| 00:00.08 | cobolfoo | just add memory and hard drive youself, not very hard |
| 00:01.17 | cobolfoo | I usually install ubuntu 10.04 LTS and add asterisk + dahdi (compile newest sources) and setup a FreePBX web interface on it |
| 00:01.44 | DeVL | thanks for that but Im looking for a complete system....im already behind deadline..lol |
| 00:01.55 | cobolfoo | ok you mean a Asterisk Appliance ? |
| 00:01.58 | DeVL | and this is just a test bed so it doesnt have to be of quality |
| 00:02.05 | cobolfoo | http://www.chinaroby.com/ |
| 00:02.17 | cobolfoo | you can have a asterisk appliance for 300$ |
| 00:02.27 | DeVL | no just a PC/workstation to test the latest build |
| 00:02.31 | cobolfoo | hehe |
| 00:02.57 | cobolfoo | Ok, I have problems understanding you, for me adding ram and HD is something that takes maybe 5 minutes to do |
| 00:03.25 | cobolfoo | buy anyway, you can find a bundle which already contains hd and memory on newegg |
| 00:03.42 | DeVL | yea thats not a problem but I dont want to search around for compatible RAM, HD etc...just a complete box that will run asterisk |
| 00:04.48 | *** part/#asterisk DMeloUK (~Dominic_M@157.214.189.72.cfl.res.rr.com) |
| 00:04.53 | DeVL | yea thats all I need is some specs on a good/average proccessor, memory, hard drive and whatever else |
| 00:08.13 | *** part/#asterisk DeVL (~IceChat7@173-17-14-167.client.mchsi.com) |
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| 00:55.38 | jsgoecke | Hello |
| 01:12.53 | ChannelZ | ohell |
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| 02:13.08 | raden | is there a way for asterisk to read a mysql table for Callerid info ? |
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| 02:20.59 | p3nguin | MYSQL() should do it. |
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| 02:58.40 | jameswf | moo |
| 03:15.45 | p3nguin | Moo you say? |
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| 03:29.06 | p3nguin | Anyone have any experience with Extreme Broadband Engineering brand coax splitters? |
| 03:29.09 | p3nguin | 5-1000 MHz / EMI -130 dB |
| 03:29.17 | p3nguin | We usually use Antronix, but I'm ready for a change if these aren't junk. |
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| 03:33.56 | jameswf | I hear they are extreme |
| 03:34.47 | jameswf | best way to find reviews is to google: <product> sucks |
| 03:34.51 | p3nguin | I'm sure they are extreme... I'm only curious if they're extremely good. |
| 03:35.22 | jameswf | marketers dont often use the word sucks |
| 03:36.19 | p3nguin | "BDS102H sucks" and "BDS102H is junk" don't turn up anything. |
| 03:36.46 | p3nguin | wait, maybe one. |
| 03:38.30 | p3nguin | Nope. |
| 03:39.00 | p3nguin | It was a thread saying Monster splitters are junk, and a recommendation to buy the BDS102H. |
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| 03:41.49 | p3nguin | I guess I'll get a handful and see how they work out. If they suck, I won't buy any more. |
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| 04:04.10 | pabelanger | When using Originate over AMI, how do you use exten => t and exten => i ? |
| 04:07.37 | pabelanger | I assume use a local channel |
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| 06:00.37 | schmidts | good morning |
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| 06:53.48 | `paul | if i have 123 => {Noop('test');} and 123 => {Noop('test2');} which one will be followed the first or last in extensions.ael? |
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| 06:57.31 | kaldemar | `paul: are they in the same context? |
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| 07:04.36 | `paul | yes |
| 07:04.40 | `paul | i did a test |
| 07:05.05 | `paul | first one is followed |
| 07:06.50 | kaldemar | why would you have two exactly same extensions in the same context? |
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| 07:08.28 | `paul | kaldemar: i have lot of tollfrees already set up im generating a php script to extract tollfrees from a db and write it to a file and include it on top of extensions.ael :D |
| 07:09.09 | `paul | if ever there are similar tf i want the one from DB to be followed lol |
| 07:09.30 | kaldemar | your dialplan sounds horrible. |
| 07:09.37 | `paul | lol |
| 07:10.37 | `paul | its like this |
| 07:11.15 | `paul | 88888888 => {&answerservice(queue-name);}; <-right now |
| 07:12.01 | `paul | but it should be |
| 07:12.01 | `paul | #include file_generated_byscript.ael |
| 07:12.01 | `paul | 88888888 => {&answerservice(queue-name);}; |
| 07:12.51 | `paul | where file generated's queue name depends on DB |
| 07:13.40 | ChannelZ | why would they cross though |
| 07:13.48 | `paul | but there is a chance that 888888888 could be in the file |
| 07:14.11 | `paul | but i want the one from the DB to be prioritized :) |
| 07:14.27 | ChannelZ | they sort of get merged if memory serves |
| 07:14.41 | ChannelZ | You shouldn't be duplicating extensions in the same context |
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| 07:15.51 | `paul | but eventually all numbers should be coming from DB |
| 07:16.24 | ChannelZ | then pay attention to what you're doing and make sure things are in their right place. |
| 07:16.42 | `paul | so i was just wondering for now, if i have 1234 from the include file and 1234 after the include which one will be followed |
| 07:16.51 | ChannelZ | Why would you bother with a DB anyway if you're going to have some of the same crap in the static dialplan? |
| 07:18.21 | `paul | hmmm its like the queue is determined by a site |
| 07:19.02 | `paul | im in the process of putting everything in the DB and removing the static stuffs |
| 07:21.05 | ChannelZ | Why make hundreds(?) of discreet extensions anyway? Go the other way around, make one that matches a pattern and calls an AGI or something to do the DB lookup and figure out what to do. |
| 07:24.23 | `paul | is that better? then for every call i do a query? cause on my method i just queryl the DB when there is a change? |
| 07:25.52 | ChannelZ | I guess it depends on how many you have, how often it changes, and how up-to-date you need it to be. Right now you have to re-generate, re-write and re-load the dialplan every time you make a change |
| 07:26.43 | `paul | thats good point |
| 07:27.42 | `paul | ok thanks |
| 07:27.49 | `paul | will look into agi approach |
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| 08:13.15 | TobSnyder | short question: is it possible to enable instant message function, e.g. when using x-lite there is an IM functionality but I do get the response "method not allowed" when trying to send messages |
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| 08:21.46 | kaldemar | TobSnyder: what version of asterisk are you using and what method is x-lite using for the messages? |
| 08:24.17 | TobSnyder | asterisk 1.4.33.1 - x-lite seems to use SIP/Simple MEssaging |
| 08:24.22 | TobSnyder | but not sure on x-lite |
| 08:24.46 | Tim_Toady | lats time i checked asterisk had not support for sip messaging, dont know if they added in 1.8 but i think not |
| 08:24.53 | kaldemar | TobSnyder: looks like SIP messages are not allowed outside of a call, and that causes the 405 method not allowed. this is the case in 1.8.0 rc2 too. |
| 08:25.09 | TobSnyder | actually I just wanted to try it internal |
| 08:25.10 | kaldemar | the messages are supported when a call is up though. |
| 08:25.24 | TobSnyder | what happens on external calls would have been my next question ;) |
| 08:25.57 | TobSnyder | does this also apply on asterisk 1.4.33 ? |
| 08:26.09 | kaldemar | TobSnyder: yes |
| 08:26.39 | kaldemar | first you'd have to define what an external call is from your point of view. |
| 08:27.05 | TobSnyder | just lets talk about internal |
| 08:27.11 | TobSnyder | and external later |
| 08:27.31 | TobSnyder | internal means from sip sofrtphone to another sip softphone (both x-lite) |
| 08:27.36 | kaldemar | what do you mean by internal? calls are just calls to asterisk. |
| 08:27.48 | TobSnyder | sure |
| 08:27.59 | TobSnyder | and external would be a call over pstn, e.g. ISDN |
| 08:28.04 | kaldemar | well, make a call between them and send a message. if x-lite uses SIP MESSAGE, it should work. |
