.. they can always speak to someone if they hit '0' |
| 21:41.03 | still-lotek | Would seem like a built in option. |
| 21:42.34 | gmfm | i think if you create an extension for o (not 0, o) in an accessible context you will succeed |
| 21:42.35 | carrar | read the voicemail.conf file? |
| 21:43.08 | gmfm | i base this on core show application Directory |
| 21:43.23 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 21:43.36 | still-lotek | An extension for 'o' |
| 21:43.39 | carrar | operator=yes |
| 21:43.39 | still-lotek | ok.. hmm |
| 21:44.38 | still-lotek | voicemail.conf:operator = yes |
| 21:44.48 | carrar | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail |
| 21:44.58 | still-lotek | How does voicemail.conf play into pbdirectory? |
| 21:45.02 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 21:45.28 | *** join/#asterisk docidu (n=eris@69.196.68.142) |
| 21:45.34 | aiksa[LV] | hmm, very strange indeed |
| 21:45.49 | aiksa[LV] | i am looking at manager events for incomming calls |
| 21:46.04 | docidu | does mgcp trunking work yet? i want to connect to an audio codecs trunkpack img box, that doable? |
| 21:46.08 | eric_hill | ScribbleJ, I'm on the same OS (Ubuntu)... I wonder if we're hitting a bug... |
| 21:46.22 | aiksa[LV] | the dialplan for all incomming calls is the same |
| 21:46.24 | marv[work] | Any particular reason why, after a channel masquerade, the channel keep's the uniqueid of it's new shell instead of it's original uniqueid? |
| 21:46.28 | aiksa[LV] | smae caller, same target |
| 21:46.41 | aiksa[LV] | no fancy agi dynamic routing stuff |
| 21:46.59 | aiksa[LV] | but the amount of newExten events differ with every call |
| 21:47.33 | carrar | still-lotek, pbdirectory is TrixBox |
| 21:47.41 | carrar | jon that channel and ask |
| 21:47.43 | carrar | join |
| 21:47.56 | aiksa[LV] | in some cases they start at the very first ext,priority (as expected) sometimes only when this call is sent to another context through goto . any ideas? |
| 21:48.08 | still-lotek | carrar, thanks. |
| 21:48.10 | still-lotek | Im new to this. |
| 21:48.22 | still-lotek | Im actually running freepbx.. I assume its the free version of trixbox? |
| 21:48.30 | carrar | same thing, join #freepbx then |
| 21:50.06 | aiksa[LV] | scratches his head, google isnt helping either. |
| 21:52.34 | ScribbleJ | eric_hill, it's hard to say, I'm really a second-party to the problem we're having, but I'm asking here because I have probably more familiarity with Asterisk overall. |
| 21:52.42 | aiksa[LV] | perhaps some bug in the mediating library |
| 21:53.26 | aiksa[LV] | but then again - why this happens on random cases? |
| 21:53.39 | aiksa[LV] | anyone else seen smth. like this before? |
| 21:56.49 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 21:57.03 | *** join/#asterisk _Sam-- (n=sam@unaffiliated/sam--/x-573746) |
| 21:57.12 | gmfm | anyone know if there are any channel banks that support sending caller id with name out the FXS interfaces? I assume they would have to connect to * with a PRI instead of just T1? |
| 21:57.40 | _Sam-- | hey we are using some php click to dial scripting here....and using the API action originate.....can the channel contain an & like the dial command? |
| 21:58.15 | _Sam-- | so that the channel for the originate would be like SIP/1&SIP/2 ? |
| 21:58.51 | aiksa[LV] | _Sam--: not sure but you could always use Local/ as a proxy |
| 22:00.11 | eric_hill | ScribbleJ: What kind of hardware do you run? This is a dell 1U 2-CPU Dual core box. |
| 22:03.23 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 22:06.15 | CapriCoRN^80 | i need some help with asterisk to MV-370 Mobile VOIP |
| 22:06.46 | aiksa[LV] | MV - the mobile voip gateway? |
| 22:06.49 | CapriCoRN^80 | i have configured it using doc . http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk |
| 22:06.54 | aiksa[LV] | sip to gsm? |
| 22:06.58 | CapriCoRN^80 | aiksa[LV]: yes |
| 22:07.03 | CapriCoRN^80 | aiksa[LV]: yes |
| 22:07.13 | aiksa[LV] | oh boy , i have had fun with that sutff :)) |
| 22:07.19 | CapriCoRN^80 | aiksa[LV]: really |
| 22:07.31 | aiksa[LV] | was that 1 SIM or 2 SIM version? |
| 22:07.38 | CapriCoRN^80 | aiksa[LV]: 1 sim |
| 22:07.40 | *** part/#asterisk kb3ien (n=kb3ien@isl177-max1.accesshighway.net) |
| 22:07.58 | CapriCoRN^80 | the problem is that i configured it but i cant able to call from mobile to my MV-370 sim |
