| 00:03.05 | *** join/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 00:04.04 | grandpapadot | Hey guys, in 1.4.27, anyone else suddenly experiencing temporary audio drop-outs? I upgraded 12 servers last week and only people using those 12 servers report audio drop outs. I can't seem to verify it as an issue so I was hoping to see if it was a known issue, the bug tracker didn't show anything. |
| 00:04.20 | grandpapadot | .. The audio does come back after 3-5 seconds .. |
| 00:05.04 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 00:07.10 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 00:09.14 | trnzmeta | anyone experienced 407 - proxy auth errors whereby a call is made, picked up but no audio on both ends |
| 00:10.28 | [TK]D-Fender | grandpapadot: Well 1.4.27.1 is out... perhaps thats why |
| 00:11.45 | ManxPower-work | trnzmeta: all authenticated SIP sessions start out unauth'd, get rejected, then retry with auth |
| 00:12.02 | ManxPower-work | grandpapadot: using IAX or SIP? |
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| 00:13.35 | grandpapadot | [TK]D-Fender: Yea, I thought of that, but that release is a security release that doesn't apply, although we will be rolling that patch out as well. |
| 00:13.44 | grandpapadot | ManxPower-work: SIP |
| 00:14.13 | grandpapadot | I can't catch the issue when it happens to start debugging, we have 80-100 channels active all the time on these boxes. |
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| 00:14.33 | trnzmeta | ManxPower-work: so you're saying that this 407 is normal error messages and I should be looking more at the RTP side of things? |
| 00:15.29 | ManxPower-work | trnzmeta: Correct. |
| 00:16.15 | grandpapadot | .. and of course I hate rolling back a version, lol |
| 00:17.54 | trnzmeta | ManxPower-work: one way audio probs usually indicate NAT/firewall on the client side |
| 00:18.22 | trnzmeta | two way would mean voip provider is firewall/nat issues on their end |
| 00:18.34 | citywok | i just had an issue with 20 seconds of silence on one leg and it was because NAT masquerading was only running on one side of our internal VPN for some reason. |
| 00:19.14 | bcrisp | and im back |
| 00:19.15 | trnzmeta | when I connect up using my asterisk box, my calls work fine, when I use my sip client to connect directly I get these rtp issues |
| 00:19.18 | grandpapadot | citywok: Thanks. This issue spans dozens of unknown NAT implementations (end-customers). |
| 00:20.04 | citywok | mine was consistently 22 seconds |
| 00:20.21 | citywok | but mine was purely an internal network (sort of, vpn WAN) |
| 00:20.33 | bcrisp | citywok: i saw the same thing when using Agents with nat in the mix |
| 00:20.43 | bcrisp | citywok: right at about 20-21 sec |
| 00:21.13 | citywok | yea, it was on one direction of the call and it was that nat masquerade rule |
| 00:21.24 | citywok | as soon as i caught that and removed it, the problem was fixed |
| 00:21.47 | citywok | i had issues with 30 minute call drops as well, every call died at 1768 to 1772 seconds, and upgrdaing from 1.6.1.8 to SVN fixed that |
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| 00:22.27 | grandpapadot | In asterisk 1.6, if no zaptel/dahdi hardware is used, are the zaptel/dahdi packages necessary? Is there a timing source (like ztdummy) in 1.6 proper? |
| 00:22.30 | bcrisp | can you use templates in queues.conf in 1.6.1.11? |
| 00:22.52 | bcrisp | (please say yes please say yes) |
| 00:23.11 | citywok | what do you mean by templates? |
| 00:23.22 | bcrisp | i.e. |
| 00:23.25 | citywok | i use PHP to generate my queues.conf for me, so it's kind of a template ;P |
| 00:23.27 | bcrisp | [queueTemplate](!) |
| 00:23.30 | Qwell | grandpapadot: yes, there is timing stuff in 1.6 now. however, dahdi_dummy does more than provide timing |
| 00:23.51 | bcrisp | [myQueue](queueTemplate) |
| 00:23.53 | Qwell | the most important thing it provides...is conference mixing |
| 00:24.21 | ManxPower-work | trnzmeta: If Asterisk is behind nat and you get no audio, maybe a localnet/externip issue |
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| 00:25.19 | sebbl | Seeking a possibility I am going with a welcome message and then continues with my random moh class |
| 00:25.22 | citywok | bcrisp: ahhh, i see that's pretty cool -- didn't know you could do that with *. not particularly useful for me but i'll keep that in mind as a possible feature to use. |
| 00:25.47 | grandpapadot | I haven't google this, maybe I should before asking, but is there a clean upgrade path from 1.4/zaptel (latest) to 1.6/dahadi (current)? |
| 00:26.02 | bcrisp | citywok: i was asking if it supports it with queues... i dont know :) |
| 00:26.20 | bcrisp | citywok: i have a bunch of queues to set up now and i dont want to keep repeating the common settings for each one |
| 00:26.22 | citywok | yea i knew what you were asking heh, but it made me google "asterisk templates" and see how they can be used for other things |
| 00:26.39 | citywok | yea, i can totally see how it would be useful, i've got about 100 queues but they are all custom configureable |
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| 00:27.18 | citywok | project managers can define how they want their queue to act individually, so its easier to just manually create each queue (150 lines of PHP takes care of pretty much everything i need) |
| 00:27.27 | bcrisp | citywok: ya.. with the complexities that are being requested im guessing ill need to use codegen for the confs (or maybe configure via mysql) |
| 00:28.29 | bcrisp | citywok, do you just fire off AGI from the dialplan to gen your queues? |
| 00:28.33 | grandpapadot | Lastly, I'm bringing nv_detect forward and it works, but I can't figure out how to re-implement USE_COUNT .. any pointers? |
| 00:29.15 | citywok | bcrisp: we built our interfaces around our in house applications so that the phone system can be mangaed by anybody in a user friendly way, so that applicatin triggers the PHP script which generates all the configs as needed (whenever something relevant gets changed) |
| 00:29.34 | bcrisp | ok, so you wrote a gui that interacts with your * sys? |
| 00:29.44 | bcrisp | cool |
| 00:29.50 | LemensTS | can you not issue cli cmds after you did a 'restart when convenient' ? |
| 00:29.51 | citywok | the gui is all stored in BMC Remedy / MSSQL |
| 00:30.09 | citywok | PHP queries the SQL server and generates the config files (sip.conf, extensions.conf, queues.conf, voicemail.conf) |
| 00:30.25 | bcrisp | citywok: i see |
| 00:30.27 | citywok | voicemail.conf i run out of * realtime, but sip, extensions, and queues i do with config files |
| 00:30.32 | Benny_132 | anyone have any ideas why hudlite will connect and display the extensions but when a call comes in/out it doesn't display in hud |
| 00:31.03 | citywok | a few hundred lines of well commented code generate most of my phone system running config |
| 00:31.03 | bcrisp | citywok: ive written some realtime interfaces via AMI, but nothing yet for managing the behavior / creation of queues |
| 00:31.33 | citywok | bcrisp: the only non static thing i do with queues is use dynamic members |
| 00:31.43 | bcrisp | ic |
| 00:31.49 | citywok | when an agent logs in to the billing system, the billing system checks to see if they are mapped to a phone project, if so, it logs them in to the phone system automatically |
| 00:32.06 | bcrisp | <-- not using agents |
| 00:32.12 | citywok | it prevents accidents from happening, people always screwed up logging in somehow and 3000 calls magically get logged under the wrong project |
| 00:32.19 | bcrisp | heh |
| 00:32.27 | citywok | our goal was to eliminate user error + reduce setup time |
| 00:33.01 | citywok | instead of taking an hour or two to build a project by an IT person, anybody can set up their own in 5 minutes. assign your own toll free numbres, agents, custom routing options (if you need a tree for instance), or just add basic members |
| 00:33.28 | bcrisp | what is a "project" ? |
| 00:33.32 | citywok | and it isn't something new we have to teach, because it's integrated in to the rest of the systems our company uses |
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| 00:33.44 | citywok | for us a project could be 1-20 agents working on a single calling campaign |
| 00:33.57 | bcrisp | hmm |
| 00:33.59 | citywok | could be calling outbund, answering on inbound, doing both, etc. for any one of our clients |
| 00:34.10 | bcrisp | i mean what kind of business are you in :) |
| 00:34.16 | citywok | call center :) |
| 00:34.35 | citywok | we make 3-5000 calls a day (not a big callcenter) |
| 00:35.18 | citywok | a couple hundred thousand minutes a month, nothign cool like the other guy with 20k simultaneous calls. we run up to 30 concurrent |
| 00:35.24 | bcrisp | ok so you host call centers for various clients then? |
| 00:35.35 | citywok | we work mostly with MSFT |
| 00:35.59 | citywok | helping re-enroll partners, awareness about products, business2business only stuff |
| 00:36.09 | bcrisp | im setting up an internal phone sys / call center for our software co |
| 00:36.12 | citywok | for the most part pre-established relationships |
| 00:36.44 | citywok | ya, we've got a system that works well for our needs for the most part, but we want to build our entire company around a datacenter, and our phone system we have now isn't flexible enough to do that |
| 00:36.54 | bcrisp | hides his company's gold partner logo |
| 00:36.56 | citywok | that was the key driving factor to going to an * solution |
| 00:37.09 | citywok | the bonus is the massive flexibility we get :) |
| 00:37.37 | citywok | bcrisp: http://www.csgopenline.com/ |
| 00:37.38 | bcrisp | ya its nice |
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| 00:38.01 | citywok | and the short rollout time. i proof of concepted the majority of our core features in a month |
| 00:38.18 | bcrisp | oh ok i see what you guys do now |
| 00:38.21 | citywok | i've spent a lot of time trackign down bugs in * that has delayed me, as have the developers integrating the UI's |
| 00:38.26 | bcrisp | your business is not "call center" lol |
| 00:39.00 | citywok | yea, we aren't bulk phoen calls do you like your long distance plan type |
| 00:39.34 | citywok | we like to think of ourselves as the premium call center. we're selling the PM's, reporting capabilities, data mangaement |
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| 00:40.15 | bcrisp | ic |
| 00:41.44 | manxpower | You help Microsoft? How do you live with yourself? |
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| 00:43.40 | citywok | hahaha |
| 00:43.47 | citywok | imagine if they new we made all their calls on * |
| 00:44.00 | manxpower | I suspect failures in MS Windows has killed more people than Charles Manson |
| 00:44.01 | citywok | you use OPEN SOURCE SOFTWARE!?!?!?!? where's UC?? |
| 00:44.22 | manxpower | Not to mention being a convicted monopolist. |
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| 00:45.06 | manxpower | citywok: I'm not totally against closed source software. |
| 00:45.20 | bcrisp | thats how i eat.... |
| 00:45.45 | manxpower | I use VMWare, TMPGEnc products. Even pay for them. |
| 00:46.14 | bcrisp | manxpower: VS products, SQL Server products here |
| 00:47.22 | bcrisp | .. here goes nothing |
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| 01:00.57 | bcrisp | sweet |
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| 01:10.29 | bcrisp | burr.. why does the office reach 80+ degrees in the day, then the AC kicks on at 5:30pm? |
| 01:10.41 | bcrisp | its ridunculous |
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| 01:25.12 | giantrobot | p3nguin: i figured out my RTP issue |
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| 01:27.10 | joobie | guys is there a function i can use to write an entry to the CDR log, triggered from the dialplan? |
| 01:27.36 | joobie | i found QueueLog() which writes to the queue log, cant seem to find one that writes to the CDR log.. basically wnat to trigger a log event to occur with specific variables, at a point in the dialplan |
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| 01:43.07 | tftech | Question - is it bad form in here to have asterisk hardware for sale? |
| 01:44.46 | bcrisp | joobie: CDR() |
| 01:46.38 | bcrisp | joobie: oh hmm.. no its readonly |
| 01:46.55 | bcrisp | you can use SetCDRUserField i think |
| 01:49.46 | bcrisp | ah hrm... asterisk was sending voicemail via email before.. but it appears to have stopped workin |
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| 02:13.04 | tftech | Have Digium Single T1/E1 card and Analog 1 FXS/ 1 FXO cards for sale if anyone is interested. Both brand new. |
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| 03:19.52 | b14ck | sup eveyrone |
| 03:20.00 | b14ck | (>'.')> |
| 03:20.04 | b14ck | <('.'<) |
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| 03:41.29 | steve745 | anyone have anyluck with AMD |
| 03:50.33 | Katty | HAI |
| 03:51.50 | TJNII | Oh hai thar |
| 03:51.52 | ChannelZ | steve745: well my * box is an AMD if that's what your asking |
| 03:54.31 | ChannelZ | This is like Where's Waldo: Can you find all 8? http://dlisted.com/node/35270 |
| 03:55.32 | TJNII | Where are the rest of her kids? |
| 03:55.42 | joobie | guys i want to structure the dialplan so that if a user dials '321', it goes to VoiceMailMain(<insert src internal extension number>@default) .. not sure what variable to use to get the src extension 321 is being dialed from.. |
| 03:55.47 | TJNII | Doesn't she have a herd numbering into the teens? |
| 03:55.58 | steve745 | Answering maching Detection |
| 03:56.03 | ChannelZ | she probably sold them |
| 03:57.11 | TJNII | exten => 321,1,VoiceMailMain(${CALLERID(num)}|s) |
| 03:57.12 | ChannelZ | joobie: VoiceMailMain(${CALLERID(num)}) - assuming your caller ID is set for the extensions |
| 03:57.59 | Katty | just bumped her head. oww. |
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| 04:00.27 | TJNII | So, I have to comment. I love my new job. Today I built a computer with 256Gb of Ram, 128 processors, and 96 PCI slots. Draws over 22kW. |
| 04:01.42 | TJNII | Actually, that memory amout is wrong. They had 4gb dimms, so that should be a terabyte of ram. |
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| 04:04.31 | maxagaz | how to grep informations in the asterisk console ? |
| 04:05.46 | ectospasm | maxagaz: grep blah /var/log/asterisk/full |
| 04:05.57 | ChannelZ | it must play a bitchen game of Solitare |
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| 04:06.16 | TJNII | Vista ready! |
| 04:06.18 | Katty | http://i.imgur.com/ycBXB.jpg <- wait, what? |
| 04:07.26 | ChannelZ | squirrels deserve dignity too |
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| 04:11.33 | maxagaz | ectospasm, thanks |
| 04:11.47 | maxagaz | what does asterisk reload do exactly ? |
| 04:12.50 | TJNII | http://3.bp.blogspot.com/_er7_VeH-o9s/Sa4lX5ev_6I/AAAAAAAACIE/HWUE9Zb9DdI/s400/squirrel_underwear2.jpg |
| 04:12.53 | maxagaz | i mean "/etc/init.d/asterisk reload" |
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| 04:17.19 | p3nguin | maxagaz: Read the script. |
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| 04:22.52 | TJNII | http://tinyurl.com/yds2abh |
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| 04:23.20 | TJNII | Okay, that was busted |
| 04:23.24 | TJNII | I think I'll stop. |
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| 04:37.13 | joobie | ChannelZ, CALLERID is set for the extension, but it's actually a name and a number |
| 04:37.24 | joobie | like in that "Firstname, Lastname <Exten>" format |
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| 04:57.55 | dlynes | As I understand it, Gosub() is almost like an include =>? i.e. after it tries an extension in the gosub'ed context, if it can't find the extension, it'll try looking in the rest of the context in which the gosub was called from? |
| 04:58.41 | dlynes | oh...nvm...it requires an explicit return |
| 04:58.45 | p3nguin | GoSub is the exact same as a Goto, but with the ability to return back to the context it was ran from. |
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| 04:59.58 | ChannelZ | joobie: yeah CALLERID(num) is just the number |
| 05:00.13 | dlynes | p3nguin: so is there something I can use that acts like an include =>, so that if it didn't find its target in where I told it to go to, it can come back to my calling context to look for a match/i extension? |
| 05:01.13 | p3nguin | That's automatic. |
| 05:01.50 | dlynes | No...gosub'd contexts seem to require an explicit call to the application, 'return' |
| 05:01.58 | p3nguin | right |
| 05:02.22 | p3nguin | What's wrong with regular old "include"? |
| 05:02.28 | dlynes | I want an option where I don't need an explicit return |
| 05:02.40 | p3nguin | Then "include" is what you need. |
| 05:03.16 | joobie | ChannelZ, only prob is that it's sometimes different from the internal extension, which is what the voicemail number is |
| 05:03.18 | dlynes | include is not the same thing for me...I need to be able to prescreen using a match against the caller id on every single match |
| 05:03.38 | joobie | sometimes the caller id number is set to the external number |
| 05:03.39 | dlynes | but I still need a generic version of the pattern as well, where I don't need to check the caller id |
| 05:04.09 | dlynes | Anyways...I guess you don't know, then |
| 05:04.12 | dlynes | Thanks |
| 05:06.33 | p3nguin | An actual number will match before a pattern, so you could easily create a Goto() at some point to send your callers to a different context, and then have exten numbers there, but also have a _X. in there with a Goto() to send the call back. |
| 05:06.38 | p3nguin | Let me write up what I mean. |
| 05:14.18 | p3nguin | dlynes: Something with a concept similar to this is what I had in mind: http://pastebin.com/d703b1171 |
| 05:17.18 | ChannelZ | joobie: Well if they are calling in from the outside they'll just have to type in their extension number at the voicemail prompt then |
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| 05:23.45 | voipmonk | does the cid change? |
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| 05:30.54 | joobie | ChannelZ, they are calling in from within the office |