| 08:28.12 | TobSnyder | ok |
| 08:28.53 | kaldemar | well, you can't send SIP messages over ISDN, some conversion would need to be made. depends on how you deliver calls. |
| 08:33.19 | TobSnyder | I still get "Error: Method not allowed" |
| 08:33.53 | TobSnyder | have just made a call and tried to send instant message |
| 08:36.49 | kaldemar | time for a sip debug and pastebin... |
| 08:37.09 | *** part/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 08:39.15 | Tim_Toady | only way i know to get sip messaging is to set a ser/openser/opensips/kamailio/whatever_is_called_this_week sip proxy infront of * and let it handle the sip traffic |
| 08:40.25 | TobSnyder | http://pastebin.com/bGqUFhEu |
| 08:43.04 | TobSnyder | Tim_Today: am I right that when using openSER (or similar) I don't need asterisk at all? |
| 08:44.23 | Tim_Toady | depends, if u just use sip and u dont want to connect to landlines/pstn/isdn/gsm or u dont want voicemail and other services then u can replace it |
| 08:44.48 | Tim_Toady | usually u get the to work together and let ser handle the sip leg of the call |
| 08:44.59 | Tim_Toady | s/the/them/ |
| 08:45.19 | TobSnyder | ah I see |
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| 08:54.35 | TobSnyder | http://www.packtpub.com/article/comparing-asterisk-and-openser made it clearer for me |
| 08:59.31 | TobSnyder | kaldemar: any idea ? |
| 09:01.14 | TobSnyder | if there would be an easy patch to asterisk or even just a config switch to enable simple sip messaging in asterisk it would be great |
| 09:01.39 | TobSnyder | the integration of SER with asterisk seems to demand some more time |
| 09:03.43 | Tim_Toady | ser is a PITA to setup |
| 09:03.45 | Tim_Toady | :P |
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| 09:17.32 | creativx | aha |
| 09:17.36 | creativx | soxmix is chewing cpu like its 1999 |
| 09:17.45 | creativx | no wonder people are complaining about choppy sip audio |
| 09:17.50 | TobSnyder | Tim_Today: thats what I guessed |
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| 09:23.56 | kaldemar | TobSnyder: don't try to use asterisk for instant messaging until it has proper support for it? see this: http://www.slideshare.net/saghul/asterisk-im-and-presence-how |
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| 09:31.56 | creativx | can i make soxmix / lame always run at lowest priority? |
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| 09:56.36 | kaldemar | creativx: use nice to run it? man nice |
| 09:58.19 | creativx | kaldemar: yeah looking into nice/renice |
| 09:58.34 | creativx | ironic choice of app name.. since the app is behaving bad :) |
| 09:59.42 | henk | i'd say that's exactly the opposite of irony?! o_O |
| 10:00.27 | creativx | hehe |
| 10:01.47 | eMBee | hmm, it's a command, like kill kills something, nice makes something be nice |
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| 10:37.53 | TobSnyder | kaldemar: thanks for the link to the slide |
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| 10:42.56 | Markous | Hello, i need help on how to connect to the Asterisk Manager |
| 10:43.04 | Markous | without getting this error [Oct 7 12:20:21] ERROR[23748]: utils.c:1173 ast_careful_fwrite: fwrite() returned error: Broken pipe |
| 10:44.49 | Markous | i use PHP as programing lang. |
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| 11:25.18 | fors1 | any aussies in here? I could use a recommendation for a good provider in the Melbourne area. I'm not from, nor based in Australia my self, so I have no local knowledge about the different providers. SIP or IAX for a small business |
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| 11:42.10 | TobSnyder | does anyone have some experiences with SNOM phones? |
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| 12:09.10 | schmidts | tobsnyder yes but you dont want to hear them ;) |
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| 12:23.54 | TobSnyder | y |
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| 12:25.51 | TobSnyder | schmidts: please let me know about ) |
| 12:26.53 | doolittlework | hi there, i am linking two servers with IAX2, working i want to use a huge extension number to pass data from one server to another something linke this Id-003_xnumber-7843233221_unicode-343332 i can get the call on the other side but if the string changes say to Id-0003_xnumber-7843233221_unicode-343332774 how can i fix this |
| 12:27.46 | doolittlework | i in turn what to add the data to a mysql database, but id the string length varies, breaks my code |
| 12:30.03 | schmidts | tobsnyder we have used arond 100 pieces off the snom360 on different places and really had strange problems |
| 12:31.15 | schmidts | tobsnyder the bug i really loved happens if you put on call on hold to start a transfer (opening a second call) and a third call comes in, when you press transfer you still have the first call on hold, and transfered the third call to the second |
| 12:32.02 | schmidts | tobsnyder and we also had some strange one way audio things after several minutes (>20) if a snom has done a transfer |
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| 12:34.49 | schmidts | and only for a personal opinion they are just ugly if you compare to a cisco spa525g for example which costs nearly the same and is much more reliable |
| 12:38.04 | drmessano | Cisco? Take that back, now! |
| 12:39.39 | [TK]D-Fender | drmessano: that is effectively a Linksys model |
| 12:39.40 | *** join/#asterisk francispereira (~francis@124.153.66.214) |
| 12:39.47 | [TK]D-Fender | drmessano: Which I'd say is jsut fine really... |
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| 12:41.35 | francispereira | I forward calls coming to a select few extensions. How do i get asterisk to show the source caller's number when the call is forwarded ? |
| 12:42.37 | drmessano | [TK]D-Fender, he said the "C" word |
| 12:50.39 | TobSnyder | mh |
| 12:50.48 | TobSnyder | it seems there is even no good SIP Hardphone? |
| 12:51.08 | schmidts | Linksys are good even if they know named with the bad C word |
| 12:51.26 | [TK]D-Fender | francispereira: First you can't forward to a "few" numbers. Second it depnds HOW you did this "forward" |
| 12:51.29 | schmidts | but they still use the old sipura webinterface which is IMHO the best out there |
| 12:51.53 | [TK]D-Fender | Best web interface = NO web interface |
| 12:57.06 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:57.06 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:58.02 | francispereira | [TK]D-Fender, The call comes in through the PRI on my media gateway. The gateway forwards all incoming calls on to the asterisk box where each DID is an extension in sip.conf whose context is "forward_calls". In the dial plan "forward_calls" i say anything that comes to an extensions forward it to another number |
| 13:00.52 | [TK]D-Fender | francispereira: I fail to see a clear definition of what is "forwarded" here. |
| 13:01.17 | [TK]D-Fender | francispereira: So far all I see is a context with the word "forward" in it's name. Name doesn't matter |
| 13:02.36 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 13:04.28 | francispereira | [TK]D-Fender, here is what my sip and extensions.conf look like http://pastebin.com/NLZSGCxN |
| 13:04.54 | TobSnyder | http://www.sipgate.de/voipshop/tiptel/tiptel_83_voip |
| 13:05.14 | TobSnyder | 60 ⬠for a SIP phone |
| 13:05.42 | [TK]D-Fender | francispereira: I fail to see a suitable description of where an actual TRANSFER occurs |
| 13:05.46 | drmessano | schmidts, I haven't touched a Linksys Web UI in ages. All TFTP/HTTP here |
| 13:05.49 | [TK]D-Fender | francispereira: Dialplan means nothing here so far. |
| 13:06.15 | [TK]D-Fender | TobSnyder: Completly no-name cheap shit phone... |
| 13:06.18 | [TK]D-Fender | TobSnyder: .... |
| 13:06.21 | [TK]D-Fender | ~ygwypf |
| 13:06.21 | infobot | it has been said that ygwypf is You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 13:06.26 | leifmadsen | drmessano: +1 |
| 13:06.41 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:07.21 | *** join/#asterisk laggo (~chatzilla@nat67.mia.three.co.uk) |
| 13:07.40 | TobSnyder | but when I see that even Snom 360 has problems like described by schmidts |