| 22:08.18 | aiksa[LV] | ok. IMHO they have the most terrible GUI for configuring that thing. |
| 22:08.27 | CapriCoRN^80 | but i can msg from MV-370 to other mobile |
| 22:08.34 | aiksa[LV] | but nevermind. |
| 22:08.58 | CapriCoRN^80 | aiksa[LV]: http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk |
| 22:09.00 | CapriCoRN^80 | i used this site |
| 22:09.59 | *** join/#asterisk voxter (n=voxter@190.241.15.218) |
| 22:10.20 | *** join/#asterisk ghento (n=ghento@d75-157-199-106.bchsia.telus.net) |
| 22:10.57 | elitecoder | Is there a log file that shows attempted connections? |
| 22:11.17 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 22:11.41 | CapriCoRN^80 | aiksa[LV]: you there ? |
| 22:11.45 | aiksa[LV] | yes |
| 22:11.59 | aiksa[LV] | trying to access one of those bastards |
| 22:12.11 | elitecoder | I'm guessing it's the event log... nothing in there so nm |
| 22:12.16 | *** join/#asterisk gultig (n=dbriggs@70.96.32.114) |
| 22:12.26 | CapriCoRN^80 | aiksa[LV]: did you check that site ? |
| 22:12.27 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 22:12.34 | _Sam-- | i see<_Sam--> so that the channel for the originate would be like SIP/1&SIP/2 ? |
| 22:12.34 | _Sam-- | -:- SignOff iEatChildren: #asterisk (Remote closed the connection) |
| 22:12.34 | _Sam-- | <aiksa[LV]> _Sam--: not sure but you could always use Local/ as a proxy |
| 22:12.38 | gultig | Hello |
| 22:12.40 | _Sam-- | er fme |
| 22:12.42 | _Sam-- | sorry. |
| 22:13.37 | gultig | Anyone up for giving me some advice on a T1 purchase? |
| 22:13.53 | aiksa[LV] | CapriCoRN^80: WHAT WAS THE DEFUAL USR, PWD FOR THAT THING? |
| 22:14.02 | aiksa[LV] | sorry for capslock |
| 22:14.37 | CapriCoRN^80 | aiksa[LV]: which thing ? |
| 22:14.52 | aiksa[LV] | the mobile gateway. |
| 22:15.13 | aiksa[LV] | i dont have documentation at hand. |
| 22:15.16 | CapriCoRN^80 | aiksa[LV]: voip and 1234 |
| 22:17.39 | docidu | T1 purchase? well, after haveing delt with an ancient switch, id say stay away from e&m wink, its kinda flakey in my experence...other than that dunno what to tell ya |
| 22:17.41 | gultig | Does anyone here know if echo cancellation would be of any use in a T1 pass-through setup (CO->asterisk->Nortel)? Since it's an all digital connection it seems like a waste to me, but I don't want to end up having to turn around and buy another card if it's not. |
| 22:18.12 | gultig | docidu: I meant T1 card. Sorry. |
| 22:18.12 | docidu | hrmmm, ya could always get one that has a modular hwec board |
| 22:18.56 | gultig | That's probably what I'll end up doing. |
| 22:19.18 | docidu | i rather like my sangoma cards, great tech support, can be a little complicated to set up, dunno if its the same with digum stuff or not....dont really have a refrence point for other hardware |
| 22:19.37 | docidu | ahhh, heh, getting asterisk to passthrough is a fun lil challange |
| 22:20.36 | docidu | i just got done doing the same, replaceing a meridian with asterisk box, and it has to pass through to a cell switch, takes a lots of hacking around, at least it did for me |
| 22:20.54 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 22:21.02 | gultig | docidu: This would be my first experience with asterisk so I don't know anything about digium or sangoma. Just seems that sangoma is more expensive so far. |
| 22:21.33 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 22:21.33 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:21.33 | *** join/#asterisk dwayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 22:21.39 | *** join/#asterisk j0 (n=dan@S0106000c29242337.va.shawcable.net) |
| 22:21.59 | j0 | how can I tell asterisk to choose a specific pri or analog card for outbound calls? |
| 22:22.06 | docidu | really? most the sites i checked sangoma was a bit cheaper, mabey cause they canadian and im in canada |
| 22:22.32 | docidu | the group you set it as in your chan conf |
| 22:22.41 | nullable_type | Hey guys, when i am using DIAL function in my dial plan how do i tell it to dial a destination extension when the call is answered? |
| 22:23.02 | docidu | j0, can route them out paticular spans with the group number you set in the chan conf |
| 22:23.51 | docidu | oh, mabey i misread that |
| 22:24.05 | docidu | nvr mind me, my brain is offline |
| 22:24.46 | gultig | docidu: They're a canadian company I see, so maybe. |