| 05:31.35 | joobie | ChannelZ, i have all the office phones connected to their own sip account on the * box.. but against each sip account, i also define the callerid=var.. and set it to their external number.. so when they dial to people outside of the office, it displays their external number on the remote phones |
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| 05:40.40 | ChannelZ | Well an extension is just a made-up thing that exists in the dialplan, it really has nothing to do with the physical device. So you need to use something from the device and translate that into a mailbox number if you want it to be automatic |
| 05:46.59 | voipmonk | yep |
| 05:47.12 | voipmonk | so what do u want to do joobie? |
| 05:49.17 | joobie | voipmonk, want to be able to get the src extension |
| 05:52.04 | voipmonk | so when "they" dial to people outside the office, it displays their external number on the remote phones |
| 05:52.18 | voipmonk | you want to be able to get to who's source extension and do what with it |
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| 06:10.27 | joobie | voipmonk, |
| 06:11.24 | joobie | when a user in the office, say with extension 4000 dials '321' on their phone.. i want the dialplan to pick up the '4000' part so i can use it in the dialplan |
| 06:11.44 | joobie | the callerid can't be used for this because on some extensions, i define this to be something totally different from what their internal extension is |
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| 06:13.14 | voipmonk | ok |
| 06:13.43 | voipmonk | the user of extension 4000 ..... do they use a device registered as 4000 or have the device set in sip.conf with a callerid of 4000 ? |
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| 06:49.56 | kevm | can somebody help me configure dahdi?/ |
| 06:54.11 | ChannelZ | start with /etc/dahdi/system.conf |
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| 08:36.22 | sajam | Hello, i have asterisk installed and softphone connected to it. However, i don't TDM to connect to PSTN but I have cisco router 1760. Anyone has an idea about connecting Asterisk to Cisco V Router |
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| 09:20.34 | basecoat | Has anybody had any problems with the asterisk image on astersik.org? I get an error message, saying a file is missing or corrupted. |
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| 09:24.34 | Soeron | good day everyone |
| 09:25.05 | Soeron | can anyone give me a hand with a problem connecting 2 asterisk boxes with sip |
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| 09:26.05 | Soeron | i have setuped both boxes but when try to make a call from AST A to AST B i get chan_sip.c:18454 handle_request_invite: Failed to authenticate device "56" |
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| 09:58.03 | _cgc | hi everyone |
| 09:58.28 | _cgc | does any one know what this means? [Dec 17 09:56:05] WARNING[22957]: chan_sip.c:16925 handle_response_invite: Received response: "Forbidden" from '"XXXXXXXXXXX" <sip:XXXXXXXXXXX@192.168.3.10>;tag=as049571e8' |
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| 10:06.47 | _cgc | does anyone know how you can setup an outboundproxy for a single sip peer without affecting the other sip devices |
| 10:06.49 | _cgc | ? |
| 10:09.05 | angryuser | _cgc, another context for peer with its outbound params |
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| 10:09.53 | _cgc | angryuser> you can put the proxy details into a context in extensions.conf? |
| 10:09.53 | angryuser | or you can add a gotoif Peer = your peer > another proxy context |
| 10:10.18 | angryuser | _cgc, sure, dial out with another peer |
| 10:12.42 | _cgc | ok ill look into it thanks |
| 10:12.43 | _cgc | :) |
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| 10:15.04 | _cgc | hang on a minute... my problem it now dialing out though, its inbound calls only work when I have the sip proxy setup in the general part of sip.conf but this is a problem because it then dials a sip phone which tries to dial out through the sip proxy |
| 10:15.08 | _cgc | not* |
| 10:17.09 | _cgc | I can dial out through the sip trunk fine without a sip proxy, but i cannot receive calls through it, i think its a natting problem, and I have only been able to fix this by using a sip proxy |
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| 10:18.33 | _cgc | when i use the sip proxy, i see the call come in over the sip line but it then tries to call my sip phones but through the sip proxy which obviously deny's the calls |
| 10:20.11 | _cgc | is there a way around this problem? |
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| 11:04.12 | ElDios | hey guys..do you have any guide/howto/site/wiki on how to configure RTP/SAVP -- SIPS on asterisk? |
| 11:04.16 | ElDios | thank you |
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| 11:39.13 | _cgc | does anyone know how to configure asterisk 1.6.1.11 for stun? |
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| 12:10.55 | sajam | Hi guys, i have asterisk installed on my network. However, i don't have TDM card, i want to use Cisco 1760 to connect to PSTN. Anyone has an idea on how to do it? |
| 12:15.20 | waa | is there someone running fax for asterisk in asterisk svn trunk r235298? |
| 12:16.18 | waa | can I only run fax for asterisk in version listed here http://www.digium.com/en/docs/FAX/faa-download.php? |
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| 12:45.19 | johnyjj2 | hello |
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| 12:51.55 | TommyBotten | Hi |
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| 12:57.38 | johnyjj2 | I would be grateful if somebody can answer me three simple questions. I'd like to install Asterisk on the server with (unfortunately) Windows. I want the user to call the server, speak twelve digits, the server to recognize those digits and tell to the user if the control sum was calculated properly; then to repeat the same procedure many times until the user says 'finish'. Somebody suggested me LumenVox but it is non-free. I created my application f |
| 12:58.36 | TommyBotten | ... |
| 12:58.41 | Faustov | ... |
| 12:59.23 | TommyBotten | Voice recognition libraries for asterisk exists.. but ... speak the digits. Why not touch tone? |
| 12:59.27 | TommyBotten | And why on earth windows? |
| 12:59.55 | johnyjj2 | because i do it for somebody who has got windows and doesn't want to install linux (even if i would prefer linux) |
| 13:00.06 | johnyjj2 | and he wants both things, i.e. speech recognition and dtmf |
| 13:00.25 | johnyjj2 | for dtmf i thought about midlet which uses httpconnection (post method) and tomcat server |
| 13:00.45 | Gido-E | johnyjj2 i would congratulate him with his M$ wisses. Ans move on. |
| 13:00.55 | Gido-E | s/Ans/And/ |
| 13:01.15 | Gido-E | nice :-) |
| 13:01.27 | TommyBotten | johnyjj2: Gido-E is right. Please don't use windows for this. |
| 13:02.05 | TommyBotten | TommyBotten: And also, don't use an external tomcat for this. Just use an internal database, or even an external mysql |
| 13:02.08 | TommyBotten | eh |
| 13:02.24 | TommyBotten | s/TommyBotten:/ johnyjj2:/ |
| 13:02.46 | TommyBotten | johnyjj2: And then use the dialplan language ... or embedded languages such as lua |
| 13:02.51 | johnyjj2 | hm, so two additional questions: 1) if i decide to use windows, you say that it is not good combination win + asterisk + scribblej + sphinx4. So what would be good choice for windows, 2) is there any way to split it into two devices, e.g. windows + asterisk on one machine, sphinx4 + linux on the other one or vice versa ? |
| 13:03.00 | TommyBotten | johnyjj2: http post to a tomcat is ... not very optimal. |
| 13:03.17 | johnyjj2 | and about tomcat - i need to save results of recognition to database. This is why i want to send recognized digits from mobile phone to server. |
| 13:03.42 | TommyBotten | johnyjj2: Use windows for what windows is intended to do. Exhchange, outlook and excel. Don't run tomcat, asterisk on windows. |
| 13:04.06 | TommyBotten | johnyjj2: You can do that from the dialplan language. Or if you chose to use an embedded language like AEL or lua. |
| 13:04.18 | johnyjj2 | yeah, i remember how much effort i put to convince that person to let me install linux, not windows... unfortunately without success |
| 13:04.37 | johnyjj2 | but this server will be somewhere else so maybe i would convince him for linux |
| 13:04.54 | johnyjj2 | which distribution would you suggest for these applications? |
| 13:05.36 | johnyjj2 | and about those other two questions given at the beginning, is this device a good choice? and can i use scribblej plugin for sphinx4, not pocketsphinx? |
| 13:05.40 | TommyBotten | Centos/Redhat or ubuntu/debian. It really doesn't matter. It's mostly a matter of taste |
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| 13:05.52 | johnyjj2 | at home i use ubuntu :) |
| 13:06.05 | TommyBotten | I haven't used scribblej, so I cannot answer that |
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| 13:08.10 | johnyjj2 | so you say not to use httpconnection, post and tomcat but to use httpconnection, post and mssql? |
| 13:08.48 | johnyjj2 | or mysql for linux |
| 13:08.54 | TommyBotten | No... I'm saying implement the logic in the dialplan. And just use a mysql (or internal BDBif that covers it) for data |
| 13:13.05 | johnyjj2 | OK, so after all I should buy this device which I wrote about (vip_281gs planet -> http://www.planet.com.tw/en/product/product_ov.php?id=4160). Then install Linux and Asterisk. Create dialplan for DTMF part and - what to do on mobile phone? (do I still need any midlet? I guess not but I connecting to server for DTMF part would be more expensive than doing the whole stuff in midlet and then connecting to mobile internet for short time to send results) |
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| 13:13.55 | Kchehab | i have a problem loading chan_dahdi.so |
| 13:14.12 | Kchehab | any one can help me ,i post my script here-->http://pastebin.com/m2629a2c5 |
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| 13:15.21 | beek | Kchehab: Not much to go on here... is this a new installation or is this suddenly a problem? |
| 13:16.01 | TommyBotten | johnyjj2: Why not use DTMF for mobile phones as well? |
| 13:19.16 | *** part/#asterisk manxpower (n=ewieling@24.42.221.26) |
| 13:19.26 | Kchehab | beek installation |
| 13:19.41 | Kchehab | i can see in /var/log/messages |
| 13:19.42 | Kchehab | [Dec 17 05:21:24] WARNING[3686] chan_dahdi.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 13:20.03 | Kchehab | what is the meaning of that and where is the problem |
| 13:20.04 | *** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 13:21.19 | beek | Kchehab: Pastebin /etc/dahdi/system.conf, and /etc/asterisk/chan_dahdi.conf |
| 13:22.21 | Kchehab | i post my script here-->http://pastebin.com/m2629a2c5 |
| 13:22.25 | Kchehab | beek -- |
| 13:22.38 | TommyBotten | Kchehab: Your endpoint is usinb CPE. Is it your provider? |
| 13:23.03 | johnyjj2 | TommyBotten: I guess I'm not too familiar with DTMF technologies. May you suggest me any tutorials or articles which cover what would be best choice for this? I only have got some experience with this Sphinx4 and SphinxTrain for ASR part. (In fact it is my first real project in IT :p). So I'd like to say to the person who gave me this task to buy this device (if it is good choice; free rental for trying the device is also possible). Later I would con |
| 13:23.26 | Kchehab | i am testing a loop back from span 1 to span4 using cross E1 cable |
| 13:23.47 | beek | Kchehab: What is span 1 plugged into? |
| 13:23.50 | Kchehab | TommyBotten i configured span1 as pri_net and span 4 as pri_cpe |
| 13:24.10 | Kchehab | span 1 is connected to span 4 using cross E1 cable |
| 13:24.21 | sajam | Hi guys, i have asterisk installed on my network. However, i don't have TDM card, i want to use Cisco 1760 to connect to PSTN. Anyone has an idea on how to do it? |
| 13:24.37 | ManxPower-work | Kchehab: does ztcfg -v / dahdi_cfg -v generate any errors? |
| 13:25.00 | ManxPower-work | sajam: check the mailing lists |
| 13:25.02 | ManxPower-work | ~mailinglist |
| 13:25.03 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 13:25.55 | Kchehab | ManxPower-work no errors |
| 13:26.09 | ManxPower-work | Kchehab: pastebin the output |
| 13:27.18 | Kchehab | ManxPower-work http://pastebin.com/m6630d3c9 |
| 13:28.11 | ManxPower-work | Kchehab: try module unload chan_dahdi.so before the module load chan_dahdi.so |
| 13:29.18 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:29.25 | Kchehab | ManxPower-work same error chan_dahdi.c:1468 dahdi_open: Unable to specify channel 1: Device or resource busy |
| 13:30.18 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.19 | ManxPower-work | Kchehab: Could the user Asterisk is running as not have permisson to access /dev/dahdi? |
| 13:30.24 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:31.28 | Kchehab | ManxPower-work crw-rw---- 1 asterisk asterisk 196, 1 Dec 17 05:04 1 |
| 13:31.59 | Kchehab | ManxPower-work same for all |
| 13:32.21 | beek | morning jaytee |
| 13:32.46 | Kchehab | ManxPower-work any mistake in chan_dahdi or any config file |
| 13:33.41 | ManxPower-work | Kchehab: not that I see, but I'm only on my 2nd cup of coffee this morning |
| 13:33.42 | *** join/#asterisk Deeewayne (n=dwayne@71.228.179.90) |
| 13:33.42 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:34.17 | beek | hands ManxPower-work a freshly-brewed cup of Sumartra. |
| 13:34.37 | Kchehab | ManxPower-work hehehe,maybe i am disturbing you ,i dont want to upset you this morning by my problems |
| 13:35.05 | ManxPower-work | Kchehab: everything does look correct, that is an odd error message |
| 13:35.51 | beek | Is this a known-good card? |
| 13:36.47 | ManxPower-work | Well, nothing looks like it could cause that error. Things like callwaiting, threewaycalling, callreturn, etc make no sense whatsoever with PRI/QSIG, but they would not cause an error. |
| 13:37.32 | beek | Kchehab: Any error when you start dahdi? |
| 13:37.36 | Kchehab | ManxPower-work its a good cart |
| 13:37.41 | ManxPower-work | Kchehab: AHA! |
| 13:38.25 | Kchehab | ManxPower-work i dont thin so |
| 13:38.54 | ManxPower-work | Remember set the options, THEN specify the channels, for example channels 1-15,17-31 are in group 0,1 not group 63 and you have no context for those channels. |
| 13:39.11 | ManxPower-work | beek: dahdi_cfg would error out if DAHDI was not loaded |
| 13:39.26 | beek | ManxPower-work: good point. |
| 13:39.29 | ManxPower-work | sorry, you do have a context. |
| 13:39.55 | ManxPower-work | that looks like a GUI config of Asterisk |
| 13:40.43 | Kchehab | ManxPower-work i have a context |
| 13:40.44 | Kchehab | group=1 |
| 13:40.48 | Kchehab | context=from-pstn |
| 13:40.48 | Kchehab | switchtype = euroisdn |
| 13:40.48 | Kchehab | signalling = pri_cpe |
| 13:40.48 | Kchehab | channel => 1-15,17-31 |
| 13:41.03 | jaytee | morning beek |
| 13:42.42 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 13:43.51 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 13:44.49 | Kchehab | ManxPower-work pbx.c:4550 ast_register_application2: Already have an application 'DAHDISendKeypadFacility' |
| 13:44.54 | Kchehab | whats the meaning of that |
| 13:46.03 | Kchehab | <PROTECTED> |
| 13:49.05 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:49.06 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 13:50.19 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 13:56.26 | Pan3D | morning |
| 13:58.09 | *** join/#asterisk RypPn2 (i=TuMbL@rosscom.co.uk) |
| 13:58.31 | *** join/#asterisk bbt (n=sam@samuels.id.au) |
| 14:00.27 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
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| 14:12.19 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 14:13.08 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 14:13.58 | *** join/#asterisk areay (n=areay@188-220-19-194.zone11.bethere.co.uk) |
| 14:17.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:20.15 | Kchehab | ManxPower-work i fix the problem,another hitch appeared PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. |
| 14:23.00 | Katty | hello |
| 14:24.53 | *** join/#asterisk jkroon (n=jkroon@dsl-244-46-242.telkomadsl.co.za) |
| 14:29.21 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) [NETSPLIT VICTIM] |
| 14:30.08 | *** join/#asterisk Akiraa (n=Akiraa@79.112.37.124) |
| 14:31.33 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) [NETSPLIT VICTIM] |
| 14:31.33 | *** join/#asterisk pa (n=pa@unaffiliated/pa) [NETSPLIT VICTIM] |
| 14:35.41 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 14:37.28 | ManxPower-work | Kchehab: that means 1) loopback on line or 2) both ends if the link are set the same (CPE/CPE or NET/NET) |
| 14:38.45 | *** join/#asterisk coppice (n=chatzill@50.131.92.116.dyn.pacific.net.hk) |
| 14:39.23 | Katty | has sugar overload |
| 14:39.48 | *** join/#asterisk gaetronik (n=gaetan@LPuteaux-156-15-5-233.w80-14.abo.wanadoo.fr) |
| 14:40.20 | *** join/#asterisk Guest40094 (n=socain00@74.255.249.66) |
| 14:40.47 | tuxx- | has caffeine overload |
| 14:40.54 | tuxx- | although |
| 14:40.59 | tuxx- | you can never have too much caffeine |
| 14:40.59 | tuxx- | ;x |
| 14:41.39 | *** join/#asterisk ecto (n=weechat@user-24-236-95-118.knology.net) |
| 14:42.39 | Katty | tuxx-: actually you can. and the side effects are pretty severe. |
| 14:43.01 | Katty | tuxx-: perhaps the even worse part is how people obtain the caffeine. |
| 14:43.07 | tuxx- | coffee. |
| 14:43.08 | tuxx- | :D |
| 14:43.25 | *** join/#asterisk bbt (n=sam@samuels.id.au) [NETSPLIT VICTIM] |
| 14:43.25 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) [NETSPLIT VICTIM] |
| 14:43.25 | *** join/#asterisk styelz (n=yoohoo@2001:5c0:1500:800:0:0:0:1) [NETSPLIT VICTIM] |
| 14:43.25 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 14:43.25 | *** join/#asterisk Obeliks (i=obeliks@gentoo/contributor/Obeliks) [NETSPLIT VICTIM] |
| 14:43.27 | Katty | coffee isn't too bad for you |
| 14:43.28 | Katty | unless you're dumping pounds of sugar into it |
| 14:43.31 | tuxx- | hehe |
| 14:43.31 | tuxx- | nope |
| 14:43.34 | Katty | k |
| 14:43.34 | tuxx- | black coffee ofcourse |
| 14:43.35 | tuxx- | :) |
| 14:43.40 | tuxx- | no sugar, no milk |
| 14:43.41 | tuxx- | \o/ |
| 14:43.50 | ManxPower-work | hugs his coffee |
| 14:43.57 | tuxx- | :) |
| 14:44.07 | Katty | i've had to give up caffeine for awhile...it makes my ears ring |