| 13:08.22 | francispereira | [TK]D-Fender, a call destined for 39382006 comes via the PRI on the media gateway which sends all calls to the Asterisk box. Since its context is forward_calls and there is a entry for it in the dial plan i.e # |
| 13:08.22 | francispereira | exten => 39382006,1,Dial(SIP/trunk_media_gateway_34/<9898989898989>) the call gets forwarded to 898989898989 |
| 13:08.31 | francispereira | 9898989898989 |
| 13:08.45 | francispereira | maybe i am doing this the wrong way |
| 13:08.47 | [TK]D-Fender | francispereira: that is NOT a forward! That is just a DIAL |
| 13:08.50 | francispereira | but it seems to work for me so far |
| 13:08.59 | drmessano | TobSnyder, you name SPECIFIC products, then when someone says "that's a POS" you gravitate to "I guess there is no good SIP phone"... No, everything you name just happens to be shit |
| 13:09.00 | [TK]D-Fender | francispereira: Never call Dial() "forward". |
| 13:09.03 | francispereira | Whats the right way to forward ? |
| 13:09.14 | [TK]D-Fender | francispereira: You are misuing the word. |
| 13:09.18 | [TK]D-Fender | mis-using* |
| 13:09.28 | drmessano | TobSnyder, Linksys, Polycom seem to be favorites.. stop pulling up no-name chinese shit |
| 13:09.35 | francispereira | so can i say Forward()? |
| 13:09.41 | [TK]D-Fender | francispereira: * does not even get out fo the way. it is sitting int he middle of the call. |
| 13:10.02 | TobSnyder | just wonder why even snom 360 for something around 200 euro, "made in germany" also is shit |
| 13:10.27 | [TK]D-Fender | francispereira: and if the number changes from the # coming in then it is your peer setup that is wrong. |
| 13:11.01 | [TK]D-Fender | TobSnyder: Because the ability to make shit products is not geographically limited <- |
| 13:11.03 | drmessano | TobSnyder, schmidts cites SPECIFIC issues with that SPECIFIC model. Do you not believe it is remotely possible for a MODEL to have issues? |
| 13:11.37 | drmessano | TobSnyder, and since when is PRICE a function of QUALITY? |
| 13:11.50 | francispereira | would saying exten => 39382006,1,Forward(SIP/trunk_media_gateway_34/<9898989898989>) be the right way to forward a call ? |
| 13:11.54 | malcolmd | possibly; but snom runs similar/same firmware across the 3xx series phones, no? |
| 13:12.04 | TobSnyder | ~ygwypf |
| 13:12.05 | infobot | ygwypf is, like, You Get What You Pay For. If the sole factor in your decision to purchase a product or service is that it's cheaper than everything else out there, don't be surprised if it's also worse in every other respect than everything else out there. |
| 13:12.47 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:12.51 | [TK]D-Fender | francispereira: Why are your continuing to use the word "forward"? YOu are incredibly unclear about what you are actually trying to do, and what your current approach's problem is |
| 13:13.33 | m_tadeu | hi...I'm trying to use PlayDTMF within the AMI. I send the command and the response is "Success: DTMF successfully queued", but no sound is hear in the softphone |
| 13:13.35 | francispereira | [TK]D-Fender, ok back to square one |
| 13:13.39 | [TK]D-Fender | TobSnyder: Cheap shit tends to be shit. Just because it is comparitively expensive doesn't necessarily mean it's GOOD however |
| 13:13.41 | francispereira | i am pretty lost |
| 13:13.47 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 13:13.47 | francispereira | i have to forward incoming calls |
| 13:13.50 | TobSnyder | so I will check out polycom and linksys |
| 13:13.57 | [TK]D-Fender | TobSnyder: today's magic phrase is "not mutually exclusive" |
| 13:13.57 | francispereira | and I am using the above approach |
| 13:14.12 | [TK]D-Fender | francispereira: DEFINE "forward". |
| 13:14.32 | *** part/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 13:14.39 | francispereira | if a call come on number A it should automatically be redirected to number B |
| 13:14.53 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 13:15.26 | [TK]D-Fender | francispereira: Well Dial() will call out and BRIDGE teh call. Does that work? |
| 13:16.18 | francispereira | yes it does |
| 13:16.26 | francispereira | works like a charm |
| 13:16.56 | [TK]D-Fender | francispereira: Then if the number that comes in is getting overriden, then it is the faul of your peer definition for where you are dialing out to. |
| 13:17.04 | [TK]D-Fender | francispereira: fault* |
| 13:17.29 | [TK]D-Fender | francispereira: Most common trouble item is seeting "fromuser" |
| 13:17.34 | [TK]D-Fender | setting* |
| 13:18.24 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:19.34 | francispereira | the call originates from a hard phone that comes in through the PRI line. are you saying that my entry in the sip.conf is not correct ? |
| 13:22.03 | [TK]D-Fender | francispereira: If you are saying * gets the proper callerID when the call arrives, but doesn't SEND the right caller-id when you call out then it is your SIP PEER that is wrong, or the receiving end is IGNORING the caller-id you send |
| 13:22.59 | francispereira | [TK]D-Fender, how do i check of the correct called id is coming in . Maybe some NoOp in the dial plan ? |
| 13:23.20 | [TK]D-Fender | francispereira: So you don't even know if it comes IN right? |
| 13:24.24 | francispereira | It does what its supposed to do. "Forwarding" from one number to another work, by using the Dial(). But no, i dont know anything more |
| 13:25.43 | [TK]D-Fender | francispereira: You seem to be telling me you haven't actually LOOKED at your actual calls yet |
| 13:26.03 | *** join/#asterisk analogkid (~analog@dynk214.osnanet.de) |
| 13:27.08 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:27.13 | francispereira | sorry, i am certain this is pretty basic, but how do i look at an actual call ? I have been using to -vvvr to see the debug info. But thats all. Please point me in the right direction |
| 13:27.27 | francispereira | and the cdr logs |
| 13:27.42 | *** join/#asterisk dacm_work (~dan@host86-166-151-3.range86-166.btcentralplus.com) |
| 13:28.17 | dacm_work | Hi guys. |
| 13:28.43 | dacm_work | Does asterisk need any special settings to work in the UK when using a SIP trunk? |
| 13:28.58 | mmlj4 | I can't see why |
| 13:29.01 | dacm_work | I'm having a problem with one of my analogue phgones |
| 13:29.24 | mmlj4 | SIP is SIP |
| 13:29.27 | dacm_work | I'm having a problem with one of my analogue phones and don't know if it is just because of the adapter or asterisk itself. |
| 13:29.30 | dacm_work | ok |
| 13:29.37 | dacm_work | Must be the adapter then. |
| 13:29.45 | mmlj4 | mjght be |
| 13:29.57 | mmlj4 | what brand? |
| 13:30.00 | [TK]D-Fender | francispereira: Go enable SIP DEBUG at CLI and pastebin a complete call in/out |
| 13:30.07 | dacm_work | Basically the dial tone seems to time out before this particular phone can be bothered to dial. |
| 13:30.28 | [TK]D-Fender | dacm_work: What does a SIP trunk have to do with an analog phone? |
| 13:30.29 | mmlj4 | tried another phone? |
| 13:30.32 | dacm_work | mmlj4: The adapter is a Linksys/Cisco 3102. |
| 13:30.44 | mmlj4 | those are decent |
| 13:31.07 | mmlj4 | you're not using a rotary phone perchance? no idea if those work |
| 13:31.30 | [TK]D-Fender | [09:30]<dacm_work>Basically the dial tone seems to time out before this particular phone can be bothered to dial. <-- please be EXTREMELY clear about which interface you are fixing here.. the FXO, or the FXS, and where the issue occurs |
| 13:31.31 | dacm_work | [TK]D-Fender: It doesn't. Just wanted to point out that asterisk was directly connection to the PTSN (which may well need special settings, I don't know). |
| 13:31.40 | *** join/#asterisk timeshell (~timeshell@204.101.237.192) |
| 13:32.06 | dacm_work | mmlj4: Nah it's a DECT phone from panasonic. |
| 13:32.30 | mmlj4 | panajunk # sorry, I used to repair electronics for a living |
| 13:32.36 | [TK]D-Fender | dacm_work: " asterisk was directly connection to the PTSN" <_ WTF? |
| 13:32.46 | [TK]D-Fender | dacm_work: Yuor description is becoming even MORE vague |
| 13:33.27 | dacm_work | [TK]D-Fender: s/was/wasn't/ |
| 13:33.34 | dacm_work | [TK]D-Fender: Sorry about that. |
| 13:33.49 | [TK]D-Fender | dacm_work: was vs wasn't != any clearer |