| 22:25.06 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 22:26.42 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:26.44 | *** join/#asterisk Neph_fl (n=no@wsip-98-175-64-146.ga.at.cox.net) |
| 22:26.55 | Neph_fl | is there a usb version of the tdm400? |
| 22:27.12 | Neph_fl | an inexpensive 4 port channel bank |
| 22:27.43 | j0 | docidu: I have all my analog channels in there.. do i need to add all my PRI channels as well? |
| 22:27.45 | still-lotek | Why would anyone use a PRI instead of SIP trunks? |
| 22:29.15 | nullable_type | Hey guys, when i am using DIAL function in my dial plan how do i tell it to dial a destination extension when the call is answered? |
| 22:30.07 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:32.25 | johndbritton | how powerful of a machine would you recommend for up to 10 simultaneous calls |
| 22:35.01 | Qwell | johndbritton: quad-core Xeon with at least 16 GB of RAM. |
| 22:35.08 | elitecoder | hahahah |
| 22:35.10 | Qwell | or a pentium 2, 233mhz... both would work fine. |
| 22:35.18 | docidu | mine runs 15+ regularly and its a p4 1.7 ghz 1 gig |
| 22:35.43 | Qwell | I think Damin managed "a few" calls on a 133mhz CPU... |
| 22:36.26 | Qwell | johndbritton: real answer; it depends on what you're doing. |
| 22:37.29 | *** join/#asterisk rift0r (i=rift@420nugs.info) |
| 22:39.27 | elitecoder | I'm trying to get a client to connect to a box I have and it's timing out. It asks for a registry server - anyone got a link to some docs for setting that up |
| 22:39.30 | elitecoder | ? |
| 22:39.46 | elitecoder | I don't remember setting up a registry server sooo heh |
| 22:42.06 | *** join/#asterisk columbo (n=columbo@pool-96-229-199-29.lsanca.fios.verizon.net) |
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| 22:52.12 | *** join/#asterisk jeff_p (n=jeff_phi@209-206-132-50.dyn.centurytel.net) |
| 22:53.38 | jeff_p | who was that guy last night who had a cable modem / cable phone service? his nick name started with a B...? |
| 22:56.58 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 23:02.08 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:02.08 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:03.15 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 23:03.15 | *** mode/#asterisk [+o Mog] by ChanServ |
| 23:03.56 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 23:07.14 | docidu | umm, kinda off topic, but is there any way i can use a t1 card as a t1 line analyzer? |
| 23:07.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:11.48 | Corydon76-dig | docidu: that would require software for that function. Theoretically, yes. |
| 23:12.36 | Corydon76-dig | but not easily |
| 23:13.43 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:13.45 | nullable_type | guys i have a dialplan to bridge two calls using a gateway... sometimes i dont hear the ringing while its dialing to the second leg. Any idea? |
| 23:15.47 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 23:17.12 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
| 23:18.03 | nullable_type | its weird, only for some numbers the second leg doesnt look like ringing |
| 23:23.09 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 23:25.08 | nullable_type | Whats the internal on calling a number when Originate is used Vs DIAL is used |
| 23:25.24 | nullable_type | some numbers seems to work with Originate, but not with DIAL |
| 23:28.16 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 23:34.18 | *** join/#asterisk ManxPower (n=manxpowe@user-24-236-95-236.knology.net) |
| 23:35.32 | k-man | EmleyMoor: you awake? |
| 23:38.54 | elitecoder | Are there any default sound files that come with asterisk? |
| 23:40.57 | ManxPower | elitecoder: 1.2: yes. 1.4: No, but menuconfig lets you specify what sound packages you want and it will download them and install them for yo. |
| 23:42.06 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:42.07 | elitecoder | whoa |
| 23:42.14 | elitecoder | and that's inside the cli? |
| 23:42.33 | elitecoder | hmm or not |
| 23:43.14 | nullable_type | anyone have any idea why a number will work on ORIGINATE but not on DIAL? |
| 23:43.32 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 23:46.46 | nullable_type | exit |
| 23:54.23 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 23:54.34 | ruben23 | hi |
| 23:54.35 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
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