| 14:44.27 | Katty | it's a blessing in disguise. |
| 14:44.44 | tuxx- | was it not your phone that was ringing? :D |
| 14:45.11 | Katty | one of my doctors put me on an a medication that my body didn't.... agree with |
| 14:45.17 | tuxx- | that sucks |
| 14:45.18 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:45.29 | Katty | and 2 months later, i'm just now getting to the point where the side effect of ringing ears is nearly gone |
| 14:45.40 | Katty | caffeine still irritates it tho, so i avoid it for the most part. |
| 14:45.51 | Katty | the blessing in disguise is i kicked the soda habit |
| 14:45.59 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 14:46.31 | Katty | but i wasn't drinking regular soda in the first place, i was drinking diet soda... |
| 14:46.39 | Katty | which in turn was always making me hungry. constantly. |
| 14:47.15 | tuxx- | doens't diet soda have aspartam? |
| 14:47.21 | tuxx- | which is even worse for you then normal sugar? |
| 14:47.21 | tuxx- | :D |
| 14:47.28 | *** join/#asterisk voipmonk (n=shido6@69.171.158.152) |
| 14:47.37 | Katty | it is a carcenagin(sp) |
| 14:47.56 | Katty | creates headaches in some...i never seemed to have an issue with it other than being ravenous all the time. |
| 14:49.34 | Katty | after discovering it was the soda that had me hungry all the time, i did a little research and found that corn syrup and sodium phosphate can have the same effect...so now i generally avoid things with that in there too. |
| 14:49.38 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:49.53 | Katty | it's really amazing how much stuff has corn syrup |
| 14:50.04 | Katty | or high fructose corn syrup |
| 14:50.12 | Chesther | A study just came out showing some really strong evidence against HFCS. |
| 14:50.20 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 14:50.37 | Katty | Chesther: it wouldn't suprise me. HFCS triggers the same area of your brain that heroine does. |
| 14:50.51 | Chesther | Equivalent diets, equivalent activity level, and people ingesting HFCS built up more body fat compared to people consuming glucose. |
| 14:50.56 | Katty | Chesther: obviously it's NOT heroine...nor does it have the effects, but it triggers the same area of the brain for that Happy Feel Good effect |
| 14:51.22 | Chesther | It also blocks the pathways in your liver that make your body process the fructose, and instead it goes straight into fat-storage mode. |
| 14:51.32 | Katty | it's all a big conspiracy |
| 14:52.01 | Chesther | Nah, just capitalism at its finest. :-) |
| 14:52.03 | Gido-E | like everything :-) |
| 14:52.22 | Gido-E | climate warm up, seems to be a big hoax to. |
| 14:52.43 | Katty | well...the poles are melting |
| 14:52.49 | Katty | and we are losing polar bears at an alarming rate |
| 14:53.02 | Chesther | (they're behind the couch, along with my socks) |
| 14:53.07 | Gido-E | Katty nope :-) |
| 14:53.15 | Gido-E | It is yust told to you. |
| 14:53.54 | Gido-E | It is snowing here, right now! |
| 14:54.55 | Katty | how do you account for a huge chunk of ice that floated away from northern canada? |
| 14:55.16 | Katty | it's 19 square miles...and now floating about the artic ocean |
| 14:55.34 | Katty | how do you account for the markham ice shelf melting? |
| 14:56.19 | sajam | ManxPower-work, i am adding exten to extension file in asterisk, i want it to send the call the router. So can I write, exten => _XXXXXXXX, 1, dial(Sip/ipaddressofrouter) ? |
| 14:56.29 | Chesther | The tricky part is not the short-term symptoms, but trying to figure out whether they represent a long-term trend when, on a geologic time scale, all of industrializaation has happened in the blink of an eye. |
| 14:56.30 | Katty | there's also cracks in one of greenland's northern glaciars... |
| 14:56.56 | Katty | there's also ellesmere island, which was entirely ice back in the 1900s. |
| 14:56.59 | Katty | now it's a bunch of islands. |
| 14:57.23 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 14:57.25 | Katty | it would be interesting to compare some photos of the arctic ocean today vs 20 years ago |
| 14:57.40 | *** join/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 14:57.47 | kannan | hello all, [TK]D-Fender, thank you for y'day help on the voicemail thing, it was just as you had said, the bloke had plugged in an Ans machine to his phone instrument which answered the call before * sent to voicemail |
| 14:58.11 | kannan | todays trouble - |
| 14:58.16 | [TK]D-Fender | kannan: What environment is it in? |
| 14:58.25 | sajam | I want to send the call from Asterisk to a Vrouter, so shall i add IP in the sip protocol? |
| 14:58.34 | kannan | it is a SIP DID, the phones are thru Linksys PAPs on the LAN |
| 14:58.49 | [TK]D-Fender | kannI meant where is this user. Who are they, what do they do? |
| 14:58.52 | Katty | there's also that pesky problem of sea level. |
| 14:59.06 | kannan | The user is in India, they are a call center |
| 14:59.14 | [TK]D-Fender | kannan: SMRT <- |
| 15:00.23 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 15:00.35 | Katty | for nearly the past century, sea level went up by 1.8mm per year, from 1993-2003 it went up to nearly 3.1mm per year |
| 15:00.37 | kannan | [TK]D-Fender - did i get your question wrong? |
| 15:01.26 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 15:01.43 | [TK]D-Fender | kannan: Only the first time. I wasn't sure if you were dealing with residential users, etc... don't remember the scenario and tried to think of a situation where the user might be excusably dumb enough to have done what they did |
| 15:02.23 | [TK]D-Fender | kannan: eg : senility |
| 15:02.51 | kannan | [TK]D-Fender - i myself feel dumb, but the voice on the AM sounded just like the USA male voices that were put into the system and it just didnt strike me |
| 15:03.00 | ManxPower-work | [TK]D-Fender is smarter than he looks |
| 15:03.13 | kannan | anyhow , todays problem coming up right now.. |
| 15:03.58 | ManxPower-work | sajam: if you had spent as much time reading as you do asking here you'd have it solved before. You set up the Cisco peer exactly as you would any other peer in Asterisk, except use host=ip.of.router and don't set a secret= |
| 15:04.37 | ManxPower-work | sajam: Maybe you should start by reading the Asterisk Book |
| 15:04.38 | ManxPower-work | ~book |
| 15:04.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:05.04 | kannan | issue for today , Asterisk Digium Card 1 -> PSTN, Card 2 -> Siemens Hipath 4000. Both are wcte12xp, with h/w echo cancellation module. After continued HDLC aborts on span 2 and seeing this issue at https://issues.asterisk.org/view.php?id=15498 , i had the echo module removed, and things are much better. Only thing is there is some static (veryu mild, like a radio or something) on all the calls |
| 15:05.51 | [TK]D-Fender | ManxPower-work: Don't think you've ever seen me... |
| 15:06.42 | kannan | it doesn't really sound bad, but compared to the calls previously (on the siemens , as well as Asterisk when there was not HDLC abort flooding the CLI) , it is poorer quality, what could be done to try and remove this? |
| 15:07.23 | kannan | thru SIP phones the quality is crystal |
| 15:07.42 | sajam | ManPower-work, i read the book, as well as searching the lists of forums, but no one mentioned what you have mentioned now, and i didn't figure out that we can do it like that. Thanks for your help, but i din't know why you are talking like that, seems all people here born professionals |
| 15:09.36 | voipmonk | I bet you 1/2 read it the first time... |
| 15:10.02 | voipmonk | read it again. :) |
| 15:11.19 | ManxPower-work | voipmonk: nobody can understand all the the Asterisk Book the first time. |
| 15:12.12 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 15:14.00 | kannan | apart from echocancel, echotraining etc what other settings can improve call quality on dahdi ? |
| 15:15.09 | *** join/#asterisk muiro (n=muiro@rrcs-24-56-88-130.ma.biz.rr.com) |
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| 15:18.00 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 15:18.12 | HaMF | hey |
| 15:18.41 | Katty | hi |
| 15:19.25 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 15:19.42 | *** join/#asterisk jpvoip (n=jpvoip@201-34-141-34.fnsce704.e.brasiltelecom.net.br) |
| 15:21.05 | *** join/#asterisk stix (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 15:21.26 | *** join/#asterisk eppigy (n=Dave@216.139.241.100) |
| 15:21.30 | eppigy | WHAT IT DO |
| 15:22.19 | stix | Hi guys. When we receive a call from abroad the number is prefixed +<countrycode> and when we use our phones to call back that number, it is not possible. I have tried to add a route with +. but it doesn't do the trick |
| 15:22.56 | ManxPower-work | stix: The #FreePBX channel is down the hall and to the left |
| 15:23.06 | minotaur01 | lmao |
| 15:23.21 | jpvoip | stix: ${EXTEN}:1 |
| 15:23.33 | ManxPower-work | jpvoip: that is incorrect |
| 15:23.48 | ManxPower-work | that would append a :1 to the end of the number |
| 15:24.08 | ManxPower-work | So if ${EXTEN} is 2125551212 that would make it 2125551212:1 |
| 15:24.36 | ManxPower-work | In regular Asterisk you prefix pattern matches with _, but I don't know how FreePBX does it. |
| 15:24.46 | stix | ${EXTEN:1} is more like it |
| 15:24.54 | ManxPower-work | stix: Correct! |
| 15:24.54 | kannan | but one country code may be more digits than another ? |
| 15:25.12 | stix | yes but it just removes the leading + |
| 15:25.31 | stix | In my case I would replace it with "00" |
| 15:25.59 | ManxPower-work | Goto(00${EXTEN:1},1) |
| 15:26.01 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:26.05 | ManxPower-work | assuming you are staying in the same context |
| 15:26.13 | stix | yes, thank you |
| 15:26.49 | ManxPower-work | stix: "route" is a FreePBX term. |
| 15:27.17 | stix | well sorry - it just makes good sense |
| 15:27.19 | stix | :) |
| 15:27.43 | ManxPower-work | ~freepbx |
| 15:27.44 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:27.56 | ManxPower-work | You can see why some of like to figure out if a user is using FreePBX or not |
| 15:28.13 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 15:28.25 | stix | yes I see |
| 15:29.15 | ManxPower-work | It's always great fun to spend 20 mins helping a user only to find out that nothing you've told them applies to them because they are running some GUI |
| 15:29.45 | *** part/#asterisk ElDios (n=ElDios@93-35-55-83.ip53.fastwebnet.it) |
| 15:29.57 | *** part/#asterisk bbt (n=sam@samuels.id.au) |
| 15:32.25 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:34.23 | stix | Is it impossible to match a "+" in the dialplan? |
| 15:35.04 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:35.04 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:35.12 | *** join/#asterisk dlynes (n=chatzill@bas5-hamilton14-1096762521.dsl.bell.ca) |
| 15:35.26 | ManxPower-work | stix: yes. |
| 15:36.29 | ManxPower-work | stix: some stupid phones (I'm looking at you Polycom) like to take things like + and convert it to URL encoded %xx. If you do a SIP debug you can confirm if Asterisk is getting the + or the %xx form. |
| 15:36.34 | Kchehab | ManxPower-work how to configure spans to use R2 protocol ? |
| 15:36.43 | ManxPower-work | Kchehab: I have no idea. |
| 15:36.50 | *** join/#asterisk jtodd (i=nbtakqbz@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 15:36.50 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 15:37.04 | kannan | mfc/r2, i think voip-info has the howto on patching.. |
| 15:37.22 | TommyBotten | Is 'Asterisk GUI' still in active development? |
| 15:37.27 | Kchehab | guys any idea on how to force span0 to use R2 protocol |
| 15:38.07 | ManxPower-work | TommyBotten: have you considered asking on #asteriskgui? |
| 15:38.28 | TommyBotten | I didn't know it existed ;) |
| 15:38.31 | ManxPower-work | Kchehab: Asterisk does not support R2 as I understand it. There are 3rd party patches |
| 15:38.34 | *** join/#asterisk bcrisp (n=bcrisp@70.102.242.138) |
| 15:38.35 | ManxPower-work | TommyBotten: look at the /topic |
| 15:38.40 | ManxPower-work | ~asteriskgui |
| 15:38.42 | bcrisp | top of the mornin |
| 15:38.44 | *** join/#asterisk RobH (n=RobH@rob.tech.wikimedia.org) |
| 15:38.58 | ManxPower-work | oh, that's nice, they removed #asteriskgui from the /topic |
| 15:39.22 | TommyBotten | I saw |
| 15:39.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) [NETSPLIT VICTIM] |
| 15:39.39 | stix | ManxPower-work, oki, I will have a look at that |
| 15:39.46 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 15:40.05 | *** topic/#asterisk by [TK]D-Fender -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.11 (2009/11/30), 1.6.0.19 (2009/11/30), 1.4.27.1 (2009/11/30), *-Addons 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow #switchvox #asterisk-bugs #asterisk-gui |
| 15:40.17 | ManxPower-work | stix: also remember ALL pattern matches start with _ |
| 15:40.22 | jaytee | Chris Henry died.......more reason never to argue with your wife while driving in a motor vehicle |
| 15:40.24 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 15:40.35 | stix | ManxPower-work, yes that's right |
| 15:40.42 | bcrisp | jaytee: didn't u see the chris rock sketch about that? |
| 15:40.46 | [TK]D-Fender | ManxPower-work: Fixed |
| 15:40.51 | jaytee | bcrisp, nope |
| 15:40.53 | bcrisp | jaytee: or was it dave chapelle |
| 15:43.46 | bcrisp | you guys seen that "evil look" baby vid on youtube? |
| 15:43.48 | bcrisp | im still laughin |
| 15:48.35 | jtodd | ManxPower-work: Asterisk does now officially support R2. http://www.russellbryant.net/blog/2009/03/17/mfcr2-support-in-asterisk-162/ |
| 15:49.32 | Katty | sooo tired. |
| 15:50.15 | ManxPower-work | jtodd: 1) has 1.6.2 actually been released and 2) I would not trust the first release of anything in Asterisk |
| 15:51.50 | *** join/#asterisk bleger (n=yvannepr@topp148-office-nyc.openplans.org) |
| 15:52.52 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:52.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:53.17 | *** join/#asterisk Tim_Toady (n=moi@77.49.239.4.dsl.dyn.forthnet.gr) |
| 15:55.24 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:55.44 | bcrisp | doesn't trust the first release of ANYTHING |
| 15:55.58 | Katty | bcrisp: you were a first release. |
| 15:56.17 | bcrisp | except my own software of course |
| 15:56.31 | bcrisp | heh |
| 15:56.50 | bcrisp | is there a best practices / recommendation for how to implement call forwarding in * ? |
| 15:57.01 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 15:57.07 | ManxPower-work | bcrisp: none at all. There are several ways it can be done. |
| 15:57.12 | bcrisp | i can use a macro in combination with db lookups i suppose.. but dont want to reinvent the wheel |
| 15:57.30 | ManxPower-work | I PERSONALLY think using the phone features to forward is a bad idea -- you have to be at the phone to change the forwarding settings. |
| 15:57.41 | bcrisp | i agree |
| 15:57.52 | bcrisp | i was thinking of adding an exten where users could specify how / when to forward |
| 15:58.44 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:59.17 | bcrisp | also.. i was playing with macros and found that i lose the original ${EXTEN} value when it enters the macro, i.e. exten =>123,1,Macro(bla,SIP/user) -- do i need to store the 123 in another chan variable? |
| 15:59.41 | ManxPower-work | bcrisp: correct. I belive channelvariables.tex talks about that "issue". |
| 15:59.56 | bcrisp | oh ok |
| 15:59.58 | Gido-E | bcrisp it jumps to s extension, as i can remember |
| 15:59.58 | *** join/#asterisk friehmaen (i=freeman@2001:41d0:1:8ed:0:0:6:beaf) |
| 15:59.59 | ManxPower-work | or maybe "core show application macro" |
| 16:00.09 | bcrisp | ya, ill look at it again |
| 16:00.35 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 16:00.42 | Gido-E | bcrisp you can pars your exten as a variable |
| 16:00.58 | eppigy | BNEIN |
| 16:01.13 | bcrisp | yeah, i was using NoOp to trace.. decided to use Set() prior to macro call |
| 16:01.22 | bcrisp | it just adds a bunch of lines to the dialplan that id prefer to ignore |
| 16:01.24 | [TK]D-Fender | bcrisp: pASS IT AS A PARAMETER <- |
| 16:01.27 | ManxPower-work | There is a special variable just for that. |
| 16:01.30 | bcrisp | [TK]D-Fender, ah |
| 16:01.31 | bcrisp | ok |
| 16:02.19 | bcrisp | there appear to be some special vars too |
| 16:02.30 | bcrisp | ${MACRO_EXTEN} |
| 16:03.36 | [TK]D-Fender | bcrisp: If you care about ti, pass it as a parameter... ${MACRO_EXTEN} = GAH |
| 16:03.49 | bcrisp | k |
| 16:03.51 | [TK]D-Fender | bcrisp: If its worth doing, its worth doing EXPLICITLY |
| 16:04.06 | ManxPower-work | bcrisp: the disadvantage of that is that macros are going away to be replaced by Gosubs, which do not support MACRO_EXTEN |
| 16:04.20 | [TK]D-Fender | bcrisp: because you'll tell youself "Oh I'll only call it from extens people will dial, so the pattern is good... **BS** |
| 16:04.24 | bcrisp | should i look to use gosubs instead ? |
| 16:04.24 | ManxPower-work | [TK]D-Fender: Yeah! More softcore Asterisk! |
| 16:04.52 | ManxPower-work | bcrisp: you'd have to check UPGRADE*.txt to see when Macro is going away |
| 16:05.15 | bcrisp | im contemplating using ael |
| 16:05.50 | *** join/#asterisk freezey (n=trees@static-64-61-84-174.isp.broadviewnet.net) |
| 16:06.02 | freezey | has anybody migrated from druid to freepbx? |
| 16:06.13 | freezey | or from druid to just CLI based asterisk? |
| 16:06.17 | [TK]D-Fender | freezey: LOL |
| 16:06.24 | freezey | i need to rip this thing apart man |
| 16:06.28 | freezey | its a piece of junk |
| 16:06.32 | [TK]D-Fender | freezey: So do it |
| 16:06.39 | ManxPower-work | freezey: no you don't. You need to build a new server from scratch. |
| 16:06.44 | freezey | trying to now but druid writes the confs all whacky |
| 16:06.53 | freezey | yeah i thought so |
| 16:06.57 | ManxPower-work | freezey: all GUIs do that |
| 16:07.01 | freezey | such crap |
| 16:07.08 | *** join/#asterisk jtodd (i=gqu9g23b@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:07.08 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:07.46 | ManxPower-work | I think sent a resume to the people that do DRUID. Never followed up since it's a GUI. |
| 16:08.09 | freezey | yeah it sucks |
| 16:08.13 | [TK]D-Fender | freezey: Welcome to the result everyone warned you about. Sorry, your soul is a non-refundable deductible |
| 16:08.36 | bcrisp | [TK]D-Fender, do you work for my car insurance co? |