| 13:33.50 | dacm_work | [TK]D-Fender: It doesn't. Just wanted to point out that asterisk wasn't directly connecting to the PTSN (which may well need special settings, I don't know). |
| 13:33.56 | [TK]D-Fender | dacm_work: Start over. |
| 13:34.12 | dacm_work | Which was an answer to <[TK]D-Fender> dacm_work: What does a SIP trunk have to do with an analog phone? |
| 13:34.20 | [TK]D-Fender | dacm_work: And WTF is a "direct" connection, let alone an INDIRECT one? |
| 13:34.38 | [TK]D-Fender | dacm_work: Just start over. Completely. |
| 13:35.19 | *** join/#asterisk timeshell (~timeshell@204.101.237.192) |
| 13:35.22 | francispereira | [TK]D-Fender, here is what sip debug looks like http://pastebin.com/HANdnKy5 |
| 13:35.26 | dacm_work | [TK]D-Fender: Well direct would be connecting to the PTSN through an analogue line or ISDN, indirect would be through a 3rd party, i.e. a SIP trunk. |
| 13:36.35 | mmlj4 | dacm_work: sorry I could not help you, I hope you get it sorted |
| 13:37.12 | [TK]D-Fender | dacm_work: Where is the problem? You have an ITSP (stop calling it a SIP TRUNK", and a Linksys ATA with 2 different kinds of ports on it. That is THREE things. Now which ONE of those is the issue? |
| 13:37.55 | dacm_work | I have an analogue phone connected to an adapter (Cisco 3102) connected to both an analogue line and an asterisk server (which connects to a SIP trunk). When I dial on the analogue phone the dial tone seems to time out before the phone dials (~3s). Was just wondering if asterisk could cause that or if it must be the adapter that's doing it. |
| 13:38.01 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:38.31 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:39.01 | dacm_work | [TK]D-Fender: I've noticed the timeout on the landline so far. Need to check if it still happens when calling internal lines over asterisk. |
| 13:39.07 | dacm_work | hang on |
| 13:39.14 | dacm_work | mmlj4: Thanks. |
| 13:39.51 | [TK]D-Fender | dacm_work: what "land-line"? Timeout WHERE? At what point of what process? |
| 13:40.01 | [TK]D-Fender | dacm_work: Still completely vague. |
| 13:43.41 | dacm_work | [TK]D-Fender: The land-line (analogue line) connected to the Cisco 3102. The time-out is on the dial tone on the phone. It you pick up the phone you get a dial tone for 3 seconds and then it changes which seems to be some kind of timeout. |
| 13:43.53 | *** join/#asterisk laggo (~chatzilla@nat67.mia.three.co.uk) |
| 13:44.09 | dacm_work | [TK]D-Fender: Sorry if I'm still being vague, I'm trying to be as specific as possible. |
| 13:44.14 | [TK]D-Fender | dacm_work: Picking up the phone is the FXS, not the FXO. Why am I hearing about *@* things at one here? |
| 13:44.33 | [TK]D-Fender | dacm_work: Those 2 ports typically have JACK SHIT to do with each other. |
| 13:44.37 | laggo | i've set up a new install of asterisk on my server and it's registering to my sip account just fine. but i can't seem to connect my softphone to it (it's like the server isn't accepting the connection, even though the firewall is turned off). how do i diagnose this? |
| 13:44.50 | [TK]D-Fender | 2* |
| 13:45.14 | [TK]D-Fender | laggo: do you see the call arrive with SIP DEBUG enabled? |
| 13:45.19 | dacm_work | [TK]D-Fender: I just wanted to know if I could eliminate * as the cause of the issue. It only seems to occur once the 3102 has registered with * you see. |
| 13:45.19 | laggo | no |
| 13:45.45 | laggo | it's not a call, i'm just trying to connect my softphone via the sip friend entry |
| 13:46.03 | [TK]D-Fender | dacm_work: You are bringing multiple elements into the picture and not isolating ONE as being the problem or describing WHT is "timing out" |
| 13:46.07 | laggo | i can see calls arrive from the sip account i've registered the server to |
| 13:46.24 | [TK]D-Fender | laggo: Yes, well do you see the REGISTER come in or not? |
| 13:46.33 | laggo | no |
| 13:46.35 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 13:47.03 | [TK]D-Fender | laggo: then check your firewalls, routing, etc, end to end |
| 13:47.53 | laggo | [TK]D-Fender: for the REGISTER to appear, the server just needs to be accepting connections on the port (default 5060) right? |
| 13:48.02 | laggo | i.e. port forwarding wouldn't be an issue yet |
| 13:48.03 | dacm_work | [TK]D-Fender: Well once the 3102 registers with * the dial tone seems to time out afetr just 3 seconds. By timeout I mean that the dial tone changes and no longer seems to accept any dialled numbers and requires the user to hang up and dial again. |
| 13:48.11 | pigpen | Hi all, I am running Asterisk 1.6.2.13, new deployment, when I have a call passed from an IAX trunk (gsm) to voicemail, I get the following: |
| 13:48.12 | pigpen | NOTICE[10258]: channel.c:3079 __ast_read: Dropping incompatible voice frame on IAX2/sjs-ccnbi-8590 of format slin since our native format has changed to 0x2 (gsm) |
| 13:48.22 | pigpen | At this time the audio stops. |
| 13:48.37 | [TK]D-Fender | dacm_work: that is a setting on the SPA for the FXS dial timeout. |
| 13:48.50 | [TK]D-Fender | dacm_work: Go read the manual to see which parameter to adjust in it |
| 13:48.54 | pigpen | like: "Plea......." as in "Please leave a message at the tone" |
| 13:49.18 | dacm_work | [TK]D-Fender: Ok. I'll look into that. Thank you very much for the pointer, as well as your patience. |
| 13:49.29 | *** part/#asterisk analogkid (~analog@dynk214.osnanet.de) |
| 13:49.34 | pigpen | I have had this issue with several production boxes, but have downgraded to 1.6.1.x to resolve the issue for the time. So it seems as if it is in 1.6.2.x at this point. |
| 13:49.38 | mmlj4 | well, that took only 15 minutes |
| 13:49.39 | pigpen | ideas? |
| 13:49.41 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-bgkyrbegtyezerbo) |
| 13:49.45 | francispereira | [TK]D-Fender, any leads ? |
| 13:52.12 | [TK]D-Fender | francispereira: From: <sip:2243484673 <--- is this the proper callerID coming in? |
| 13:53.10 | [TK]D-Fender | francispereira: -- Executing [39382000@forward_calls:1] Dial("SIP/trunk_media_gateway_32-000033be", "SIP/trunk_media_gateway_34/9890960855") in new stack <-- this is you calling out. |
| 13:53.15 | *** join/#asterisk eugeneoden (~goden@rrcs-97-77-255-86.sw.biz.rr.com) |
| 13:54.01 | [TK]D-Fender | francispereira: From: "2243484673" <sip:2243484673@10.236.153.123>;tag=as3b148e76 <--- This is the called CALLERID coming in being used as the "from" when you call out. |
| 13:54.01 | francispereira | yes it is |
| 13:54.08 | laggo | is port 5060 supposed to be open for TCP as well? |
| 13:54.21 | [TK]D-Fender | francispereira: So * is passing this # on. If the other side is overriding this, then it isn't *'s fault or problem. |
| 13:54.33 | [TK]D-Fender | laggo: Not by default |
| 13:54.36 | *** join/#asterisk b_d (~brian@199.172.227.165) |
| 13:56.06 | francispereira | line 79 and 80 right ? |
| 13:57.09 | *** join/#asterisk ronndonn (~sam@c-68-34-165-45.hsd1.pa.comcast.net) |
| 13:57.54 | [TK]D-Fender | francispereira: 76 is the real "to" which is exactly who you dialed. 79 is the callerID you are passing |
| 13:58.09 | [TK]D-Fender | francispereira: So you are giving them the right number, and they are ignoring it |
| 13:58.15 | laggo | [TK]D-Fender: could you give me a bit more help testing this setup? |
| 13:58.38 | *** join/#asterisk BANSAL (~bansal@117.199.123.228) |
| 13:58.49 | [TK]D-Fender | laggo: As in? |
| 13:59.07 | laggo | i'm just lost, i've been doing this for hours. i can't seem to get this stupid softphone to register |
| 13:59.07 | ronndonn | Hi all. Trying to set up two separate home offices with the a single phone number. I have asterisk set up and working, call the number and rings in both offices. Problem I have is we can only pick up and talk from my location. Here's what my sip.conf looks like and some output from asterisk when the other office picks up: http://pastie.org/1205290. Sorry if I sound dopey, I don't know how else to explain the issue. |
| 13:59.11 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 13:59.21 | laggo | i have no idea how to diagnose where the problem is |
| 13:59.28 | [TK]D-Fender | laggo: If packets aren't arriving, it isn't *'s fault. |
| 13:59.38 | [TK]D-Fender | laggo: And you haven't described your scenario at all. |