| 16:08.58 | freezey | haha |
| 16:09.31 | bcrisp | when i left for work this morning there were 2 kids standing on the road smiling at me with rocks in their hands.. right next to my house |
| 16:09.51 | bcrisp | makes you feel secure doesnt it? |
| 16:10.00 | freezey | but of course |
| 16:13.04 | tzafrir | freezey, could you please be more specific? |
| 16:13.27 | tzafrir | DRUID saves some of the configuration in a DB (pgsql) and other parts in config files |
| 16:13.43 | Katty | crank dat druid |
| 16:13.45 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 16:14.01 | tzafrir | It has its own parser of asterisk config files, that supports most of the features of asterisk config files, but not all |
| 16:14.14 | tzafrir | (e.g.: I don't think it understands templates) |
| 16:14.21 | freezey | tzafrir: I wanted to move the current druid i guess over to a different version.. i have to make it available for other people to be able to use |
| 16:15.03 | freezey | tzafrir: and it is not running any type of database |
| 16:16.02 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 16:16.06 | tzafrir | freezey, their distro runs a postgresql server |
| 16:16.34 | freezey | tzafrir: do not see that service running at all |
| 16:16.44 | Geminizer | Hello all. Can I use DISA to allow one outside line establish a call to another outside line? |
| 16:16.46 | *** join/#asterisk xmitter (n=xmitter@c-24-21-212-187.hsd1.or.comcast.net) |
| 16:17.03 | ChannelZ | Yes that's pretty much what it's for |
| 16:17.08 | ManxPower-work | Geminizer: That is the entire reason DISA exists. It's a bad idea and a major security rick |
| 16:17.09 | [TK]D-Fender | Geminizer: With or without it.... yes |
| 16:17.10 | ManxPower-work | and risk |
| 16:17.20 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:18.37 | ChannelZ | I wouldn't say it's any more risky than a badly written dialplan |
| 16:18.45 | [TK]D-Fender | ChanIt isn't |
| 16:18.49 | [TK]D-Fender | ChannelZ: It isn't |
| 16:18.56 | tzafrir | freezey, maybe they switched to mysql, then? |
| 16:18.59 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 16:19.07 | [TK]D-Fender | Chanit POINTS to dialplan.... which is as badly written as you make it |
| 16:19.10 | freezey | tzafrir: that instance is also not running on this machine |
| 16:19.16 | ChannelZ | exactly |
| 16:19.44 | freezey | tzafrir: it looks like it was never configured properly. In the asterisk -rvvv i can see it trying to connect but when i check the settings the database was not setup properly |
| 16:20.15 | tzafrir | how have you installed the system? |
| 16:20.56 | freezey | tzafrir: thats the thing i did not install it somebody else did and it just functions like crap.. so i wanted to fix it |
| 16:21.25 | tzafrir | freezey, and how did that someone install it? from where? |
| 16:21.28 | Geminizer | or would a safer approach be to originate two calls from the pbx, then bridge the channels together? |
| 16:21.33 | freezey | tzafrir: i was going to do just straight up asterisk none of the GUI stuff but other people have to use it and instead of explaining something every 2 seconds i will just put up a GUI |
| 16:21.54 | [TK]D-Fender | Geminizer: Describe what you want to do. |
| 16:21.54 | tzafrir | It is configured as real-time, so you need the DB |
| 16:21.57 | freezey | tzafrir: on a physical box and it looks like they used the druid rom's |
| 16:22.07 | [TK]D-Fender | Geminizer: And evrything is as safe as you make it. |
| 16:22.09 | freezey | rpm's |
| 16:22.20 | tzafrir | They also use an imap server for voicemail |
| 16:22.41 | *** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net) |
| 16:23.00 | tzafrir | The imap server also has accounts configured from the same database |
| 16:23.32 | freezey | tzafrir: the machine is not running imap either |
| 16:23.53 | freezey | tzafrir: like i said it was put together poorly |
| 16:24.04 | tzafrir | Maybe they install just some of the packages? Just the asterisk ones? |
| 16:24.19 | *** join/#asterisk GuillermoHurtado (n=Guillerm@fw1-ext-core.net-uno.net) |
| 16:24.21 | Geminizer | ok... basically I would like to run a script which would originate a call from asterisk and call my cell phone... once I pick up, asterisk calls another outside line... once that second line is established, it is transferred to my cell phone |
| 16:24.30 | *** join/#asterisk xrmx__ (n=rm@host143-252-dynamic.8-87-r.retail.telecomitalia.it) |
| 16:25.00 | GuillermoHurtado | Hello, I'm trying to send a fax to a Asterisk from a Cisco Gw and its not working |
| 16:25.12 | GuillermoHurtado | on my Cisco I have ECM disble |
| 16:25.30 | GuillermoHurtado | do I need to change udptl.conf ? |
| 16:25.31 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.co.uk) |
| 16:25.37 | [TK]D-Fender | Geminizer: So this script already knows the # you want to dial out? |
| 16:25.47 | Geminizer | right |
| 16:25.55 | [TK]D-Fender | Geminizer: then this has nothing to do with DISA |
| 16:25.55 | Gido-E | GuillermoHurtado do you have a error in your log? |
| 16:26.01 | freezey | tzafrir: its possible... |
| 16:26.13 | GuillermoHurtado | I get no fax reply |
| 16:26.16 | [TK]D-Fender | Geminizer: You are then looking to use an AMI or CLI Originate, or CALL FILES |
| 16:26.34 | Gido-E | GuillermoHurtado you can see it going into the fax ap? |
| 16:26.38 | GuillermoHurtado | a log in the cisco? |
| 16:26.39 | [TK]D-Fender | Geminizer: Go look those up, they're all in THE BOOK |
| 16:26.45 | GuillermoHurtado | or in the Asterisk? |
| 16:26.55 | Gido-E | you are on #asterisk :-0 |
| 16:26.59 | freezey | tzafrir: i guess this is just the point where i realize that i am going to have to rip it apart and try to piece it back together |
| 16:27.17 | GuillermoHurtado | hehe right |
| 16:27.20 | Geminizer | I figured out how to originate. With that approach, I could originate two calls, one to my cell and another to the other outside line. Then how would I join those two channels? |
| 16:27.28 | GuillermoHurtado | let me get that log |
| 16:27.29 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:27.41 | Gido-E | GuillermoHurtado please use pastebin! |
| 16:29.57 | [TK]D-Fender | Geminizer: You fail to understand. Originate CALLS the "Channel:" you specify. When it ANSWERS, it dumps them int the DIALPLAN whever you tell it to. So tell it to use an extension that dials OUT again. |
| 16:30.31 | [TK]D-Fender | Geminizer: there is no "join". * calls out. Gets answered. they are then dumped into the dialplan just as if they called IN. You then do whatever the hell you want in your dialplan |
| 16:30.45 | GuillermoHurtado | sorry, what is pastebin? I'm new in this... |
| 16:31.01 | [TK]D-Fender | ~pb |
| 16:31.02 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 16:31.04 | [TK]D-Fender | ^^^^^^^^^ |
| 16:31.23 | Gido-E | [TK]D-Fender i think he noticed it :-) |
| 16:32.17 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 16:33.48 | Geminizer | thank you for the assistance, [TK]D-Fender |
| 16:34.43 | ayeso | How can I tell * to respond to SIP options? |
| 16:35.37 | [TK]D-Fender | ayeso: vi chan_sip.c |
| 16:36.10 | GuillermoHurtado | infobot just for anything over 3 lines? |
| 16:36.13 | ayeso | [TK]D-Fender: if qualify=yes in sip.conf will it respond to that peer? |
| 16:36.40 | [TK]D-Fender | ayeso: qualify tells * to SEND Options packets |
| 16:36.55 | [TK]D-Fender | GuillermoHurtado: No, PASTEBIN is for that. Infobot TOLD you about it |
| 16:37.12 | [TK]D-Fender | Gido-E: Yeah... he "understood" all right :p |
| 16:39.19 | *** join/#asterisk yoshx (n=yoshx@88-138-188-188.adslgp.cegetel.net) |
| 16:39.36 | Gido-E | [TK]D-Fender :-) |
| 16:40.06 | Gido-E | GuillermoHurtado are you trying to talk to a bot? |
| 16:40.47 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
| 16:40.49 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 16:41.02 | *** join/#asterisk came0 (n=came0@rrcs-71-42-25-233.se.biz.rr.com) |
| 16:41.13 | nny | is there any useful reason to have allowguest set to the default "yes" |
| 16:41.27 | Naikrovek | nny depends on a few things |
| 16:41.41 | Naikrovek | nny: in my case it's required due to how my provider sends me calls |
| 16:41.46 | Naikrovek | requred to be yes, i mean |
| 16:41.54 | [TK]D-Fender | nny: So random people can call you without * bitching. |
| 16:41.58 | nny | i have some anoying script kiddies hitting a box I have with a scanner, have alwaysauthreject now set to yes, somehow they were able to ring some phone lines |
| 16:42.11 | Naikrovek | nny: firewall |
| 16:42.16 | nny | oh i have one |
| 16:42.28 | nny | thats not the solution imho but yeah |
| 16:42.41 | Naikrovek | don't dismiss them by saying "script kiddies" - they're telemarketers |
| 16:42.48 | [TK]D-Fender | nny: Maybe you should pay more attention as to what extens you make available to them. |
| 16:42.55 | Naikrovek | and they're much more coordinated than any "script kiddie" |
| 16:43.11 | ManxPower-work | nny: don't set the password to be the same as the device |
| 16:43.17 | nny | ffs |
| 16:43.18 | came0 | hey what sofphone do you guys recomened? the ones I've tried suck. i'm on OSX. I prefer opensource but whatever works |
| 16:43.20 | nny | so many assumptions |
| 16:43.22 | nny | don't be snide |
| 16:43.23 | came0 | softphone* |
| 16:43.31 | nny | 1.) no password = device name |
| 16:43.41 | ManxPower-work | nny: do you have allowguest=no ? |
| 16:43.50 | nny | i just turned it on |
| 16:43.56 | came0 | also, do you get better sound quality out of a hardware SIP phone? |
| 16:44.02 | nny | 'it's nice and vulnerable by default, no? |
| 16:44.07 | ManxPower-work | nny: that will stop them from connecting without a user/password |
| 16:44.14 | nny | yeah that's what I figured |
| 16:44.21 | nny | just suprised it defaults to yes |
| 16:44.22 | ManxPower-work | nny: don't get me started on my rant about the default being allowguest=yes |
| 16:44.27 | *** join/#asterisk pyite_hsv (n=pyite_hs@unaffiliated/pyite) |
| 16:44.28 | Naikrovek | came0: all softphones suck compared to real phones. i use ekiga and it's acceptable. |
| 16:44.30 | nny | ha agreed |
| 16:44.31 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 16:44.47 | ManxPower-work | nny: chan_iax2 doesn't even HAVE an allowguest=no option. Guests are allowed always |
| 16:45.14 | nny | i have alwaysauthreject=yes too, my passwords are alphanumeric, but that keeps them from at least knowing which extensions are available |
| 16:45.18 | *** join/#asterisk OmbreNoir (n=OmbreNoi@modemcable070.50-177-173.mc.videotron.ca) |
| 16:45.25 | nny | but ffs, never knew about guest |
| 16:45.36 | nny | my bad for not reading every parameter in sip.conf |
| 16:45.38 | ManxPower-work | nny: *nod* I suspect not having allowguest=no will fix your issue. |
| 16:45.47 | ManxPower-work | nny: it's new in 1.4 or 1.6 |
| 16:45.51 | nny | ahh |
| 16:46.12 | ManxPower-work | nny: I always set context=INVALID in sip.conf [general] and not have an [INVALID] context in extensions.conf |
| 16:46.13 | nny | iax2 just allows guest connections? |
| 16:46.24 | nny | ok good point, i'll do that too |
| 16:46.25 | ManxPower-work | nny: yes, they come in as the LAST entry in iax.conf |
| 16:46.40 | came0 | hmm maybe I should just buy some real phones then. I'm trying to get the guys in my office to switch to an asterisk set up but they are not happy with the softphones. is there a commercially available softphone thats stable with a good user interface? or is it just harder to get good quailty out a softphone? |
| 16:46.50 | ManxPower-work | no matter what that entry is, so I recommend putting in a dummy entry at the end pointing to an invalid or sandbox context |
| 16:47.12 | ManxPower-work | came0: Linksys/SIPura make good, inexpensive hardphones. |
| 16:47.15 | ManxPower-work | ~phones |
| 16:47.16 | infobot | While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 16:47.45 | voipmonk | grrr |
| 16:47.57 | *** join/#asterisk high-freq (n=hfreq@99.188.101.168) |
| 16:48.04 | Naikrovek | grr what |
| 16:48.17 | ManxPower-work | nny: As my last job, starting about March we started getting about 1 call a week about some poor sod that did not secure their PBX and got "hacked". |
| 16:48.34 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 16:49.34 | nny | ugh heh, yeah I didn't get any auths, but a lot of attempts, alwaysauthreject got rid of that one mostly |
| 16:49.42 | nny | now I am adding allowguest=no to my default configs |
| 16:50.55 | nny | ManxPower-work: i have snort sam on the boxes, but when they are the main gateway device, i don't usually turn it on unless I want a bunch of calls complaining something doesn't work. I eventually will just reload the snort sam configs to only use the sip rules, but the distro provider does not make it easy |
| 16:51.11 | nny | is using clarkconnect/clearos as the base |
| 16:51.16 | nny | mostly |
| 16:52.07 | nny | should probably email the dev and ask if that can be a subscription option for the snort rules, right now it's more designed as an all in one ids, which is a PIA in a network if you don't manage it every day |
| 16:52.50 | Naikrovek | came0: i use polycom phones and love them. |
| 16:53.35 | nny | but over all the fallacy is mine, time to find a good article on hardening asterisk and just make it the norm on my installs, thanks all for the advice |
| 16:54.06 | ManxPower-work | nny: I saw a link for something like that, maybe [TK]D-Fender sent it? |
| 16:54.31 | came0 | MaxPower-work, Naikrovek: Ok thanks guys! I've been having some issues with sound quality that I finally figured out was the softphone. :/ |
| 16:54.58 | Naikrovek | nny: do you have phones on the internet that connect to your system normally |
| 16:55.07 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 16:55.26 | nny | Naikrovek: mostly, depends ont he client |
| 16:55.48 | ManxPower-work | nny: At one my my jobs we solved the problem by just not allowing access from the outside. |
| 16:55.55 | Naikrovek | nny: okay. i've never worked at a place where this would even be considered, but i'm sure people have their reasons. |
| 16:56.12 | ManxPower-work | SIP phones talking to Asterisk talking via PRI to the telco. Not an option for many people, obviously. |
| 16:56.17 | Naikrovek | previously if somoene wanted to use a phone from home they had to have a firewall hole poked just for them, or they had to do it over vpn |
| 16:58.30 | nny | Naikrovek: well, it could be argued that with a proper authenitcation mechanism and setup, you shouldn't have to worry about that. The systems we build are designed for small to medium businesses, and usually they don't have that kind of setup. |
| 16:58.49 | nny | (no vpn, or admin to "poke holes" on a needed basis) |
| 16:59.16 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.co.uk) |
| 16:59.22 | nny | allowguest=no and setting the default context (outside of the auth entries) to "invalid" is a good step though |
| 16:59.39 | Naikrovek | nny: this is true. with proper authentication mechanisms in place, allowguest can probably be turned on, but in *all* other cases I think it should be turned off, because phones don't need to register or authenticate to asterisk to make calls if allowguest=yes |
| 16:59.52 | nny | Naikrovek: er well, it should always be no |
| 16:59.53 | *** join/#asterisk rollitup (i=gtgt@firewalled.with.ipfw.us) |
| 17:00.08 | Naikrovek | nny: then we agree |
| 17:00.08 | nny | Naikrovek: any remote phones have sip.conf entries |
| 17:00.12 | nny | aye |
| 17:00.34 | nny | just was unaware of the default, it's my bad for not knowing though |
| 17:01.01 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:01.15 | Naikrovek | yeah lots of people don't know about that little feature |
| 17:03.03 | johnyjj2 | TommyBotten: You previously told that you haven't used scribblej. I cannot contact its author to ask him if it can work with Sphinx4 (it works with PocketSphinx). Do you know about anything else what can enable communication between Sphinx4 and Asterisk? |
| 17:03.36 | Geminizer | Is there a way to pass arguments from a .call file into a dialplan context? |
| 17:03.39 | *** join/#asterisk Tim_Toady (n=moi@77.49.27.164.dsl.dyn.forthnet.gr) |
| 17:04.35 | *** join/#asterisk ruied (n=ruied@95.69.10.85) |
| 17:04.44 | nny | so if the default context has ways to dial out (access to zap, etc) and the guest account is on, doesn't that mean people could just dial out of the system by default regardless of sip credentials? |
| 17:04.59 | nny | wtf |
| 17:05.06 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.co.uk) |
| 17:05.14 | bcrisp | im gettum |
| 17:05.34 | bcrisp | ... sorry gremlin in my paste buffer |
| 17:05.40 | nny | hehe |
| 17:05.48 | bcrisp | starting typing something here also heh |
| 17:05.54 | rollitup | hi everyone |
| 17:07.10 | *** join/#asterisk MindTheGap (n=caio@mail.lpj.com.br) |
| 17:09.33 | *** join/#asterisk cesar_CR (n=cesar@201.191.135.225) |
| 17:09.48 | *** join/#asterisk testsniper (n=sniper_j@12.25.228.67) |
| 17:10.06 | testsniper | hi all |
| 17:10.21 | rollitup | o hi :) |
| 17:10.32 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:10.33 | testsniper | how are you rollit? |
| 17:10.41 | rollitup | good how are you? |
| 17:10.47 | testsniper | i m fine |
| 17:10.55 | rollitup | :] |
| 17:10.55 | testsniper | are u good in asterisk |
| 17:10.56 | testsniper | ? |
| 17:11.11 | testsniper | coz i need some help |
| 17:11.15 | rollitup | not really |
| 17:11.19 | testsniper | humm |
| 17:11.22 | testsniper | ok |
| 17:11.32 | testsniper | any one how is goooooooood in asterisk |
| 17:11.33 | testsniper | ? |
| 17:11.35 | sajam | testsniper, You need some help, so you have to ask voipmonk, |
| 17:11.36 | testsniper | who* |
| 17:12.04 | sajam | testsniper, he's the expert here. |
| 17:12.07 | Qwell | ~ask |
| 17:12.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:12.11 | testsniper | good |
| 17:13.39 | Naikrovek | nny: yes, people on thepublic internet can simply tell their phones to make calls through your system and your system will happily let them with allowguest=yes |
| 17:13.47 | Naikrovek | nny: lots of people get burned by this |
| 17:13.49 | nny | heh |
| 17:13.51 | nny | no kidding |
| 17:13.58 | *** part/#asterisk sajam (n=chatzill@beta.intelligile.com) |
| 17:14.04 | nny | grepping through the csv it didn't seem to happen, but holy smokes |
| 17:14.16 | Naikrovek | nny: yeah |
| 17:14.51 | Naikrovek | i HAVE to have allowguest=yes due to how my provider sends me calls |
| 17:15.04 | Naikrovek | so i must take other precautions to prevent unauthorized calls. |
| 17:15.08 | nny | i use vitelity, it just has an entry in sip.conf |
| 17:15.13 | nny | with an IP auth |
| 17:16.08 | nny | but I could see how a provider would have to send the calls in a way that didn't allow any auth to happen |
| 17:16.46 | Naikrovek | I register to one IP, their registration server, but I get calls from another server. |
| 17:16.55 | nny | ahh |
| 17:17.10 | Naikrovek | with allowguest=no, I would only be able to receive calls from the registration server, but that's not how they're set up |
| 17:17.22 | *** join/#asterisk jtodd (i=ew0n8x0z@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:17.22 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:17.29 | Naikrovek | so i have to open it wide up and use a firewall to restrict incoming calls. |
| 17:17.55 | nny | yeah I see. couldn't you allow the second server to send sip via an entry that just defined it's ip address? |
| 17:18.23 | tftech | Is it OK to announce if I have Asterisk Hardware for sale in the room? |
| 17:18.54 | nny | Naikrovek: http://pastebin.com/m426a8db1 is what I use |
| 17:19.44 | Naikrovek | tftech: you just did. and i think it's fine |
| 17:20.14 | tftech | Naikrovek - Thanks, I just wanted to check before I said what I had |
| 17:20.17 | Naikrovek | nny: i love that "insecure=very" lol |
| 17:20.28 | nny | Naikrovek: yeah heh |
| 17:20.33 | nny | Naikrovek: but it's just for that IP |
| 17:20.54 | tftech | I have a TE122 and a TDM411B with one FXS and one FXO, both brand new |
| 17:21.00 | *** join/#asterisk borga454 (n=alex@80.172.25.71) |
| 17:21.19 | Naikrovek | why you selling them |
| 17:23.08 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 17:23.15 | tftech | Naikrovek - Because I just got them from the Faststart class and I dont need them |
| 17:23.22 | Naikrovek | ah |
| 17:23.26 | tftech | My kids could use a Wii |
| 17:23.29 | tftech | and me a PS3 |
| 17:23.31 | Naikrovek | fair enough |
| 17:23.36 | Naikrovek | yes i want ps3 as well |
| 17:23.41 | Naikrovek | final fantasy xiii comes out in march |
| 17:23.51 | came0 | When you say softphones suck compared to hardphones do you mean in sound quailty or just general usuability |
| 17:23.52 | came0 | ? |
| 17:23.53 | tftech | I still have the original XBOX |
| 17:23.54 | tftech | lol |
| 17:23.56 | nny | it's for pc too Naikrovek, no? |
| 17:24.05 | nny | I have a wii, i don't even play it |
| 17:24.08 | Naikrovek | nny: ps3 & 360 i belive, not sure about pc |
| 17:24.17 | tftech | Wii is for Kids and Wife |
| 17:24.19 | nny | even hacked it for homebrew, still kinda boring heh |
| 17:24.21 | Naikrovek | nny: my 4yo daugther MAKES me play NSMBWii daily |
| 17:24.22 | nny | indeed |
| 17:24.39 | nny | haven't played it yet, is it any good? |
| 17:24.42 | Naikrovek | homebrew would be neat if it weren't all complete crap |
| 17:24.44 | Naikrovek | nny: yes |
| 17:24.44 | sebbl | yes |
| 17:24.45 | *** join/#asterisk ruyo (n=psantos@195-23-253-223.net.novis.pt) |
| 17:24.58 | nny | have to check it out |
| 17:25.01 | tftech | Still not 100% sure about PS3 or 360 |
| 17:25.05 | tftech | still debating |
| 17:25.05 | mchou | came0: both |
| 17:25.18 | borga454 | Hello all, I have a strange problem with my digium fxs board. When it is connected to a fax machine often the fxs channel jams in the state offhook. I have to restart the dahdi drivers to go to the onhook state. |
| 17:25.31 | nny | i gave my 360 away :\ |
| 17:25.36 | tftech | lol |
| 17:25.36 | Naikrovek | gave it away? |
| 17:25.37 | Naikrovek | rrod? |
| 17:25.40 | nny | to a buddy of mine. |
| 17:25.41 | borga454 | anyone has an idea how to solve this?? :) |
| 17:25.51 | nny | nah it works |
| 17:25.54 | nny | just never played it |
| 17:26.06 | nny | my buddy gets way more use out of it |
| 17:26.11 | mchou | nny: was it blacklisted? :) |
| 17:26.22 | tftech | I figure people in here would have use for the hardware |
| 17:26.23 | nny | nah |
| 17:26.30 | [TK]D-Fender | came0: Imagine a shitty sound card witha shitty headset vs a dedicated device where you don't have a fugly interface, have to rush trying to find a keypad just so you can enter a VM pass before it kicks you out, etc |
| 17:26.30 | nny | mchou: this was waay before all that stuff |
| 17:26.31 | tftech | I dont need the IP330 either, woudl see it |
| 17:26.41 | tftech | Sell it |
| 17:26.43 | nny | mchou: heh the thing hasn't been online in so long it has the original dashboard |
| 17:26.45 | [TK]D-Fender | came0: Softphones are lame shit that straps your head to a PC. |
| 17:27.23 | Naikrovek | borga454: you'll likely need to find out from your telco provider how they signal the other end is onhook and configure your board to look for the same signal. there are different signals and they must match. reverse polarity, short, drop voltage, etc. |
| 17:27.36 | mchou | [TK]D-Fender: yeah, but they are good for testing and sanity check |
| 17:29.09 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:30.01 | tftech | I dont like softphone for production because they rely on the PC not crashing or hanging up |
| 17:30.14 | tftech | and we all know windows is perfect and never crashes |
| 17:30.15 | borga454 | Are you talking about the tones configuration?? Normally it works thine. One or two times a month the fxs board jams on the offhook state and I can't get out of it |
| 17:30.18 | Naikrovek | softphones are just crap period |
| 17:30.42 | tftech | only use the softphones if traveling or testing |
| 17:30.45 | Naikrovek | borga454: i'm talking the signal the telco sends your board when the other end disconnects |
| 17:31.00 | Naikrovek | borga454: but if it only happens once or twice a month it's not a signalling configuration issue |
| 17:31.24 | Naikrovek | borga454: does unplugging the physical line from the board reset the board? |
| 17:31.46 | borga454 | no. It stays the same |
| 17:32.25 | borga454 | I heaven connect there a phone, instead of a fax. The same result. |
| 17:32.45 | Naikrovek | borga454: the party line in here usually is to not let asterisk handle fax duties, because of things like this, but there may be someone in here who's seen the issue you're seeing. i don't know the answer so i'll digress |
| 17:33.19 | bcrisp | im playing with users.conf and find that when i set up a new user with alternateexts, i still receive "extension not found" notices in CLI |
| 17:33.19 | tftech | Wife faxes all the time and I have not seen this |
| 17:33.32 | tftech | sending and recieving |
| 17:34.24 | borga454 | thanks for the effort! ;) |
| 17:34.27 | bcrisp | i.e. context = somecontext alternateexts = 1000 when another users dials 1000 within the same context i receive extension not found |
| 17:34.48 | borga454 | In the channels I get the state reserved?? can this help? |
| 17:36.27 | [TK]D-Fender | bcrisp: Don't... |
| 17:36.45 | bcrisp | [TK]D-Fender, users.conf lost cause? |
| 17:36.57 | [TK]D-Fender | ~users.conf |
| 17:36.58 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:37.10 | bcrisp | ah |
| 17:37.48 | Qwell | ~book |
| 17:37.48 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:38.12 | bcrisp | hmm |
| 17:42.44 | bcrisp | regexten, where have you been all my life? (ah yes, in sip.conf.sample) |
| 17:43.18 | leifmadsen | sip.conf.sample is incredibly useful to read |
| 17:44.07 | bcrisp | ya.. i keep going back to it |
| 17:44.13 | bcrisp | when something "just seems too hard" |
| 17:44.19 | [TK]D-Fender | bcrisp: I've never really seen a need for regexten. |
| 17:44.57 | bcrisp | [TK]D-Fender, i was thinking at first it was going to help with something |
| 17:45.02 | *** join/#asterisk danj1980 (n=dan@91.108.6.182) |
| 17:46.05 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 17:46.52 | Katty | hungries :< |
| 17:47.39 | danj1980 | Hi, does anyone have a cheap SIP wholesaler for call to UK mobiles? |
| 17:47.54 | Chainsaw | You're in the UK. Nothing's cheap there. |
| 17:48.45 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 17:50.43 | bcrisp | im trying to associate a sip device with an extension number. the name is SIP/username. When a call originates from this device, i want to be able to somehow gather their extension number |
| 17:50.57 | gr0mit | danj1980, what are you looking for exactly? |
| 17:51.01 | bcrisp | im pretty sure im going about things in the wrong way |
| 17:51.09 | p3nguin | bcrisp: callers don't have extension numbers. |
| 17:51.10 | ManxPower-work | bcrisp: You cannot do that. You can only get the callerid info or the sip userid |
| 17:51.18 | bcrisp | ya.. |
| 17:51.26 | file | setvar. |
| 17:51.39 | bcrisp | i guess i could make their voice mail box # be the same as the extension used to dial these users |
| 17:51.56 | p3nguin | You could... if you wanted to. |
| 17:51.59 | ManxPower-work | bcrisp: most of my users have their caller*ID set to their extension |
| 17:52.05 | file | setvar=ORIGINATING_EXTENSION=100 |
| 17:52.49 | gr0mit | danj1980, if you are looking for a UK provider let me know via PM |
| 17:53.03 | ManxPower-work | bcrisp: you can also use function CUT to remove the SIP/ and the trailing -blah from ${CHANNEL} |
| 17:53.13 | p3nguin | manxpower-work: It certainly makes it easier to call a person back when they do. |
| 17:53.17 | bcrisp | ManxPower-work, yeah ive played with that |
| 17:53.30 | file | bcrisp: 'twas for you above. |
| 17:53.32 | ManxPower-work | bcrisp: file's idea may be the best option for you. |
| 17:53.50 | file | setvar is there for this reason, so you can set information you want like that |
| 17:54.19 | bcrisp | optimally tho id like it when the device first registers |
| 17:54.31 | *** join/#asterisk rigid (n=dude@95-91-83-22-dynip.superkabel.de) |
| 17:54.45 | bcrisp | i think using caller id / vm # might be easier |
| 17:55.04 | gr0mit | hmmm, danj1980 has evaporated! |
| 17:55.10 | rigid | ahoy... i have a "no audio" problem with a remote peer, is there a way I can debug RTP? |
| 17:55.19 | joako | In asterisk 1.4 can I disable the !rm -Rf / ability? |
| 17:55.21 | ManxPower-work | bcrisp: setting the extension and the mailbox to the same value does have it's uses and is very common |
| 17:55.28 | voipmonk | be back in 40 |
| 17:55.49 | Qwell | joako: sure. don't run Asterisk as root |
| 17:56.23 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 17:56.25 | bcrisp | ManxPower-work, cool - ya i figured i could extract the mailbox using SIPPEER function |
| 17:56.26 | [TK]D-Fender | [12:55]<joako>In asterisk 1.4 can I disable the !rm -Rf <- that is not an Asterisk command |
| 17:56.48 | gr0mit | yes, what ability is that?! |
| 17:56.56 | *** join/#asterisk PabloM (n=user@190.245.131.114) |
| 17:57.24 | PabloM | hi, is there any way to relate AMI Events with the action that originated them? |
| 17:57.32 | joako | Type e.g !uptime in your asterisk console.... |
| 18:00.13 | PabloM | hello? is there anyone using AMI? |
| 18:00.38 | Naikrovek | joako: the ! tells asterisk to run the command in the OS and return the output. it's not an asterisk command itself |
| 18:00.41 | kannan | joako : e.g !uptime says command not found ? |
| 18:00.58 | dlynes | joako: You'll need to modify the C code |
| 18:01.29 | ManxPower-work | in 1.6 there is some CLI access control, but I don't know much about it. See the .sample config file. |
| 18:01.44 | dlynes | joako: in main/asterisk.c, you have two functions: consolehandler() and remoteconsolehandler() |
| 18:02.04 | *** join/#asterisk abatista (n=chatzill@63.214.236.169) |
| 18:02.30 | dlynes | joako: Just comment out the section from if( s[0] == ...) down to the last line before the final closing brace of the function |
| 18:03.05 | dlynes | joako: and then rebuild and reinstall |
| 18:04.13 | dlynes | ManxPower-work: based on what I see in the code, I would imagine the access control wouldn't affect these two functions |
| 18:04.27 | dlynes | ManxPower-work: unless there's something obvious I'm not seeing |
| 18:04.43 | rigid | is "exten => exten,2,Dial(peer1&peer2&...&peerN)" the right way to have incoming calls ring all peers and achieve a first-come-first-serve forward? |
| 18:05.15 | Naikrovek | rigid: that's how to call more than one phone at a time, and i believe, yes, the first person to pick up will stop the other phones from ringing |
| 18:05.15 | rigid | for I get strange "no-audio" problems with some peers |
| 18:05.46 | dlynes | rigid: if you're getting a lack of audio on some peers, you've got other issues |
| 18:05.52 | Naikrovek | yes |
| 18:05.56 | Naikrovek | listen to dlynes |
| 18:06.05 | rigid | ah thanks... that's what I wanted to know |
| 18:06.29 | dlynes | rigid: also, if you're calling that many extensions at once, you're probably better off using a queue, and using the 'ringall' strategy |
| 18:06.40 | dlynes | rigid: it's much cleaner looking code |
| 18:06.57 | ManxPower-work | dlynes: I just remember some mention of it in UPGRADE*.txt and the sample configs. It's not something that is currently useful to me. |
| 18:07.24 | dlynes | ManxPower-work: nod...I just thought I'd mention what I just found in the code, so you'd know for the next person you talk to |
| 18:07.35 | dlynes | ManxPower-work: mostly because you're on here more than I am |
| 18:08.00 | rigid | dlynes: it's just 4 peers for now, but i'll read about the "ringall" strategy (never heard of it before) |
| 18:08.49 | dlynes | rigid: it basically means it'll ring everyone in the queue until somebody picks up the phone |
| 18:09.03 | dlynes | rigid: it's more or less the same thing as chaining them altogether in one dial statement |
| 18:09.31 | rigid | dlynes: i see, looks like the cleaner way |
| 18:09.35 | dlynes | rigid: but you get to additionally have features like telling the caller they're next in line, or that they should expect a 3 hour wait, or whatever |
| 18:09.43 | p3nguin | It kinda depends if you want the caller to be on hold while all your phones are being rung. |
| 18:10.36 | rigid | ah cool... :) |
| 18:10.36 | dlynes | rigid: your way, if all the phones are busy, they'll get a busy signal |
| 18:10.36 | dlynes | rigid: with a queue, they'll get to listen to annoying music instead |
| 18:10.54 | p3nguin | And after one phone isn't busy, it should ring from the queue. |
| 18:12.09 | *** join/#asterisk RypPn_OuT (i=TuMbL@rosscom.co.uk) |
| 18:12.41 | rigid | dlynes: ah, then my way is right. it's more the fallback for the case that the "receptionist" can't receive calls, then anybody may answer it |
| 18:12.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:13.03 | jkroon | hi guys, is it possible to evaluate a dialplan function from the CLI ? |
| 18:13.09 | p3nguin | I would still want to use queue. |
| 18:13.32 | [TK]D-Fender | jkroon: No |
| 18:13.41 | jkroon | ta |
| 18:13.56 | p3nguin | Dial(SIP/recept,30) then Queue(myqueue) == if recept doesn't answer phone after 30 seconds, it will go into a queue where someone else can answer it. |
| 18:13.58 | rigid | p3nguin: atm i have to solve "no audio" problems, but i'll fiddle with queues when it works :) |
| 18:14.19 | *** join/#asterisk bleger (n=yvannepr@topp148-office-nyc.openplans.org) |
| 18:15.05 | Geminizer | is there a way to set the caller id in a dialplan? |
| 18:16.06 | etfonhomey | Geminizer, lookup the dialplan applications on www.voip-info.org |
| 18:16.32 | *** part/#asterisk etfonhomey (n=etfonhom@74-143-192-74.static.insightbb.com) |
| 18:16.44 | [TK]D-Fender | Geminizer: "core show function CALlerid" |
| 18:17.07 | [TK]D-Fender | Geminizer: "core show function CALLERID" |
| 18:18.42 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 18:25.24 | bcrisp | grr |
| 18:27.38 | high-freq | fender, i almost got it working ;) |
| 18:29.45 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 18:31.13 | ariel_ | ~nat |
| 18:31.14 | infobot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 18:31.24 | ariel_ | ~sipnat |
| 18:31.25 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:31.43 | nny | ~nattyice |
| 18:31.48 | nny | damn |
| 18:32.03 | nny | this place is so not bro friendly |
| 18:32.24 | ariel_ | bro? |
| 18:32.50 | nny | bro: Obnoxious partying males who are often seen at college parties. |
| 18:33.09 | rigid | what could be the reason for not getting any packet to the UDP ports defined in rtp.conf when having "no audio" problems? |
| 18:33.22 | rigid | could it be that RTP isn't used after all? |
| 18:33.32 | nny | rigid: nat? |
| 18:33.37 | *** join/#asterisk Alagar (n=Administ@122.164.36.14) |
| 18:33.50 | *** join/#asterisk RypPn (i=TuMbL@rosscom.co.uk) |
| 18:33.55 | rigid | nny: hm, i have nat=off on remote peers |
| 18:33.57 | p3nguin | I'm still having trouble getting faxes using iaxmodem and hylafax. http://pastebin.com/d5b39bd89 |
| 18:34.04 | nny | rigid check the link above |
| 18:34.06 | nny | ~sipnat |
| 18:34.07 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:34.18 | nny | the first link (aocomputing) has helped me plenty |
| 18:34.41 | bcrisp | hmm |
| 18:34.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:35.56 | *** join/#asterisk montu (n=dpearson@mail.databalance.com) |
| 18:36.27 | p3nguin | In the pastebin, I have included the log of a failed received fax, as well as the log for the only successful fax I ever did get. |
| 18:37.59 | rigid | nny: normally there shouldn't be any NAT between the nodes |
| 18:38.34 | p3nguin | * and the phone are both on the same local network? |
| 18:38.35 | nny | rigid: is there NAT now? |
| 18:38.43 | TSM2 | for thoes that a server room savvy, for a norm, you require 1watt of cooling power for every 1watt of server power, this assumes a sealed room without natural ventilation |
| 18:38.47 | TSM2 | is this correct? |
| 18:38.48 | *** join/#asterisk voipmonk (n=shido6@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com) |
| 18:39.07 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 18:39.14 | rigid | nny: nat=no everywhere... |
| 18:39.19 | nny | rigid: er |
| 18:39.24 | nny | rigid: what p3nguin said |
| 18:39.30 | p3nguin | Setting nat=no does not mean NAT does not exist... |
| 18:39.37 | p3nguin | It just means you didn't enable it. |
| 18:39.38 | nny | are the phones and the asterisk box on the same network? |
| 18:39.54 | rigid | p3nguin: ofcoz :) |
| 18:40.23 | p3nguin | So let's get back to something that matters. Are Asterisk and the phone on the same network? |
| 18:41.19 | *** join/#asterisk Alagar (n=Administ@122.164.36.14) |
| 18:46.08 | p3nguin | oooookay, then. |