| 14:01.21 | laggo | asterisk is connecting to a sip account i have at a free provider, set up as a sip peer - incoming calls on this work just fine. i'm trying to connect a softphone to asterisk with a sip friend account i've set up in sip.conf, but nothing happens at all, it's as if the softphone never tried to connect. |
| 14:01.24 | [TK]D-Fender | ronndonn: who is "my location"? and describe the actual symptom of the failure |
| 14:02.09 | [TK]D-Fender | laggo: Anything else to add? |
| 14:03.02 | laggo | i've disabled all firewalls on the server. locally i'm behind a NAT router but outbound connections work fine (i can ping the server, i can connect to it by ssh) |
| 14:03.23 | ronndonn | [TK]D-Fender: sorry, my location is Sam and the symptom is Owen hears the ring but cannot talk |
| 14:03.53 | *** join/#asterisk zoid_ (~awainer@190.2.14.213) |
| 14:05.35 | [TK]D-Fender | ronndonn: Does * acknowledge that he answers? |
| 14:05.44 | ronndonn | yes |
| 14:06.01 | ronndonn | -- SIP/Owen-101c4cf0 answered SIP/mynumber-101cbba0 |
| 14:06.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:07.02 | [TK]D-Fender | ronndonn: And where is your phone located relative to *? |
| 14:07.19 | [TK]D-Fender | laggo: Same goes with YOU <- |
| 14:07.38 | [TK]D-Fender | laggo: You haven't described ANYTHING about what is trying to conenct to *, and what path t takes. |
| 14:08.17 | laggo | it is just a piece of software, i've tried a couple of different softphones. it's behind a NAT router |
| 14:08.23 | ronndonn | [TK]D-Fender: on my local network |
| 14:09.20 | [TK]D-Fender | laggo: Please provider a proper complete description.... |
| 14:09.39 | laggo | i don't know what else to say |
| 14:09.44 | [TK]D-Fender | ronndonn: no packets means firewall, pointing to the wrong IP, or general networking issue |
| 14:09.50 | [TK]D-Fender | laggo: ....... |
| 14:10.20 | laggo | maybe i'll try a different bindport |
| 14:12.00 | ronndonn | [TK]D-Fender: pointing to the wrong IP, do you mean the externip in my sip.conf? |
| 14:13.30 | [TK]D-Fender | ronndonn: I'm talking about your PHONE pointing to ASTERISK |
| 14:15.59 | laggo | [TK]D-Fender: is there a way to try everything using TCP instead of UDP? |
| 14:20.18 | ronndonn | [TK]D-Fender: I'm confused, wouldn't the fact that * recognizes Owen's answer mean his phone is pointing to * |
| 14:21.47 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.84.68.adsl.dynamic.totbb.net) |
| 14:23.32 | BugKhaM | I'm using zaptel + libpri with TE110P and experiencing a problem where the caller is hung up when entering DTMF too fast. |
| 14:23.48 | BugKhaM | any idea where I should look at? |
| 14:24.24 | [TK]D-Fender | ronndonn: Mixed your issue up. Disregard. pastebin a NEW call with SIP DEBUG enabled |
| 14:24.36 | BugKhaM | I also have relaxdtmf=yes in zapata.conf |
| 14:24.57 | [TK]D-Fender | BugKhaM: I do not see any relation between the two |
| 14:24.58 | drmessano | BugKhaM, Zaptel is old as hell |
| 14:25.14 | drmessano | BugKhaM, could be some ancient bug.. you should try DAHDI |
| 14:25.28 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 14:26.39 | *** join/#asterisk Tim_Toady (~moi@193.92.224.201.dsl.dyn.forthnet.gr) |
| 14:30.06 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:30.06 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:31.03 | BugKhaM | drmessano: yeah, for some reasons I need to use asterisk 1.2, so I can't use DAHDI |
| 14:31.36 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:34.16 | *** join/#asterisk Nwab (~Schnitzel@unaffiliated/benwa) |
| 14:34.53 | [TK]D-Fender | ronndonn: Check your firewalls on the remote side? What ports do you have forwarded to*? What router are THEY behind? |
| 14:35.24 | *** join/#asterisk bcrisp (~bcrisp@wsip-184-191-141-38.ph.ph.cox.net) |
| 14:35.32 | drmessano | BugKhaM, 1.2 is extinct. If this were a bug, there is nothing anyone can do to help you |
| 14:35.51 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:36.32 | bcrisp | hi all .. using 1.6.1.11 for quite a while now. Noticed that we were losing dialplan functionality for no apparent reason. After rebooting the server and running CLI, we see "Unable to connect to remote asterisk message |
| 14:36.36 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:37.40 | bcrisp | not sure how to proceed |
| 14:37.40 | drmessano | bcrisp, so, Asterisk isn't running at startup then |
| 14:38.04 | bcrisp | i ran service asterisk start, receive an OK message, but still can't launch CLI |
| 14:38.19 | ronndonn | [TK]D-Fender: the port is 5060, 7070 - 7080, the remote side sip is pointing to 5060 |
| 14:39.11 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:39.38 | bcrisp | what we noticed before this issue was that dialplan functionality was being lost.. actually it looked like it was taking a very long time to load the dialplan with dialplan reload from CLI it was normally very fast.. then very very slow, writing an info message every 1 second or so for each extension found |
| 14:39.49 | [TK]D-Fender | ronndonn: You should be forwarding 10000-20000 unless you changed your port range in rtp.conf |
| 14:40.41 | bcrisp | ok it is a space issue |
| 14:40.46 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:40.47 | bcrisp | out of disk space on the server yikes |
| 14:41.05 | bcrisp | "unable to open pid file '/var/run/asterisk.pd' |
| 14:41.30 | drmessano | bcrisp, 20GB HD? |
| 14:41.43 | bcrisp | honestly i don't remember .. hosted rack shack server |
| 14:42.09 | bcrisp | with revolving logs and no call recording didn't think we'd have a space issue |
| 14:43.56 | *** join/#asterisk garymc (~chatzilla@host81-139-127-32.in-addr.btopenworld.com) |
| 14:44.55 | ronndonn | [TK]D-Fender: my rtp.conf has: rtpstart=7070 rtpend=7080 |
| 14:45.18 | [TK]D-Fender | ronndonn: Remote side shouldn't have any forwarding at all |
| 14:45.22 | laggo | [TK]D-Fender: okay, i've got the register packets going now! |
| 14:45.30 | [TK]D-Fender | ronaand I asked what ROUTER they are using |
| 14:45.39 | laggo | asterisk is responding to all of them with 401 Unauthorized |
| 14:45.57 | ronndonn | [TK]D-Fender: airport extreme |
| 14:46.07 | [TK]D-Fender | ronndonn: try another |
| 14:49.34 | pigpen | Ok, I have narrowed down my issue. On 1.6.2.x, when I have calls that have any sort of playback (either by the command playback or from the VM prompts), the audio might start, and if it does, it is stopped. No errors. |
| 14:50.26 | pigpen | I have tried this with the default sounds that came with the 1.6.2 dist, 1.6.1 dist, recordings from voice vector and custom recordings recorded directly on the system. |
| 14:50.47 | pigpen | any help would be greatly appreciated, as I don't want to stay on 1.6.1 forever. |
| 14:51.48 | ronndonn | [TK]D-Fender: sarcasm? He does not have another router handy. Im using the same router here, seems to be working |
| 14:52.13 | bcrisp | my /var/log/secure file is HUGE |
| 14:53.57 | *** join/#asterisk bent_screwdriver (~socain00@173-146-34-253.pools.spcsdns.net) |
| 14:54.26 | *** join/#asterisk deadpigeon (~kvirc@office.xpressamerica.net) |
| 14:55.53 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 14:55.53 | deadpigeon | anyone familiar with smdi? ive gotten it to work fine except i have these ping messages that I see when I cat my serial port, every 2 minutes.. and they eventually add up until asterisk is ready to send a MWI message.. what happens is all those ping msgs get tossed at the switch infront of asterisk's mwi, and the switch fails to process the message. |
| 14:56.12 | deadpigeon | if I send the mwi again right after this before another ping message can be generated from wherever, it works great. |
| 14:57.43 | deadpigeon | Just not sure where to start looking for the culprit... it's clearly not the Asterisk process doing this, and I don't believe it's the switch either. |
| 14:59.06 | bcrisp | hmm can you list contents of a file in reverse order with cat? |
| 14:59.48 | deadpigeon | No. |
| 14:59.54 | bcrisp | dangit |
| 15:01.35 | deadpigeon | Even if there's a way to strip the data strings, which there is via the RS232 to IP converter, I wouldn't want to have to go that far to match everything that isn't an SMDI message. |
| 15:02.43 | ronndonn | [TK]D-Fender: would you be willing to look at a screenshot of my remote devices config and let me know if you anything strange? |
| 15:03.37 | bcrisp | yay network attacks on the phone server |
| 15:04.37 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:06.03 | *** join/#asterisk mltlnx (~mltlnx@asams.mserve.com) |
| 15:07.48 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 15:07.55 | mltlnx | Quick question here, I have a timeout on a queue, after the timeout it should go to Voicemail, however it just says "nobody picked up in 60000ms" and then continues in the Queue? Any suggestions? |
| 15:08.57 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 15:10.41 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:11.30 | *** join/#asterisk Indi0 (~bt4@200.110.174.149) |
| 15:12.19 | [TK]D-Fender | mltlnx: Maybe you you should SHOW US. |
| 15:12.38 | mltlnx | good idea... |
| 15:13.40 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:15.00 | p3nguin | Showing us what's going on so we can get a better idea of the problem and devise a plan to correct it...? That's a TERRIBLE idea! |
| 15:19.53 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
| 15:25.34 | *** join/#asterisk BANSAL (~bansal@117.199.120.223) |
| 15:29.45 | bcrisp | ugh |
| 15:29.58 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 15:30.02 | visik7 | hi |
| 15:30.16 | WIMPy | lo |
| 15:30.18 | metiu_ | my asterisk is using console/dsp to talk to a speaker. It starts with less than 5% CPU. After a while, it rockets to almost 100%, how could I debug it? |
| 15:30.23 | visik7 | anyone using iaxmodem with mgetty ? |
| 15:31.03 | *** join/#asterisk doctorpepper (298defa9@gateway/web/freenode/ip.41.141.239.169) |
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| 15:40.17 | pigpen | [TK]D-Fender, would you have a few minutes? |
| 15:41.16 | [TK]D-Fender | possibly |
| 15:41.44 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 15:41.45 | pigpen | as noted above versions, 1.6.2.x, upon any playback (playback, background, vm) |
| 15:41.58 | pigpen | the audio starts, then quits |
| 15:42.09 | pigpen | Also, I noticed the dialplan progress stopps. |
| 15:42.33 | pigpen | Asterisk does not show any error or issue. Other calls are not affected. |
| 15:43.04 | pigpen | I have only had this issue on 1.6.2.x (I have been trying 1.6.2.x across many versions, hoping this would be resolved) |
| 15:43.35 | pigpen | So I can't believe I am the only one with this issue, so I am starting to believe either it is a bug or it is my dialplan. |
| 15:43.57 | pigpen | but, bare dialplan, with noting more than a simple playback, it happens. |
| 15:44.46 | leifmadsen | pigpen: what is the dialplan? |
| 15:45.00 | leifmadsen | pigpen: you probably need a Progress() or Answer() prior to the Playback() |
| 15:45.22 | pigpen | Ah, tried that. |
| 15:45.29 | pigpen | I saw those too. |
| 15:45.38 | pigpen | I am not doing early media, but I tried that anyway. |
| 15:46.25 | *** join/#asterisk BANSAL (~bansal@117.199.112.147) |
| 15:46.37 | pigpen | One thing I will try is a direct sip phone. Currently I am SIP->Asterisk 1.6.1.x-> IAX -> Asterisk 1.6.2.x |
| 15:50.09 | pigpen | Ok, just tried it locally (Polycom SIP -> Asterisk 1.6.2.x) and it worked fine. |
| 15:50.40 | pigpen | So it must be something involving the IAX |
| 15:52.02 | pigpen | Ok, more info: |
| 15:52.34 | *** join/#asterisk Faithful (~Faithful@180.194.2.112) |
| 15:53.15 | pigpen | PRi -> Asterisk 1.6.1.x -> IAX -> Asterisk 1.6.2 -> dialplan which simply "exten => 888,1,Playback(en/vm-theperson)" |
| 15:53.21 | pigpen | ^^^ does not work. |
| 15:53.32 | pigpen | Audio hangs after about 1 second of audio. |
| 15:54.13 | pigpen | The First asterisk box (1.6.1.x) has 6 other iax trunks to other 1.6.1.x boxes with varying loads. |
| 15:54.25 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.109.144.dsl.dyn.forthnet.gr) |
| 15:54.31 | pigpen | same codec. NAT is not involved. |
| 15:55.13 | pigpen | I can call in this fashion, and carry on a conversation to a remote person just fine. |
| 15:55.23 | pigpen | Just when there is a playback involved, it gets borked. |
| 15:55.37 | *** join/#asterisk fleixius (~fleixius@unaffiliated/fleixius) |
| 15:55.57 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:56.01 | pigpen | If I don't hear any ideas soon, I'll just open a bug. |
| 15:56.08 | *** join/#asterisk umay (~chris@174-29-61-74.hlrn.qwest.net) |
| 15:56.21 | pigpen | doesn't know why he gets all the weird shit. |
| 15:57.05 | fleixius | My calls are not being disconnected on my Cisco CME even when the client hangs up. Any ideas? |
| 15:58.25 | bmoraca_work | fleixius: the moon was backwards last night |
| 15:59.23 | pigpen | shit, I knew there was something. |
| 15:59.57 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:00.20 | Qwell | bmoraca_work: Is that when the werehumans come out? |
| 16:00.51 | bmoraca_work | must be |
| 16:01.26 | pigpen | wonders is jitterbuffer is screwing me. |
| 16:02.58 | *** join/#asterisk wierdo (jimmy@77.78.3.197) |
| 16:04.01 | *** join/#asterisk pbxec (~dcaballer@186.104.171.255) |
| 16:04.41 | pbxec | does anybody have experience using outbound fax detection with 1.6.2.0 DAHDI? |
| 16:06.33 | krion | i got a deadlock again |
| 16:06.47 | krion | can the AMI be in relation with it ? |
| 16:07.08 | krion | because it often happen just after one of my script connect to the AMI |
| 16:08.18 | *** join/#asterisk Z_God (~julius@2001:610:1908:8000:21e:ecff:fe5d:679e) |
| 16:15.02 | *** join/#asterisk m_tadeu (~quassel@89.180.108.21) |
| 16:15.19 | pbxec | what do you use AMI for? |
| 16:16.05 | pbxec | I try to use AMI to generate calls from URL for outbound calls,, but fails with AMD and fax detection |
| 16:16.40 | pbxec | because channel answer is needed for detection and connects the call before |
| 16:18.26 | pbxec | does 1.6.2.0 have something similar to NVfaxdetect? |
| 16:19.02 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 16:19.11 | bmoraca_work | nvfaxdetect has been adapted for 1.6, yes |
| 16:19.17 | bmoraca_work | you can find the source online |
| 16:19.27 | *** join/#asterisk moy_ (~moy@74.12.102.191) |
| 16:20.34 | *** join/#asterisk stoffell (~kristof@dD576AE12.access.telenet.be) |
| 16:21.20 | pbxec | is it compiled by default? |
| 16:25.08 | pbxec | NVfaxdetect is the only way to detect outbound fax using Dahdi? |
| 16:26.44 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:30.33 | bmoraca_work | i am not aware of a statically linked library available for download |
| 16:35.37 | pigpen | https://issues.asterisk.org/view.php?id=18105 |
| 16:35.38 | pbxec | I already compiled spandsp 0.0.5 and found diff file for nvfaxdetect |
| 16:37.47 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:37.52 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 16:39.30 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 16:40.01 | BenC[UK] | Hi Guys, I think I have a weird one - im trying to set a queue to not call busy users.. |
| 16:40.07 | BenC[UK] | I have two extensiosn assigned to the queue |
| 16:40.19 | BenC[UK] | but they're still getting incoming calls when they're already on the phone |
| 16:40.38 | frigidzephyr | is call-limit set for the users ? |
| 16:41.32 | BenC[UK] | checking |
| 16:42.11 | BenC[UK] | call waiting was on! |
| 16:42.23 | BenC[UK] | I've been trying to sort the queue settings :/ |
| 16:43.04 | frigidzephyr | yeah look into call-limit= and limitonpeers= , they may be useful to resolve that issue |
| 16:44.31 | pbxec | I patched 1.6.2.0 with nvfaxdetect diff and app did not show |
| 16:46.10 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 16:47.32 | *** join/#asterisk eugeneoden (~goden@99-62-173-58.lightspeed.austtx.sbcglobal.net) |
| 16:48.42 | leifmadsen | 1.6.2.0? wow |
| 16:48.46 | leifmadsen | <PROTECTED> |
| 16:49.02 | leifmadsen | BenC[UK]: what version of Asterisk? |
| 16:49.18 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 16:49.18 | BenC[UK] | 1.6.2.12 |
| 16:49.44 | leifmadsen | BenC[UK]: you want to enable 'callcounter=yes' in sip.conf |
| 16:49.55 | leifmadsen | BenC[UK]: then enable ringinuse=no in queues.conf |
| 16:50.14 | BenC[UK] | thank you |