| 18:47.38 | rigid | hm... i've found this: "Dec 17 19:46:19 DEBUG[3228]: chan_sip.c:3217 sip_alloc: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP)" is this good or bad? :) |
| 18:47.45 | rigid | my thought: no RTP -> no audio |
| 18:49.32 | p3nguin | Is there some reason that you won't answer the question about the location of the phone? |
| 18:49.37 | ManxPower-work | rigid: not all SIP transactions have audio |
| 18:49.52 | *** join/#asterisk dwxreaper (n=quinn@66.165.129.81) |
| 18:50.03 | rigid | p3nguin: yes, it weren't highlighted... i'm reading/fiddling asterisk in the meanwhile :) |
| 18:50.04 | dwxreaper | how do I see logs of old phone calls, I'm looking for a phone number that dialed in recently |
| 18:50.05 | ManxPower-work | In fact, unless your use your phone all the time MOST SIP transactions won't have audio/RTP |
| 18:50.40 | [TK]D-Fender | dwxreaper: Go look at your CDR's |
| 18:51.03 | rigid | p3nguin: to be correct: the phone is a SIP-router on a public ip, it connects to an asterisk server behind a NAT (nat=yes in [general]) i configured "nat=no" only for local nodes |
| 18:51.14 | dwxreaper | no idea what that means, I want to see a number dialed |
| 18:51.26 | [TK]D-Fender | ~cdr |
| 18:51.27 | infobot | [cdr] Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
| 18:51.31 | p3nguin | The number dialed is IN the CDR. |
| 18:51.37 | rigid | ManxPower-work: ah, so the option is only for this transaction, not in general for the peer |
| 18:51.46 | dwxreaper | okay great, where is the log file |
| 18:51.54 | ManxPower-work | dwxreaper: look in /var/log/asterisk |
| 18:52.02 | dwxreaper | I did, what are they named by default |
| 18:52.02 | ManxPower-work | We made it hard to guess on purpose. |
| 18:52.28 | ManxPower-work | dwxreaper: The #freepbx channel is two door down the hall to the left. |
| 18:52.30 | ManxPower-work | ~freepbx |
| 18:52.31 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 18:52.31 | *** join/#asterisk ecto (n=weechat@user-24-236-95-118.knology.net) |
| 18:52.39 | rigid | dwxreaper: the complete log? there is no error message... just missing RTP debug output compared to working calls |
| 18:52.54 | dwxreaper | damn can't simply tell someone how to look at a call in asterisk |
| 18:52.55 | dwxreaper | lame |
| 18:53.09 | [TK]D-Fender | dwxreaper: You've been told to the folder to search in |
| 18:53.14 | ManxPower-work | dwxreaper: BECAUSE WE DIDN'T SET UP YOUR SYSTEM SO WE DON'T KNOW HOW YOU CONFIGURED ASTERISK |
| 18:53.25 | rigid | dwxreaper: start asterisk -r and type "sip debug" or "debug level ..." depends on what you want to "see" |
| 18:53.27 | dwxreaper | default |
| 18:53.34 | dwxreaper | never mind |
| 18:53.37 | dwxreaper | to hard |
| 18:53.42 | rigid | dwxreaper: "help" is also a good starting-point |
| 18:53.43 | ManxPower-work | On MY systems they are in /var/log/asterisk/company_name-DDMMYYY.csv |
| 18:53.44 | [TK]D-Fender | dwxreaper: there are 2 CDR folders in there, go look in them |
| 18:53.58 | dwxreaper | there we go |
| 18:54.31 | rigid | is even more confused now :) |
| 18:55.01 | [TK]D-Fender | rigid: He wants call history, not live debug of calls that are about to come in |
| 18:55.30 | ManxPower-work | i.e. He wants Detailed Call Records |
| 18:55.46 | [TK]D-Fender | Or Call Detail Records :) |
| 18:56.58 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 18:57.09 | rigid | ah |
| 18:58.14 | rigid | is there a difference between "Nat : Always" and "Nat : RFC3581" that could cause a no-audio problem? |
| 18:58.30 | [TK]D-Fender | rigid: Go read the guide |
| 18:59.16 | rigid | [TK]D-Fender: the 2 urls? aocomputing.net was helpful about canreinvite, but that didn't solved it... i knew the other already |
| 18:59.49 | [TK]D-Fender | rigid: And do you think we trust you did it right? Or that you followed all of the directions? |
| 19:00.21 | rigid | [TK]D-Fender: i could give proof? :) |
| 19:00.41 | rigid | i also used "externalip = hostname" instead of "externalhost" but that also didn't solve it |
| 19:01.14 | ManxPower-work | rigid: It wouldn't since externalip ONLY takes an IP. |
| 19:01.36 | ManxPower-work | I think it's externip, but I'd have to check to be sure. |
| 19:01.48 | rigid | ManxPower-work: yes i did it wrong in the beginning... but there were no error-messages i noticed... just read it on that website |
| 19:02.08 | rigid | ManxPower-work: yes, externip |
| 19:02.08 | ManxPower-work | rigid: Asterisk does not complain about invalid options. |
| 19:02.17 | *** join/#asterisk errotan (n=errotan@81.0.115.3) |
| 19:02.29 | rigid | asterisk is too brave ;) |
| 19:02.36 | ManxPower-work | you could put youkilledkenny=yes in sip.conf and Asterisk would not complain |
| 19:02.54 | ManxPower-work | rigid: no, it allows external programs to store their settings in Asterisk's config files. |
| 19:03.11 | rigid | ha "The host specified by externip can be either a fully qualified domain name (FQDN) or an IP address." from http://www.voip-info.org/wiki/view/tips :) |
| 19:03.33 | rigid | ManxPower-work: why would I want to do that? |
| 19:03.48 | [TK]D-Fender | rigid: taht WIKI line is WRONG |
| 19:04.02 | Qwell | [TK]D-Fender: mmm, not really. as long as it's static |
| 19:04.26 | Qwell | maybe |
| 19:04.57 | rigid | Qwell: only <v1.2 |
| 19:05.13 | Qwell | rigid: ? |
| 19:05.14 | ManxPower-work | rigid: thousands of reasons. I used the feature for voicemail.conf to set a destination number for the system to call up the user and tell them they have voicemail |
| 19:05.41 | Qwell | wait, you used the option "externalip"? |
| 19:06.01 | *** join/#asterisk jtodd (i=ey17hsk7@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 19:06.01 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:08.52 | *** join/#asterisk l2cache (n=l2cache@c-98-213-134-102.hsd1.il.comcast.net) |
| 19:09.10 | l2cache | Hello |
| 19:11.08 | l2cache | I have an asterisk pbx with xorcom FXS ports. Those FXS ports go to to an LG Nortel system. When a call goes over the FXS -> FXO to LG Nortel, they pick up and hear the DTMF tones for the FXS extension number. ANy ideas? |
| 19:11.31 | GuillermoHurtado | Hello, can anyone help me use PASTEBIN? |
| 19:11.43 | tzafrir | ~pb |
| 19:11.44 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:12.21 | tzafrir | l2cache, where exactly do you hear that DTMF noise? On a phone connected to the Nortel PBX? |
| 19:12.41 | l2cache | Yes..They pick up, and hear the DTMF for the exact extension dialed from asterisk |
| 19:12.43 | p3nguin | guillermohurtado: Go to pastebin.com, paste your text into the box, add your name, click submit, give us the resultant URL. |
| 19:12.44 | ManxPower-work | l2cache: stop sending the DTMF or make your PBX read the DTMF |
| 19:13.20 | l2cache | How can I stop sending it? Dial (blank at g1?) |
| 19:13.32 | ManxPower-work | Dial(Zap/g1) |
| 19:14.02 | *** join/#asterisk galeras (n=galeras@190.146.13.27) |
| 19:14.08 | l2cache | I don't think I can do that with FreePBX in the gui, since outbound routes point to the Trunk |
| 19:14.26 | l2cache | and pass whatever digits in the outbound routes for dial pattern |
| 19:14.32 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:14.33 | ManxPower-work | l2cache: Not our problem. The FreePBX channel is down the hall and to the left |
| 19:14.40 | l2cache | haha |
| 19:14.40 | ManxPower-work | ~guis |
| 19:14.41 | infobot | well, guis is "FreePBX/Trixbox is to Asterisk as Windows 95 is to DOS" |
| 19:14.46 | ManxPower-work | ~freepbx |
| 19:14.47 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:15.02 | ManxPower-work | There's 5 mins of my life I'll never get back. |
| 19:15.08 | l2cache | I didn't ask for FreePBX help, just stating |
| 19:15.33 | l2cache | thanks |
| 19:16.26 | GuillermoHurtado | thanks p3nguin |
| 19:16.49 | GuillermoHurtado | for small messages, is it ok to write directly here? |
| 19:16.59 | [TK]D-Fender | GuillermoHurtado: 3 or less |
| 19:17.19 | GuillermoHurtado | ok |
| 19:17.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:17.31 | bcrisp | except of course if you are lauding me.. then take as many lines as you like |
| 19:17.36 | p3nguin | If it's something you would normally say in conversation, put it here. |
| 19:17.41 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:19.02 | GuillermoHurtado | Hello, I need to disable T38 error correction mode and configure redundancy for T38... |
| 19:19.38 | GuillermoHurtado | can I do this just modifiyng udptl.conf ? |
| 19:20.47 | GuillermoHurtado | maybe, comment line: T38FaxUdpEC = t38UDPFEC |
| 19:20.58 | GuillermoHurtado | and uncomment line: T38FaxUdpEC = t38UDPRedundancy |
| 19:21.01 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 19:24.38 | ManxPower-work | GuillermoHurtado: not many people on this channel use T.38 |
| 19:25.05 | coppice | t38UDPFEC is best avoided |
| 19:26.13 | *** join/#asterisk Micc (n=Micc@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:26.56 | Micc | If I turn on Qualify and a peer goes unreachable, will asterisk still attempt to connect to that peer if I dial it? |
| 19:27.14 | *** join/#asterisk ajp01 (n=ajp01@dsl081-067-143.sfo1.dsl.speakeasy.net) |
| 19:27.26 | ManxPower-work | Micc: no. |
| 19:27.46 | Micc | Isn't there a setting to make it dial even if qualify failed? |
| 19:27.58 | Micc | I know there is something like that for realtime. |
| 19:28.04 | Qwell | that's the entire point of qualify... |
| 19:28.10 | ManxPower-work | Micc: not that I'm aware of. the REASON for qualify is to not dial the peer if it's down |
| 19:28.10 | Qwell | if you want that, don't use qualify |
| 19:28.31 | Micc | I just want to use it to see ping times. |
| 19:29.00 | ManxPower-work | Micc: Qualify does not measure ping times |
| 19:29.19 | rigid | just to be sure: i don't need port forwarding if asterisk ip==NAT router ip and listening on 0.0.0.0, right? |
| 19:29.49 | [TK]D-Fender | rigid: Pardon? How would 2 devices be using the same IP? |
| 19:30.02 | rigid | [TK]D-Fender: why 2 devices? |
| 19:30.21 | rigid | asterisk server ip == NAT router ip |
| 19:31.55 | ajp01 | i had quick question about using originate in the asterisk command line, if somebody has a second |
| 19:31.57 | p3nguin | So Asterisk runs on the router? |
| 19:32.09 | rigid | p3nguin: exactly |
| 19:32.51 | p3nguin | If Asterisk is on the router and the router has a public IP address which Asterisk is listening on, then nat needs to =no in the general section. |
| 19:32.56 | *** part/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 19:33.06 | rigid | ah |
| 19:33.17 | p3nguin | because it isn't behind NAT. |
| 19:33.19 | coppice | GuillermoHurtado: If you specify t38UDPRedundancy in the conf file you will always get t38UDPRedundancy. if you specify t38UDPFEC, the system will try to use t38UDPFEC, and negotiate back to t38UDPRedundancy if the far end doesn't support t38UDPFEC. t38UDPFEC is supposed to be the more sophisticated way to do things, but it actually sicks and is best avoided. if the far end doesn't like long... |
| 19:33.20 | coppice | ...packets you may need to adjust the amount of redundancy sent |
| 19:34.00 | *** join/#asterisk jnfuller (n=jnfuller@d64-180-195-142.bchsia.telus.net) |
| 19:34.19 | rigid | p3nguin: and canreinvite=yes ? |
| 19:34.25 | p3nguin | rigid: And you don't need port forwarding, but you still have to allow the ports to pass into the system. |
| 19:34.53 | rigid | p3nguin: i did... but the peer doesn't send data to them |
| 19:34.59 | p3nguin | rigid: Unless your devices are on the same network, I would set canreinvite=no for them. |
| 19:35.20 | GuillermoHurtado | ok Thanks |
| 19:36.00 | p3nguin | rigid: re ports: if you use iptables, you allow them in the input chain rather than forwarding them past NAT. |
| 19:36.08 | GuillermoHurtado | now, how do I set the amount of redundancy? I have set it on my Cisco to 2 redundancy packet for Low Speed connections |
| 19:36.23 | Micc | Qualify doesn't measure ping times, but it does tell me how long it takes for the phone at the other end to respond. |
| 19:36.42 | rigid | p3nguin: yep |
| 19:36.57 | rigid | p3nguin: 5060 and 10000-20000 |
| 19:38.07 | ManxPower-work | Micc: you can always set up two peers, one with qualify and one without and just dial using the one without. |
| 19:39.26 | Micc | ManxPower, thats an interesting idea, I didn't think of that. I wouldn't want to do it for everyone, but it could come in handy sometime. |
| 19:41.59 | *** join/#asterisk blkry (n=chatzill@64.147.222.130) |
| 19:43.00 | *** part/#asterisk jnfuller (n=jnfuller@d64-180-195-142.bchsia.telus.net) |
| 19:43.28 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 19:45.09 | blkry | hello, changes I make to vm_general.inc and vm_vm_email.ing are not being recognized. reload, reboot still nothing. is there a file farther upstream that is messed up? |
| 19:45.37 | Qwell | blkry: what are those files? |
| 19:45.49 | Qwell | they aren't config files that Asterisk reads |
| 19:46.01 | blkry | oh? |
| 19:46.15 | Qwell | Are you using some GUI? |
| 19:46.23 | blkry | yes freepbx |
| 19:46.27 | Qwell | #freepbx |
| 19:46.33 | [TK]D-Fender | blkry: Move along... |
| 19:46.40 | *** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi) |
| 19:47.03 | ManxPower-work | ~freepbx |
| 19:47.04 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:47.06 | blkry | those files are in /ect/asterisk/vm_email.inc, but ok thanks |
| 19:47.17 | ManxPower-work | blkry: those files do NOT come with Asterisk |
| 19:47.37 | rigid | ok, for everyone interested: NEVER set "externhost = domain" to a domain that is bound to a local ip in /etc/hosts :-/ |
| 19:48.08 | [TK]D-Fender | SMRT :p |
| 19:48.13 | rigid | thanks p3nguin & all the others for your support |
| 19:48.23 | bcrisp | thanks for flying Asterisk |
| 19:48.26 | ManxPower-work | rigid: You're trying to do everything but what yo need to do |
| 19:49.16 | rigid | ManxPower-work: what do you mean? |
| 19:49.38 | ManxPower-work | rigid: I mean you set a domain where a host is needed. you set a host where an IP is needed. |
| 19:50.16 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 19:50.25 | rigid | ManxPower-work: no i didn't :) i used externip = domain (which is legal) in the beginning and later externhost... problem was that it didn't point to the public ip |
| 19:50.38 | rigid | and i really love asterisk but debugging it is a pain :( |
| 19:50.39 | [TK]D-Fender | externip = domain (which is legal) <- wrong |
| 19:51.04 | Qwell | [TK]D-Fender: should work, as long as the IP never changes |
| 19:51.09 | rigid | [TK]D-Fender: really? "For versions of Asterisk pre v1.2, the hostname associated with externip is only looked up when Asterisk startups up or if a 'sip reload' is executed.2 |
| 19:51.10 | rigid | " |
| 19:51.29 | ManxPower-work | rigid: That is NOT the official Asterisk documentation. That is the rantings of one random user. |
| 19:51.37 | [TK]D-Fender | Qwell: Shouldn't. All the real docs say IP only, and externhost is made for that. the fact that externrefresh can update it is superfluous |
| 19:51.56 | rigid | ManxPower-work: it seems pretty like someone reading Changelogs :) |
| 19:52.20 | ManxPower-work | So there is either a documentation error or the user that posted that is an idiot. |
| 19:52.27 | ManxPower-work | I know what of those two I'd bet on. |
| 19:52.37 | rigid | [TK]D-Fender: dunno, we'd had to check the code :) but normally hostnames are resolved by the app if it expects an ip... *shrug* |
| 19:52.54 | ManxPower-work | rigid: not in Asterisk. |
| 19:53.04 | diatonic_ | I've got a strange issue I'm trying to figure out, and hoping someone can help. If I call to a bad phone number over a T1 (PRI) from my Polycom IP 550, I hear a message about the number not being in service. If I call the same number from an Audiocodes MP-124D (sip) I hear a split second of that message, then ringing. |
| 19:53.15 | rigid | i guess the real problem is that I found my config-fault by accident, not due to debugging/logging |
| 19:53.29 | diatonic_ | I've done an SIP debug of each peer to see if I could see an obvious reason, but I'm unsure what I'm looking at |
| 19:53.57 | ManxPower-work | rigid: I do agree that if you put a host in externip Asterisk should tell you you're an idiot |
| 19:54.08 | diatonic_ | If I send the call out on a line on an E&M Wink T1, I get the desired behavior (hearing the message) on the Audiocodes endpoint |
| 19:54.24 | ManxPower-work | BTW, does anyone know if I_AM_NOT_AN_IDIOT is still used in Asterisk. |
| 19:54.36 | outtolunc | can't you just feel the holiday cheer! <G> |
| 19:54.37 | ManxPower-work | diatonic_: sounds like you are using inband messages |
| 19:54.47 | diatonic_ | I have 'Early Media' enabled on the Audiocodes device, which I think is supposed to allow what I'm looking for |
| 19:55.03 | ManxPower-work | diatonic_: um, the telco does not normally play those mesages on PRIs |
| 19:55.09 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:55.29 | ManxPower-work | On a PRI the telco normally sends you back a cause code and hangs up the call. |
| 19:55.30 | diatonic_ | ManxPower-work, Our PRI is 'switched local' |
| 19:55.43 | ManxPower-work | diatonic_: I don't care it's it's switched intergallatic. |
| 19:55.45 | outtolunc | unless the far end isn't digital |
| 19:55.55 | ManxPower-work | outtolunc: I don't see how that matters |
| 19:55.57 | diatonic_ | ManxPower-work, If that means anything - sorry, telco is not my strong suit |
| 19:56.23 | ManxPower-work | diatonic_: look at the pri indication options in chan_dahdi.conf and turn them all off |
| 19:56.40 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 19:56.47 | outtolunc | if the far end isn't digital it switches to inband.. and the sounds <G> |
| 19:56.50 | outtolunc | anyways |
| 19:57.05 | ManxPower-work | outtolunc: maybe in your part of the world |
| 19:57.12 | jblack | has anybody heard anything about a verizon wireless outage? |
| 19:57.52 | diatonic_ | outtolunc, thanks |
| 19:58.28 | diatonic_ | ManxPower-work, I'll look in the chan_dahdi.conf sample for those options, my chan_dahdi.conf doesn't have any pri indication options set |
| 19:58.30 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 19:58.36 | Geminizer | does Asterisk have native support for receiving SMS messages? does it require an additional package? |
| 19:59.05 | ManxPower-work | Geminizer: didn't you ask this yesterday? |
| 19:59.06 | Qwell | Geminizer: how would you be getting SMS messages into Asterisk? |