| 16:50.19 | leifmadsen | BenC[UK]: call-limit has been deprecated, so don't bother with that |
| 16:51.25 | frigidzephyr | wha, i can't keep up with that stuff |
| 16:51.27 | frigidzephyr | lol |
| 16:51.35 | frigidzephyr | what version was it deprecated in? 1.8 ? |
| 16:52.09 | BenC[UK] | ok, set that now |
| 16:52.09 | BenC[UK] | thanks |
| 16:53.11 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 16:54.20 | leifmadsen | frigidzephyr: 1.6.x -- look at the CHANGES and UPGRADE.txt |
| 16:54.32 | leifmadsen | frigidzephyr: that stuff does get documented funny enough :) |
| 16:54.37 | *** join/#asterisk riscphree (~riscphree@h105.0.91.75.dynamic.ip.windstream.net) |
| 16:54.47 | frigidzephyr | ahh, yeah then in 1.8.X changes I also see the option "busylevel" |
| 16:57.47 | *** join/#asterisk Faithful (~Faithful@180.194.2.201) |
| 17:06.40 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 17:07.04 | ronndonn | if a sip phone rings but cant respond when picked up, is this an issue with the config of the device, [TK]D-Fender: was helping me out earlier and I'm still struggling to figure out the problem |
| 17:07.42 | pigpen | ronndonn, sounds like a nat issue. |
| 17:10.56 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:11.21 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 17:11.40 | pbxec | got NVFaxDetct compiled |
| 17:11.52 | pbxec | and showing on CLI |
| 17:14.36 | *** join/#asterisk jeff (~jeff@unaffiliated/jeff) |
| 17:17.21 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 17:17.36 | ronndonn | pigpen: bit of a noob, where do i start |
| 17:22.14 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:22.37 | *** join/#asterisk yoruk (~yoruk@host7-181-dynamic.52-82-r.retail.telecomitalia.it) |
| 17:29.15 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 17:30.22 | keith4 | ~sipnat |
| 17:30.22 | infobot | hmm... sipnat is Quick guide on configuring Asterisk + 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 |
| 17:30.32 | keith4 | ~nat |
| 17:30.32 | infobot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 17:35.46 | citywok | is there an easy way to tell if the QoS values are being set on packets coming out of asterisk? |
| 17:36.26 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:37.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 17:37.55 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 17:39.57 | stoffell | citywok, tcpdump is your best bet i guess |
| 17:40.14 | citywok | yea, i just did that actually and it appears as though * is NOT setting the ToS bits. 10:39:24.018173 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 200) |
| 17:41.09 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 17:43.48 | *** join/#asterisk datalay (~datalay@85.105.107.83) |
| 17:43.54 | citywok | lol, if i set tos_sip, tos_audio in [genera] of sip.con it works, but those settings on peers in realtime* don't actually work. argh! |
| 17:43.58 | datalay | No compatible codecs, not accepting this offer, how can i solve this, im using g729 |
| 17:44.01 | datalay | how can solve this :( |
| 17:44.26 | [TK]D-Fender | datalay: You aren't offering a compatible choice. So go see what each side want s and make them happy |
| 17:45.09 | datalay | dear sir, if you advise me something for solution i ll be happy |
| 17:45.10 | datalay | thanks |
| 17:48.29 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
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| 18:01.20 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
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| 18:26.16 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:26.16 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:26.17 | *** join/#asterisk ccesario (~ccesario@189-29-53-173-ac.cpe.vivax.com.br) |
| 18:26.28 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 18:26.35 | dandate2 | how do i reload asterisk without disconnecting all active calls? |
| 18:26.44 | citywok | what version? |
| 18:26.45 | dandate2 | peticularly so my new music on hold song will start playing |
| 18:26.49 | dandate2 | 1.4 |
| 18:26.55 | citywok | "reload" |
| 18:27.35 | dandate2 | just reload? not "asterisk reload" , u positive that wont disconnect the 20 people listening to music right now heh |
| 18:27.53 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 18:28.04 | citywok | well, that reloads the config. if your config is screwed up and somehow * crashes, then it might disconnect everything :P |
| 18:28.27 | citywok | in my call center our system reloads itself every 20 minutes all day long applying new changes to queues ,etc. |
| 18:28.38 | dandate2 | k |
| 18:29.17 | *** join/#asterisk dzup2 (~alex@unaffiliated/dzup2) |
| 18:29.41 | dandate2 | righteous it worked |
| 18:30.10 | leifmadsen | trust no one on this channel :) |
| 18:30.24 | leifmadsen | <-- especially that guy |
| 18:31.46 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:32.04 | *** join/#asterisk tbenson (~tbenson@c-67-174-228-93.hsd1.ca.comcast.net) |
| 18:34.11 | *** join/#asterisk deonv (~adium@196.1.28.226) |
| 18:36.03 | datalay | how can see status of my outbound trunk |
| 18:36.07 | datalay | with rasterix |
| 18:37.14 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:37.28 | p3nguin | bash: rasterix: command not found |
| 18:38.44 | datalay | rasterisk |
| 18:39.07 | datalay | iam calling with x-lite but i cant see anything in tail -f /var/log/asterisk/full |
| 18:39.25 | datalay | i want to see what s wrong for calling |
| 18:40.14 | tbenson | Anyone have a good production 1.6 using Bridge? I am trying to see if there is a big performance difference between bridge and meetme |
| 18:40.21 | [TK]D-Fender | datalay: "asterisk -rvvvvvvvvvvvvvvvvvvv" |
| 18:40.27 | [TK]D-Fender | datalay: and enable SIP DEBUG |
| 18:41.10 | datalay | how can i enable SIP DEBUG my sir |
| 18:41.25 | [TK]D-Fender | datalay: "help sip" <- go look at the directions for your version |
| 18:42.09 | datalay | help sip |
| 18:42.55 | datalay | there s no command like help sip |
| 18:43.06 | [TK]D-Fender | datalay: SHOW US |
| 18:43.08 | [TK]D-Fender | ~pb |
| 18:43.08 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 18:43.10 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:43.47 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 18:44.36 | datalay | dhcppc3*CLI> core show version |
| 18:44.37 | datalay | Asterisk 1.6.2.10 |
| 18:44.38 | *** join/#asterisk baddemanax (~baddemana@host-85-27-42-254.brutele.be) |
| 18:44.42 | datalay | sip debug on |
| 18:44.46 | datalay | not working for me :( |
| 18:44.50 | datalay | or sip debug |
| 18:45.02 | [TK]D-Fender | datalay: PASTEBIN your attempts including the one showing the instructions |
| 18:45.17 | datalay | okay |
| 18:46.09 | datalay | http://pastebin.com/n1TxtQQa |
| 18:46.15 | drudge` | good thing docotrs dont require pastebins before they take a look |
| 18:46.36 | Naikrovek | one day they will |
| 18:46.37 | [TK]D-Fender | datalay: You did NOT do the command I showed you |
| 18:46.49 | datalay | sorry iam trying agaion |
| 18:46.57 | datalay | asterisk -rvvvvvvvvvvvvvvvvvvv |
| 18:47.02 | drudge` | doc, my shoulder stopped working yesterday. doc: prove it. pb it. |
| 18:47.36 | Naikrovek | one day we'll have little measurement equipment measuring everything, ready for doctor visit. doctor will review data and say "oh, i see you had H1N1 there 3 months or so ago." |
| 18:48.02 | drudge` | do they make clamav for hoomans yet |
| 18:48.03 | datalay | http://pastebin.com/3bfNH9a8 |
| 18:48.04 | Naikrovek | "and i can see you have some loss of feeling in your shoulder.. okay." |
| 18:48.06 | drudge` | defeat the bad virii |
| 18:48.11 | datalay | [TK]D-Fender can u check it agaion please |
| 18:48.27 | Naikrovek | sip set debug on |
| 18:48.39 | Naikrovek | actually |
| 18:48.50 | Naikrovek | sip set debug <ip address or peer name> |
| 18:49.01 | *** join/#asterisk baddemanax (~baddemana@host-85-27-42-254.brutele.be) |
| 18:49.08 | Naikrovek | ahh no such command sip... weird |
| 18:51.11 | datalay | there s no command |
| 18:51.12 | datalay | sip |
| 18:51.15 | datalay | core set debug |
| 18:51.18 | datalay | something |
| 18:53.51 | tbenson | Is anyone using Bridge() in production with more then 20 conference rooms? |