| 19:59.40 | Geminizer | ManxPower-work, yes... but I got disconnected and missed what followed the question |
| 19:59.56 | ManxPower-work | Geminizer: Asterisk supports SMS over PSTN using FSK encoding. SMS in the United States does not use this and Astersk's SMS support will not work in the USA |
| 20:00.28 | [TK]D-Fender | Geminizer: * is not an SMS platform. |
| 20:00.32 | tzafrir | ManxPower-work, it was removed before 1.2 |
| 20:00.41 | diatonic_ | ManxPower-work, outtolunc What I'm trying to achieve is for the peer at the Audiocodes gateway to be able to hear any audio that is on the dahdi pri channel, instead of 'ringing' |
| 20:01.25 | ManxPower-work | tzafrir: Huh? |
| 20:02.05 | ManxPower-work | tzafrir: I just verified app_sms.so exists in 1.4 |
| 20:03.06 | ManxPower-work | diatonic_: why not just use POTS then? |
| 20:03.18 | ManxPower-work | I mean out of band signalling is one of the major reasons to use a PRI |
| 20:03.29 | tzafrir | <ManxPower-work> BTW, does anyone know if I_AM_NOT_AN_IDIOT is still used in Asterisk. |
| 20:03.39 | ManxPower-work | tzafrir: Oh! Thanks for the response. |
| 20:05.14 | diatonic_ | ManxPower-work - Call center environment. Agents are being monitored using ChanSpy and disposition calls based on varoius criteria. ChanSpy'ing you hear the message about the number not in service - Agent only hears ringing, and thinks the number is good. |
| 20:07.06 | ManxPower-work | diatonic_: why not handle that sort of thing like every other Asterisk user with a PRI on the planet? |
| 20:07.25 | voipmonk | gets the popcorn |
| 20:07.36 | rigid | ManxPower-work: not accepting FQDNs instead of IPs is so non-standard |
| 20:07.38 | diatonic_ | ManxPower-work, I don't know how every other Asterisk user with a PRI on the planet handles it |
| 20:07.42 | *** join/#asterisk eppigy (n=Dave@snugglenets.com) |
| 20:08.20 | ManxPower-work | diatonic_: Dial, check the result of HANGUPCAUSE, play a message using Playback depending on the value of HANGUPCAUSE. The HANGUPCAUSE codes are standard Q.931 codes. |
| 20:08.36 | [TK]D-Fender | rigid: * has specific fields for each, please just deal with it... |
| 20:08.51 | rigid | ManxPower-work: asterisk even notices local IPs that must be public at other places, but not there... it just silently stuffs it into the packet... so the user definetly isn't the idiot... shit happens |
| 20:09.23 | p3nguin | If you misconfigure it, it's not Asterisk's fault. |
| 20:09.38 | rigid | p3nguin: of course not... |
| 20:09.52 | p3nguin | If your complaint is that Asterisk didn't "catch" it, file a bug about it. |
| 20:10.17 | rigid | p3nguin: asterisk catches it at other places... it should be consistent |
| 20:10.27 | p3nguin | ~bugs |
| 20:10.31 | ManxPower-work | diatonic_: Here's a very, very old (and very, very ugly) Asterisk 1.2.x script that handles PRI result codes http://www.fnords.org/~eric/macro-dial-result-v2.inc |
| 20:10.33 | rigid | why don't you just admit that it's a @todo :) |
| 20:10.41 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:11.45 | ManxPower-work | yes, why don't you files a bug at issues.digium.com, rigid? |
| 20:12.04 | diatonic_ | ManxPower-work, I think on the vast majority of our calls, HANGUPCAUSE is used... but I believe that in rural areas and such (non-digital far ends) we are seeing this - guessing. Like I said, I'm not a telco guru |
| 20:12.43 | ManxPower-work | diatonic_: Maybe when a call terminates in a non-SS7 CO. |
| 20:12.54 | ManxPower-work | diatonic_: you're running a call center. become a telecom guru |
| 20:13.23 | ManxPower-work | Makes about as much sense as the owner of an auto racing team not knowing anything about cars. |
| 20:14.05 | diatonic_ | ManxPower-work, Working on it - this channel is helping |
| 20:14.44 | ManxPower-work | diatonic_: Sign up for the Asterisk-Users mailing list at lists.digium.com |
| 20:14.52 | rigid | ManxPower-work: yea it's on my todo-list |
| 20:15.16 | diatonic_ | ManxPower-work, I didn't say I don't know anything about telco, but thanks. |
| 20:15.24 | p3nguin | manxpower-work: The owner often doesn't know much -- he has a crew for that type of thing. |
| 20:16.05 | ManxPower-work | rigid: I'm pretty sure externip/externhost were deliberately separate options since Asterisk's DNS support sucks more than Paris Hilton. |
| 20:16.12 | ManxPower-work | Hmm..that sounded better in my head. |
| 20:16.35 | diatonic_ | ManxPower-work, We have a telecom company that handles a lot of out telco needs, but sometimes I don't know how to describe issues we're seeing in Asterisk |
| 20:16.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:16.43 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-eqcehnftiedivynz) |
| 20:17.57 | voipmonk | snatches diatonic_ and puts him in a time machine... back to a time before yahoo, google, altavista, Qlink and compuserve.... |
| 20:18.34 | voipmonk | come back when you've read telecom for dummies :) |
| 20:19.16 | GuillermoHurtado | is there an IRC for dealing T.38 issues? |
| 20:19.42 | ajp01 | i had quick question about using originate in the asterisk command line, if somebody has a second |
| 20:20.22 | [TK]D-Fender | ajp01: Just ask |
| 20:21.13 | diatonic | thanks voipmonk, we only started using PRI about 6 months ago, and we don't have any issues on the 6 e&m wink DLD T1s we're using. I thought maybe this would be a good place to ask for some help. I guess I was wrong. |
| 20:21.35 | ajp01 | okay, using originate i type the following originate SIP/2913 application Playback hello-world |
| 20:21.38 | ajp01 | this works |
| 20:21.43 | ajp01 | and it calls my extension |
| 20:21.56 | ajp01 | but i can't figure out how to make this work to call my cellphone? |
| 20:22.07 | ajp01 | i tried ZAP/g1/<my number> |
| 20:22.17 | ajp01 | i tried ZAP/g1/91<my number> |
| 20:22.43 | [TK]D-Fender | ajp01: originate SIP/2913 extension 12125551234@from-internal |
| 20:22.43 | ajp01 | and sip/91<mynumber> |
| 20:22.44 | ajp01 | nothing works |
| 20:22.53 | steve745 | do you use a sip gateway or a zap |
| 20:22.59 | ajp01 | sip gateway |
| 20:23.03 | [TK]D-Fender | agIf you want to call that number when your sip phone answers |
| 20:23.21 | [TK]D-Fender | ajp01: What is your FINAL goal? |
| 20:23.52 | steve745 | i use Sip/1234567890@?sipgateway |
| 20:23.52 | ajp01 | final goal is that it calls a user and asks them a question they answer and then hangup, but for me, baby steps |
| 20:24.16 | steve745 | i have a setup to call play recording press 1 transfer to queue |
| 20:24.26 | steve745 | then hits agents in ringgroup |
| 20:24.35 | ajp01 | cool |
| 20:24.56 | steve745 | i'm a rookie and learned it all the hard way |
| 20:25.11 | ajp01 | i'm a rookie, my former boss, 3 bosses ago, installed the system |
| 20:25.13 | ajp01 | i inherited it |
| 20:25.35 | ajp01 | started with freepbx interface and am reverse engineering it to figure things out |
| 20:25.46 | steve745 | i use call files created and uploaded by my custom Desktop Database |
| 20:25.49 | ajp01 | not actually taking it apart but drawing the paralells between the gui and system |
| 20:26.11 | ajp01 | i tried call files but not knowing what i was doing it was a pain to constantly have no direct response to my attempts |
| 20:26.15 | [TK]D-Fender | ajp01: originate Local/12125551234@from-internal extension 12345@acustomcontextyoumake |
| 20:27.02 | [TK]D-Fender | ajp01: Now with that you'll have to make the exten I pointed to in a custom context |
| 20:27.34 | ajp01 | so there has to be a custom extension |
| 20:27.59 | [TK]D-Fender | ajWell you want to do something special like asking a question... which sounds like you need an ANSWER |
| 20:28.02 | ajp01 | the custom context and extension do what |
| 20:28.08 | [TK]D-Fender | ajp01: So yes, or course its "custom |
| 20:28.11 | ajp01 | ah |
| 20:28.13 | ajp01 | i get it |
| 20:28.32 | ManxPower-work | ajp01: just remmber FreePBX uses a totally non-standard dialplan |
| 20:28.49 | [TK]D-Fender | ajp01: originate Local/12125551234@from-internal application Playback hello-world |
| 20:29.11 | [TK]D-Fender | ajp01: That MIGHT work for the ultr minimal test so you se see * doing anything at all after placing the out-call |
| 20:30.15 | *** join/#asterisk HaMF (n=hannes@nat-wh.rz.uni-karlsruhe.de) |
| 20:31.24 | ajp01 | still trying it too much congestion going on for me to pick out the result |
| 20:32.15 | ajp01 | it says congestion in new stack...does that mean anything |
| 20:32.23 | [TK]D-Fender | ~pb |
| 20:32.24 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 20:32.25 | [TK]D-Fender | ^^^^^^^^ |
| 20:32.34 | ajp01 | there's no ring but it plays you're call cannot be completed as dialed |
| 20:32.40 | ajp01 | in the window |
| 20:32.45 | [TK]D-Fender | ajp01: Everything from your issuing the command through the end |
| 20:32.58 | ajp01 | one sec |
| 20:33.43 | hardwire | other than the pika warp.. does anybody know of a small unit that supports fxo/fxs and has dual ethernet of about the same size? |
| 20:33.52 | hardwire | I'd love to use the fit-pc2 but it lacks fx* |
| 20:33.57 | hardwire | fit-pc2i |
| 20:33.58 | hardwire | heh |
| 20:34.20 | voipmonk | atom proc + digium cards and a dual nic |
| 20:34.30 | ajp01 | OMFG! YOU ROCK (and i'm a retard) |
| 20:34.32 | voipmonk | try an asus ion |
| 20:34.40 | ManxPower-work | I've used a PIKA before. I'm STILL in therapy |
| 20:34.45 | voipmonk | PREACH, ajp01 |
| 20:34.52 | ajp01 | i copied and pasted the command but forgot the change the phone number in my paste |
| 20:34.57 | ajp01 | lol |
| 20:35.04 | ajp01 | bows, humbled. |
| 20:35.09 | hardwire | voipmonk: the asus has no expansion and single ethernet |
| 20:35.26 | [TK]D-Fender | ajp01: You failed the "555? OMG I'm in a movie!" tes |
| 20:35.28 | [TK]D-Fender | tes* |
| 20:35.30 | [TK]D-Fender | test |
| 20:35.45 | hardwire | 555-867-5309 |
| 20:35.49 | hardwire | I'm in a movie AND a snog |
| 20:35.50 | hardwire | song |
| 20:36.07 | [TK]D-Fender | hardwire: And a snog :) |
| 20:36.25 | hardwire | [TK]D-Fender: no snog for you, sir. |
| 20:36.25 | HaMF | does anyone know if it's "normal" that memtest86+ returns an mem-error if you set "memory sizing" to "probe" or "bios all"? |
| 20:36.30 | [TK]D-Fender | hardwire: :( |
| 20:36.56 | voipmonk | hardwire: try zotac? |
| 20:37.01 | hardwire | is that a drug? |
| 20:37.05 | hardwire | do I need it? |
| 20:37.06 | ManxPower-work | HaMF: that sounds like a question for a different channel |
| 20:37.08 | hardwire | are you judging me? |
| 20:37.26 | [TK]D-Fender | gavels |
| 20:37.45 | hardwire | voipmonk: I'm actually pretty familiar with the ITX world |
| 20:38.17 | hardwire | what I don't want to do right now is throw a pci card in a system and call it a value added feature to some other services... since they can just take he PCI card out and ruin everything :) |
| 20:38.28 | hardwire | the WARP would work well for OEM |
| 20:38.32 | hardwire | however.. meh |
| 20:38.32 | *** join/#asterisk dkirker-openmobl (n=dkirker@pcp065241pcs.wireless.calpoly.edu) |
| 20:38.44 | HaMF | ManxPower-work, you're right. But I did this memtest becaus asterisk freezes randomly and it might be that someone by chance knows the answer,. |
| 20:38.52 | hardwire | tries to find a small footprint ITX enclosure with a full PCI slot |
| 20:39.01 | hardwire | or maybe just use the sangoma USB fxs |
| 20:40.34 | voipmonk | try www.logicsupply.com |
| 20:40.44 | coppice | most ITX stuff seems to get disappointingly hot |
| 20:40.53 | hardwire | voipmonk: right, that's my usual spot |
| 20:42.40 | hardwire | I've been looking at the aaeon stuff for a while too |
| 20:42.42 | hardwire | but I can't afford them |
| 20:42.44 | hardwire | http://www.aaeon.com/PD_Products_Detail_FA0E4D23045E4E86B2_54F70EFFE6394F8CBD_77632894F2154E62A8_US_UTF-8.html |
| 20:42.48 | hardwire | amazingly cool chassis |
| 20:43.19 | hardwire | http://www.aaeon.com/PL_Products_List_FA0E4D23045E4E86B2_54F70EFFE6394F8CBD_US_UTF-8.html |
| 20:43.32 | hardwire | hmm.. the wall mount one isn't available any more |
| 20:43.38 | hardwire | err rather.. the 4 PCI one |
| 20:45.19 | hardwire | oh.. still there.. AEC-6915 |
| 20:45.21 | *** join/#asterisk simplydrew (n=simplydr@pool-74-97-177-245.prvdri.fios.verizon.net) |
| 20:45.31 | diatonic | I just got a Christmas card & gift from logicsupply.com. Dig those guys. |
| 20:45.49 | LemensTS | i dont need to install dahdi unless im using a card right? |
| 20:45.52 | LemensTS | 1.4 |
| 20:46.49 | [TK]D-Fender | LemensTS: Or want it for other timing requirements |
| 20:46.51 | leifmadsen | LemensTS: dahdi_dummy for timing (no hardware required) for meetme, iax2 trunking, etc... |
| 20:48.13 | LemensTS | Ok thats what i thought it had been a while for me. Thanks |
| 20:50.33 | hardwire | diatonic: aww I didn't! |
| 20:50.58 | hardwire | says we all hijack simplydrews FIOS |
| 20:52.59 | diatonic | hardwire: I'm using this enclosure: http://www.logicsupply.com/products/gs_l05 with an atom/ion board, but it doesn't have a pci slot. There isa low profile slot on the chassis if you know of a digium card that could work in it :) |
| 20:52.59 | rigid | what is the default way to have 2 peers connected by dialing something like 1*, 2*, 3* ... ? |
| 20:53.19 | bcrisp | ? |
| 20:53.38 | hardwire | diatonic: low profile + sangoma |
| 20:53.39 | bcrisp | rigid: im not sure what you mean |
| 20:54.04 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:54.11 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:54.23 | bcrisp | rigid: could you illustrate with an example? |
| 20:54.41 | SuPrSluG | anyone using ata to connect to security systems |
| 20:55.24 | rigid | bcrisp: i have three registered peers, one sip-provider and 2 friends who can receive/place call over the provider. I want the 2 friends to call each other directly using an internal number |
| 20:55.34 | rigid | calls |
| 20:55.44 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 20:56.13 | bcrisp | rigid: 2 sip peers to talk to each other? see extensions.conf |
| 20:56.17 | rigid | pretty basic, but I see there are various ways to do it |
| 20:56.31 | bcrisp | set up a context and extensions |
| 20:58.18 | [TK]D-Fender | rigid: Go make extens that each can dial to Dial() the other |
| 20:58.26 | [TK]D-Fender | rigid: time to sit down with The BOOK |
| 20:58.28 | [TK]D-Fender | ~book |
| 20:58.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:59.37 | nny | stupid question, is there an eloquent way for the DID dialed to show up in the CDR? |
| 21:00.13 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 21:00.21 | a1fa | can asterisk listen on multiple ports? |
| 21:00.50 | nny | looks at asterisk account codes |
| 21:01.18 | rigid | looks ashamed @ [TK]D-Fender |
| 21:01.27 | citywok | nny, on the inbound set CDR(DID)=${EXTEN} & then use custom CDR output |
| 21:01.34 | *** part/#asterisk mold (i=gtgt@firewalled.with.ipfw.us) |
| 21:01.40 | a1fa | (sip) |
| 21:01.41 | diatonic | a1fa, You could just use iptables to simulate listening on multimple ports I'd imagine |
| 21:01.43 | citywok | (if you dont already get it somewhere else) |
| 21:01.47 | *** join/#asterisk TheDavidFactor-H (n=thedavid@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 21:01.48 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 21:01.48 | a1fa | ye.. i don't want to do that |
| 21:01.51 | nny | citywok: ok thanks |
| 21:02.16 | jblack | what is with blackberries? They seem unable to work for longer than a month |
| 21:02.28 | bcrisp | my iphone hates blackberries |
| 21:02.51 | citywok | my droid hates iphoines |
| 21:02.58 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 21:03.02 | citywok | no more dropped calls ftw |
| 21:03.13 | tzafrir | a1fa, 4569 and 5060 |
| 21:03.23 | bcrisp | ya iphone going with ATT was a mistake ... going after world market i guess |
| 21:03.45 | *** join/#asterisk johnyjj2 (n=mainacco@p9g117.traco.pl) |
| 21:03.50 | johnyjj2 | hello :) |
| 21:03.50 | ManxPower-work | a1fa: iptables will be easier than rewriting part of Asterisk. |
| 21:04.34 | tzafrir | ManxPower-work, isn't the SIP port number included somewhere in the protocol? |
| 21:04.45 | tzafrir | Or is it just the RTP? |
| 21:04.46 | a1fa | :( |
| 21:05.18 | *** join/#asterisk btsteve (n=tstevens@24-196-234-39.dhcp.stpt.wi.charter.com) |
| 21:05.43 | ManxPower-work | tzafrir: I'm sure both specify ports as part of the packet |
| 21:06.00 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 21:10.10 | *** join/#asterisk jozza (n=chatzill@BSN-77-186-224.dsl.siol.net) |
| 21:11.42 | *** part/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 21:12.08 | jozza | hi everyone, how do i dial a device and after, let say 10 sec. dial an additional device, while keep ringing the first device? |
| 21:12.17 | nny | hmm anyone seen any useful asterisk + CMS implementations? |
| 21:12.33 | citywok | for... |
| 21:12.59 | [TK]D-Fender | jozza: dial the initial device and a local channel exten that Wait() 's first then dials the 2nd |
| 21:13.12 | nny | Mostly unique number data management. The ability to track new customers, which number the called etc |
| 21:13.40 | a1fa | shurgs |
| 21:13.48 | citywok | ah, we built all of that in to our in house application |
| 21:13.48 | a1fa | i guess i went ahead and used iptables |
| 21:14.00 | jozza | hmm, good, sounds like a plan |
| 21:14.06 | a1fa | really didnt want to do that.. but oh well.. |
| 21:14.17 | a1fa | hi [TK]D-Fender |
| 21:14.20 | citywok | if it works, why not do it? |
| 21:14.33 | citywok | seems like a fairly simple 1 line solution to me |
| 21:14.39 | a1fa | because it does not scale |
| 21:14.53 | citywok | iptables can handle a lot of traffic without issue |
| 21:15.06 | a1fa | that's not what i worry about |
| 21:15.11 | jozza | thx fender, i'll try that |
| 21:15.30 | a1fa | has anyone implemented zoneminder+asterisk for alerting? |
| 21:16.23 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 21:16.54 | nny | citywok: yeah this would kind of be a stand alone solution, or rather an implementation of asterisk + something else or asterisk +php/mysql hackery. I'll delve further |
| 21:17.37 | nny | actually Asterisk + Outlook/Outcall and Salesforce would be a nasty but effective combo |
| 21:17.56 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 21:19.24 | *** join/#asterisk ticoit (n=ticoit@201.191.187.35) |
| 21:19.58 | citywok | php/mysql hackery != CMS lol |
| 21:20.11 | nny | citywok: lol i know |
| 21:20.12 | citywok | it really just depends how you plan on managing your lists :) |
| 21:20.16 | nny | citywok: exactly |
| 21:20.28 | citywok | if you need help with call center stuff, we can help you out :) |
| 21:20.37 | citywok | we've got a lot of software developed just for this purpose |
| 21:20.38 | nny | although asterisk + outlook would be able to interface with quite a bit of cms implementations |
| 21:20.58 | citywok | you must not be managing thousands of contacts |
| 21:21.07 | citywok | cuz outlook + thousands of contacts would be... hilarity |
| 21:22.05 | citywok | especially if you wanted to do any kind of reporting |
| 21:22.08 | nny | yeah not sure how outlook + salesforce handles that right now |
| 21:22.23 | nny | but I know salesforce is a pretty popularCMS |
| 21:22.28 | p3nguin | jozza: You could also do a Dial(tech/first_device,10) followed by a Dial(tech/first_device&tech/second_device,50) |
| 21:22.55 | citywok | p3nguin: ya that would work, but that's uglier than the previously suggested solution since it terminates the first call and re-dials |
| 21:23.02 | p3nguin | jozza: So the first one would ring for 10 second and then timeout; then both would ring for an additional 50 seconds before timine out. |
| 21:23.03 | jozza | no, that would make a missed call on the first device |
| 21:23.05 | citywok | much of the 10 seconds can be spent while the cell carrier just locates the phone |
| 21:23.47 | ManxPower-work | jozza: If you have a min I can explain one other way to do it. |
| 21:24.19 | jozza | sure |
| 21:25.55 | ManxPower-work | Set up two extensions: exten => 1111,1,Dial(SIP/1234) exten => 4444,1,Wait(10) exten => 2222,n,Dial(SIP/4321) and your real extension does Dial(Local/1111@whatevercontext&Dial(Local/2222@whatevercontext) |
| 21:26.35 | ManxPower-work | jozza: see localchannel.tex in the doc/ directory of the Asterisk source. |
| 21:26.53 | ManxPower-work | let me pastebin that and fix my errors |
| 21:26.57 | jozza | i agree, its the best solution |
| 21:27.08 | jozza | most suitable |
| 21:27.59 | [TK]D-Fender | jozza: Same as mine except he nested the ones that were to ring immediately within local channels as well. this adds lod for no reason and wastes chars on your line |
| 21:28.13 | LemensTS | what can this be: |
| 21:28.13 | LemensTS | [Dec 17 15:27:25] WARNING[8230]: chan_sip.c:8711 check_auth: username mismatch, have <2294607234>, digest has <namecom> |
| 21:28.22 | jozza | i agree |
| 21:28.59 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 21:29.11 | ManxPower-work | http://pastebin.ca/1718725 Extensions don't have to be numbers. I prefixed the extensions with L to prevent them from being dialed from a phone |
| 21:30.05 | [TK]D-Fender | checkout time, BBIAb |
| 21:30.20 | jozza | i want to make the first device ring and then after 10 sec., dial the other one, if the frst one did not answer,otherwise cancel the second one |
| 21:30.57 | ManxPower-work | jozza: My example should do that |
| 21:31.09 | ManxPower-work | But you won't get an extra missed call using my way |
| 21:31.43 | p3nguin | manxpower-work: Why wouldn't this work just as well? http://pastebin.ca/1718729 |
| 21:32.32 | ManxPower-work | p3nguin: Yes that would also work. My example was trying to be instructive rather than the shortest way |
| 21:32.47 | p3nguin | I see. I just wanted to make sure I understood it correctly. |
| 21:32.53 | ManxPower-work | p3nguin: you did |
| 21:33.53 | p3nguin | I've tried to find places to use Local channels, but I never seem to come up with anything useful on my system. |
| 21:34.07 | rigid | excellent, everything works just fine... |
| 21:34.27 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 21:34.33 | rigid | now I need an answering-machine extension with email+mp3 attachment :) |
| 21:34.34 | ManxPower-work | p3nguin: Think "chan_local" any time you think "Gee, I wish this dialplan application could dial extensions instead of just a device" |
| 21:34.49 | *** join/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
| 21:34.54 | bcrisp | i love local channels |
| 21:35.15 | p3nguin | manxpower-work: I'll try to keep that concept in mind. |
| 21:35.25 | HubguruJR | hello |
| 21:35.43 | p3nguin | Seems like an excellent way to think of it. |
| 21:36.10 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:36.24 | jozza | how would L1235 not ring if 1234 did answer? |
| 21:37.06 | HubguruJR | lab testing SIP with 1.6.1.12-rc1, running into call limit of 100 calls, 200 active channels, then asterisk crashes with too many open files, have limits set to 65535, any ideas? |
| 21:37.09 | p3nguin | If only my fax would work correctly, I wouldn't have anything not working. |
| 21:37.33 | HubguruJR | running debian etch |
| 21:39.37 | jozza | by youe pastebin example,i should check if the first device answered before dialing the second one, right? |
| 21:42.02 | ManxPower-work | joako: no, the 2nd one will never dial if the first is answered |
| 21:42.12 | ManxPower-work | (well answered withing the first 10 seconds) |
| 21:42.44 | jozza | really? why |
| 21:42.54 | p3nguin | That's what I'm wondering. |
| 21:43.00 | jozza | ah yes |
| 21:43.02 | jozza | <PROTECTED> |
| 21:43.06 | jozza | ufcouse |
| 21:43.10 | p3nguin | It looks like it rings one and starts waiting on the second right away. |
| 21:43.12 | jozza | makes sense |
| 21:43.24 | jozza | silly me |
| 21:44.02 | p3nguin | Even if the first answers, why won't the Wait() continue? |
| 21:44.23 | p3nguin | The Wait() would continue and then it would ring the other device once it finishes waiting. |
| 21:44.36 | jozza | probably it will get canceled because of the first one answered |
| 21:44.48 | jozza | no? |
| 21:44.49 | p3nguin | How would it be cancelled, though? |
| 21:45.06 | p3nguin | One begins to ring at the same time one begins to wait. |
| 21:45.25 | p3nguin | The extension has already been executed, so what would cancel it? |
| 21:45.25 | jozza | probably the same way as the real second device would on a real call |
| 21:45.47 | p3nguin | That's different, though. |
| 21:45.52 | jozza | dial does that |
| 21:46.06 | p3nguin | Dial() also doesn't execute another exten. |
| 21:46.25 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:47.00 | jozza | local is real device like simulation |
| 21:47.16 | citywok | local channels do have a few problemsi 've noticed especially around queues |
| 21:47.16 | p3nguin | what? |
| 21:48.57 | jozza | who does your what go to? |
| 21:48.59 | p3nguin | I'm just saying that when Dial() dials both of the local channels, those local channels call extens. Those extens are going to execute at the same time. |
| 21:49.36 | jozza | hmm |
| 21:49.48 | jozza | a test would show a result |
| 21:49.48 | p3nguin | If those two completely mutually exclusive extens are executed at the same time, what makes one magically die off just because there was an answer on the other? |
| 21:50.23 | jozza | well, on a real call, after the first device answers, the second one stops ringing |
| 21:50.33 | p3nguin | Only in a Dial() |
| 21:50.43 | p3nguin | Dial() usually is dialing DEVICES. |
| 21:50.47 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:51.01 | p3nguin | In this case, Dial() is executing two mutually exlusive extensions. |
| 21:51.27 | p3nguin | Once you run it, you can't unrun it. |
| 21:51.45 | jozza | probably the example with one real device and one local channel would work |
| 21:51.49 | jozza | but i cant say |
| 21:51.59 | [TK]D-Fender | did |
| 21:52.14 | jozza | did? |
| 21:52.19 | Kobaz | p3nguin: sure you can... just like you can become unpregnant |
| 21:52.25 | jozza | lol |
| 21:52.30 | Kobaz | p3nguin: soft hangup |
| 21:52.33 | [TK]D-Fender | jozza: from the first answer I gave you |
| 21:52.37 | p3nguin | I think the same thing will happen both ways. Think of it like both being phones getting answered at the exact same microsecond. |
| 21:52.44 | citywok | [TK]D-Fender: when you use a local channel for a queue member, ringinuse does not work properly. any ideas? |
| 21:52.48 | [TK]D-Fender | jozza: dial on SIp device immediately, and 1 delayed Local channel |
| 21:52.59 | jozza | great |
| 21:53.05 | [TK]D-Fender | citywok: Loot harder |
| 21:53.09 | [TK]D-Fender | look* |
| 21:53.13 | jozza | you saved me a testing |
| 21:53.13 | p3nguin | Except in this case, that phone is waiting 10 seconds before ringing. |
| 21:53.59 | Kobaz | [TK]D-Fender: wouldn't ringunuse not apply to local channels? |
| 21:54.00 | p3nguin | I'll have to test it, because I still feel that even if the first phone is answered, the second will still ring because nothing stops the other exten from executing. |
| 21:54.11 | jozza | gotta run, later...my gf is here :) |
| 21:55.09 | jozza | i'll come back to check on this later in case it doesnt work |
| 21:55.19 | jozza | bye |
| 21:58.19 | p3nguin | Okay, I tested it. |
| 21:58.56 | p3nguin | I don't know why, but when the first one picked up, the second one exited with a non-zero code. |
| 21:59.09 | p3nguin | To me, this does not follow logic. |
| 22:03.15 | *** join/#asterisk joobie (n=joobz@mx01.anric.com.au) |
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| 22:05.40 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 22:07.11 | *** part/#asterisk machoman48 (n=machoman@89.203.164.69) |
| 22:09.17 | joobie | guys i want to structure the dialplan so that if a user dials '321', it goes to VoiceMailMain(<insert src internal extension number>@default) .. not sure what variable to use to get the src extension 321 is being dialed from.. |
| 22:09.36 | joobie | i have all the office phones connected to their own sip account on the * box.. but against each sip account, i also define the callerid=var.. and set it to their external number.. so when they dial to people outside of the office, it displays their external number on the remote phones |
| 22:09.37 | Qwell | ${EXTEN} ... |
| 22:09.48 | Qwell | oh wait |
| 22:09.49 | joobie | cant use callerid because of this.. |
| 22:10.41 | joobie | Qwell, similar to EXTEN but rather getting the src of the sip phone |
| 22:11.10 | joobie | like the voicemail's are for example 1000@default, 1001@default, etc.. so if the phone with extension 1000 dials 321, i want to grab teh 1000 part.. and cant use caller id because something the callerid num is set to an extenral number which is different from the internal extensino |
| 22:11.17 | ChannelZ | we went over this last night |
| 22:11.25 | joobie | nod, but didnt get a solution |
| 22:11.34 | ChannelZ | Well not one you liked |
| 22:12.05 | ChannelZ | An extension is an arbitrary thing in the dialplan |
| 22:12.51 | ChannelZ | In order to automatically pass a mailbox number to voicemail, you have to have that information come from the *device* |
| 22:13.16 | *** join/#asterisk ayrjola (n=ayrjola@a88-115-162-170.elisa-laajakaista.fi) |
| 22:13.34 | ChannelZ | Typically that's with the Caller ID number being the extension, not a DID number. In your case you're going to have to write a bunch of dialplan script to convert the DID number into what their voicemail box number is |
| 22:13.42 | *** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26) |
| 22:13.49 | [TK]D-Fender | joobie: use the channel name, or set a var from the peer definition |
| 22:14.05 | jblack | joobie: Make the extensions and the voicemail boxes the same numbers. :) |
| 22:14.14 | ChannelZ | You might be able to cheat and use the channel name if they are named as your extensions (as TK just said) |
| 22:14.44 | ChannelZ | TK: That was my question, I wasn't sure how to define arbitrary variables from like sip.conf? |
| 22:15.30 | ChannelZ | oh, setvar!! DUHHH |
| 22:15.39 | ChannelZ | I've read right over that 100 times |
| 22:18.05 | ChannelZ | so joobie, in your sip.conf (for instance) you could do "setvar=VMBOX=1000" etc for each device, and then use ${VMBOX} in your dialplan for the mailbox number when you call VoiceMailMain |
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| 22:19.06 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 22:20.12 | paulc | I love it when a plan comes together :) |
| 22:22.02 | *** join/#asterisk spiky-{roam}- (n=spiky-{r@188-221-85-47.zone12.bethere.co.uk) |
| 22:23.09 | ChannelZ | heh |
| 22:23.17 | ChannelZ | always learning something new |
| 22:23.27 | ariel_ | I think setvar is now just set |
| 22:24.19 | ChannelZ | hmm |
| 22:25.06 | ChannelZ | setvar works in 1.4, at least on the sample configs in 1.6 it still says setvar |
| 22:25.43 | paulc | Revisiting ExternalIVR - seems its grown up a fair bit in 1.6, which is nice |
| 22:25.50 | paulc | Anyone doing anything with it? |
| 22:25.59 | paulc | s/anything/anything exciting/ |
| 22:26.14 | paulc | thanks infobot |
| 22:26.27 | ChannelZ | paulc: you're thinking of dialplan applications |
| 22:26.41 | ariel_ | Set works in 1.4 and in 1.6, setvar did not work in my test I just made |
| 22:27.01 | ChannelZ | err sorry I meant ariel_ |
| 22:27.47 | ChannelZ | ariel_: setvar in sip.conf, not an app in extensions.conf |
| 22:27.55 | ariel_ | OH ok |
| 22:29.06 | joobie | back sry phone.. |
| 22:29.24 | ChannelZ | joobie: I assume these are SIP channels ? |
| 22:29.40 | joobie | nod |
| 22:30.12 | ChannelZ | good then that will work |
| 22:32.13 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 22:32.31 | joobie | sweet |
| 22:32.34 | joobie | i might use setvar |
| 22:32.48 | joobie | will that work if i define it in the sip.conf and carry through to the dialplan? |
| 22:32.56 | ChannelZ | yeah that's the point |
| 22:33.00 | joobie | sweet |
| 22:33.04 | joobie | i was googling about that yesterday |
| 22:33.15 | joobie | couldnt find anything about it |
| 22:33.20 | joobie | but that's perfect - cheers |
| 22:33.21 | ChannelZ | setvar=BLAH=400 will show up as ${BLAH} when that channel dials something |
| 22:35.27 | *** part/#asterisk HubguruJR (n=jrichard@mail.ntegratedsolutions.com) |
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| 22:41.22 | Kobaz | grandstream phones are so tehsux |
| 22:41.50 | Kobaz | if you're idle for 5 minutes, it will hang up on you |
| 22:42.12 | ChannelZ | nice |
| 22:45.58 | *** part/#asterisk ayrjola (n=ayrjola@a88-115-162-170.elisa-laajakaista.fi) |
| 22:47.37 | nny | so outcall is dead more or less, as always. there is another outlook addon that does that same thing (or similar) but the name escapes me. anyone know it? |
| 22:47.58 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 22:49.54 | nny | dammit ha |
| 22:49.57 | nny | CRM ! CMS |
| 22:50.02 | nny | long day |
| 22:50.09 | nny | found a list, perusing now |
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| 22:52.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:56.33 | bcrisp | PBS? |
| 23:03.15 | *** join/#asterisk s0lid (n=s0lid@124.104.45.64) |
| 23:12.23 | *** join/#asterisk dkirker-openmobl (n=dkirker@pcp065241pcs.wireless.calpoly.edu) |
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| 23:22.25 | Kobaz | i need some help with ast_streamfile()... the devs in #asterisk-dev are sleeping |
| 23:23.14 | vally | basically i have an ast_streamfile problem as well :D |
| 23:23.19 | Kobaz | hah |
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| 23:28.34 | *** join/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 23:30.02 | bcrisp | and im back |
| 23:30.40 | bcrisp | anyone else use their headset as ear warmers? |
| 23:31.47 | johnyjj2 | is there anybody here willing to answer some simple questions from newbie about automatic speech recognition with asterisk :) ? if yes, please visit my thread: http://lists.digium.com/pipermail/asterisk-speech-rec/2009-December/000448.html |
| 23:37.34 | voipmonk | hehe |
| 23:37.36 | bcrisp | im trying to play with call parking, transfers etc ... |
| 23:37.58 | bcrisp | i setup features.conf and included the parkedcalls (default context) in extensions.conf |
| 23:38.05 | bcrisp | not seeing any of the functionality happen |
| 23:38.06 | *** join/#asterisk blkry (n=chatzill@96.37.24.97) |
| 23:38.19 | bcrisp | (v1.6.1.11) |
| 23:40.45 | bcrisp | nm... don't think my Dial() is setup properly |
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| 23:45.18 | Deeewayne | anyone know if local number portability is for wireless only or if it includes landline numbers ? |
| 23:45.25 | doolittlework | hi there anyone ever installed a atcomX1600P fxo card |
| 23:46.01 | doolittlework | I read that it uses wctdm.c witch is customized |
| 23:46.05 | etfonhomey1 | Deeewayne, I know a guy who ported his land line to a cell. |
| 23:46.07 | *** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202) |
| 23:46.21 | etfonhomey1 | and got rid of his landline. |
| 23:46.46 | Deeewayne | etfonhomey1, thanks |
| 23:46.55 | doolittlework | if i reload dahdi do i need to reload asterisk as well? |
| 23:48.44 | loather-work | if you made a hardware change you'll likely have to *restart* asterisk |
| 23:49.25 | Kobaz | doolittlework: if you have 1.6, you can do a module unload chan_dahdi |
| 23:49.30 | Kobaz | and it will keep everything else going |
| 23:50.00 | doolittlework | atcom says i have to recompile dahdi and dahdi tools, and change the wctdm.c file with theirs,,,???? do i need install asterisk again? |
| 23:50.15 | Kobaz | 1.4 was restart happy for major changes... 1.6 you can change just about everything without a restart |
| 23:50.36 | Kobaz | doolittlework: no, asterisk has it's own dahdi module that communicates with the kernel modules |
| 23:51.35 | etfonhomey1 | Deeewayne, it was an AT&T/Bellsouth landline to an AT&T Wireless phone, FYI. |
| 23:51.36 | Kobaz | if you're rebuilding dahdi, you'll have to unload the current dahdi kernel modules before the new ones will load |
| 23:51.51 | doolittlework | they say i must change the wctdm.c file with theirs before install, where do i find the drivers in dahdi? |
| 23:52.34 | doolittlework | Kobaz: how do unload the current one? |
| 23:52.47 | doolittlework | i |
| 23:52.54 | Kobaz | the dahdi drivers are *in* the dahdi tarball |
| 23:53.01 | Kobaz | rmmod <insert module here> |
| 23:53.24 | doolittlework | ta |
| 23:53.44 | Kobaz | lsmod to list currently running modules |
| 23:54.52 | doolittlework | does all the wcXXX need to unloaded? |
| 23:55.13 | Kobaz | obviously |
| 23:55.17 | doolittlework | ta |
| 23:55.22 | Kobaz | if you're going to be changing it and rebuilding it |
| 23:57.27 | Kobaz | you would think... playing a random track to a channel would be pretty straightforward |
| 23:57.31 | Kobaz | er |
| 23:57.31 | Kobaz | wrong chan |