| 18:55.48 | [TK]D-Fender | tbenson: Bridge is not meant for "conferencing". Its a 1-shot hook |
| 18:56.00 | datalay | [TK]D-Fender why are u saying wrong commands |
| 18:56.05 | datalay | there s no command like: sip debug on |
| 18:56.13 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 18:56.22 | [TK]D-Fender | [14:55]<datalay>there s no command like: sip debug on<--- I did not tell you to do this |
| 18:56.28 | drudge` | lol |
| 18:56.38 | citywok | sip ? |
| 18:56.46 | citywok | it will tell you what the available commands are |
| 18:57.03 | *** part/#asterisk tbenson (~tbenson@c-67-174-228-93.hsd1.ca.comcast.net) |
| 18:57.07 | [TK]D-Fender | datalay: Does "sip show peers" give you anything? |
| 18:57.14 | *** join/#asterisk tbenson (~tbenson@c-67-174-228-93.hsd1.ca.comcast.net) |
| 18:57.30 | tbenson | hmm, bounced out before I read whatever you sed. |
| 18:57.32 | tbenson | said |
| 18:57.39 | [TK]D-Fender | [14:55]<[TK]D-Fender>tbenson: Bridge is not meant for "conferencing". Its a 1-shot hook |
| 18:58.12 | datalay | No such command 'sip show peers' |
| 18:58.23 | citywok | lolol |
| 18:58.30 | citywok | module load chan_sip.so |
| 18:58.37 | [TK]D-Fender | datalay: pastebin "ls -la /etc/asterisk |
| 18:58.58 | tbenson | [TK]D-Fender: sorry meant ConfBridge. |
| 19:00.08 | citywok | apparently he didn't want that much help |
| 19:00.36 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:06.24 | Naikrovek | or, he figured it out |
| 19:13.53 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 19:15.09 | ccomp5950 | either that or he thinks "ls -la" is one of those prank commands like "rm -rf" |
| 19:18.30 | *** join/#asterisk breardo (~no@234-200-29-134.hcc.mnscu.edu) |
| 19:19.20 | breardo | so.. I have a digium TE412P (w/ onboard EC) and users are complaining of echo.. from what I can ascertain, I do NOT not need to be adding anything to chan_dahdi.conf or system.conf to enable this HWEC...yet it doesnt seem to be working, so I'm clearly wrong... |
| 19:19.22 | breardo | any advice? |
| 19:19.40 | breardo | do I need to set 'echocanceller=1-12,mg2' (or whatever the syntax is) in system.conf ? |
| 19:20.15 | *** join/#asterisk Faithful (~Faithful@180.194.1.217) |
| 19:20.25 | WIMPy | Check your dmesg after starting dahdi. |
| 19:20.42 | WIMPy | Or look if the EC module makes a big lightshow. |
| 19:25.03 | breardo | light show? |
| 19:25.08 | breardo | sorry I dont follow you :) |
| 19:25.13 | breardo | I will check that dmesg.. thanks :) |
| 19:28.13 | [TK]D-Fender | ccomp5950: Do you know why I asked? |
| 19:28.41 | ccomp5950 | to see if he even has asterisk installed, I would hazard to guess. |
| 19:29.07 | ccomp5950 | or just to see if he is actually trying. |
| 19:29.07 | citywok | to see if he configured anything at all, ever? |
| 19:29.13 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 19:29.22 | citywok | check if all the file timestamps are the samples lol |
| 19:29.24 | [TK]D-Fender | ccomp5950: No, he showed that much. I doubt he install the samples, had a modules.conf at all because chan_sip didn't load. Or perhaps he screwd his permissions on the files. |
| 19:29.47 | [TK]D-Fender | ccomp5950: He may have left from embarrassment realizing his error, etc... |
| 19:30.03 | [TK]D-Fender | ccomp5950: Or just quit out of frustration |
| 19:30.07 | [TK]D-Fender | qho knows |
| 19:30.09 | [TK]D-Fender | who |
| 19:30.23 | citywok | lol, either way the world is a better place now :P & by world i mean #asterisk |
| 19:31.03 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 19:34.14 | ccomp5950 | makes sense. |
| 19:34.36 | ccomp5950 | I must admit I'm not on the up and up on the diagnostic process here, thanks for explaining the reasoning. |
| 19:35.38 | [TK]D-Fender | ccomp5950: Step 1 to seeing if your car works isn't putting the key in the ignition, its verifying that the car even exists.... |
| 19:35.53 | *** join/#asterisk Indi0 (~bt4@200110174149.ip23.static.mediacommerce.com.co) |
| 19:35.54 | citywok | or making sure it has all the parts installed... tires are important. |
| 19:35.54 | drudge` | lol |
| 19:36.03 | [TK]D-Fender | "But the key fell through open air!" |
| 19:36.14 | [TK]D-Fender | (thump) |
| 19:36.20 | carrar | Didn't wonderwoman have a jet like that? |
| 19:36.51 | carrar | still trying to find that damn island |
| 19:37.07 | carrar | 'Paradise Island' |
| 19:37.37 | drudge` | i know where paradise city is |
| 19:38.03 | [TK]D-Fender | Take me down.... |
| 19:38.08 | [TK]D-Fender | rocks out |
| 19:38.25 | [TK]D-Fender | If I knew any more hair-metal songs I should buy stock in Revlon. |
| 19:39.16 | *** part/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 19:44.25 | *** join/#asterisk p4tr0p1 (~jam@201.47.74.147) |
| 19:45.46 | p4tr0p1 | Got SIP response 500 "Previous INVITE still in progress" back from XXX.XXX.XXX.XXX |
| 19:45.49 | p4tr0p1 | anybody? |
| 19:47.51 | [TK]D-Fender | p4tr0p1: I'm sorry, you forgot to phrase that in the form of a question </trebek> |
| 19:48.09 | p4tr0p1 | hehehe |
| 19:48.11 | p4tr0p1 | sorry |
| 19:51.41 | p4tr0p1 | i have an 1.4.3X asterisk box and when one of my SIP peers receive a call, it lengths about 10 min and hangs up |
| 19:52.04 | *** join/#asterisk Iamnach0 (~Iamnacho@ip174-70-139-208.ks.ks.cox.net) |
| 19:52.40 | p4tr0p1 | CLI returns that response |
| 19:52.57 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 19:53.17 | [TK]D-Fender | p4tr0p1: Then perhaps you should do a SIP trace on the complete call |
| 19:54.08 | *** join/#asterisk Brack10 (~tbrackett@unaffiliated/brack10) |
| 19:54.12 | Brack10 | Hey |
| 19:54.29 | Brack10 | so I have to unhook my local loop from the phone company to use asterisk to handle analog phone calls right? |
| 19:54.41 | *** join/#asterisk mducharme (~nothing@S0106000e0cc03cff.wp.shawcable.net) |
| 19:54.58 | [TK]D-Fender | Brack10: no |
| 19:55.02 | *** join/#asterisk sekil (~Ognjen@78.24.104.79) |
| 19:56.36 | Brack10 | won't the dialtones compete? |
| 19:56.46 | [TK]D-Fender | Brack10: With what? |
| 19:56.49 | Brack10 | each other |
| 19:56.58 | [TK]D-Fender | Brack10: Takes 2 (or more) to compete... |
| 19:57.01 | Brack10 | I've got asterisk providing dialtone and the phone company providing one too |
| 19:57.14 | p3nguin | You won't be providing a dialtone over SIP. |
| 19:57.29 | Brack10 | I'm talking about hooking up my existing analog phones to asterisk |
| 19:57.33 | Brack10 | in my house |
| 19:57.35 | p3nguin | And you won't be providing dialtone over your FXO port either. |
| 19:57.43 | [TK]D-Fender | Brack10: First Astisk is software. You cant plug a copper wire into software. Next there has to be 2 DEVICES on the line. Why would you do this? What would they be? Why would they be used concurrently? |
| 19:57.45 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.93) |
| 19:58.25 | Brack10 | ok I'm aware that I need an analog phone card to accomplish this |
| 19:58.37 | p4tr0p1 | [TK]D-Fender, I'll do a trace again, and back with the results |
| 19:58.45 | Brack10 | Right now my local loop goes into the phone company which provides dialtone |
| 19:59.16 | p3nguin | And that's the end of it. You won't be adding more dialtone on that line. |
| 19:59.36 | Brack10 | Yeah but I want my analog phones to use asterisk to dial out instead of the phone company |
| 20:00.09 | p3nguin | You're not going to continue using the phone company to make calls? |
| 20:00.30 | Brack10 | Right |
| 20:00.37 | p3nguin | How will you make calls? |
| 20:00.44 | Brack10 | Asterisk over SIP |
| 20:00.48 | p3nguin | to what? |
| 20:01.04 | [TK]D-Fender | Brack10: If you want to put your whole house's wiring onto a single FXS interface so it goes through * then yes you would split at your demac. FXo > Telco, hose wiring > FXS |
| 20:01.05 | p3nguin | to an ITSP? |
| 20:01.07 | Brack10 | the PSTN via a sip provider's gateway |
| 20:01.08 | citywok | why don't ITSP's set the QoS bits on calls? I imagnie they'd be ignored by the vast majority of the world but it couldn't hurt, right? |
| 20:01.21 | Brack10 | ITSP, sure |
| 20:01.34 | Brack10 | citywok: they get stripped out by ISPs |
| 20:01.56 | p3nguin | As [tk]d-fender mentioned, you'll have to disconnect your telco from your house at the NID. |
| 20:02.17 | Brack10 | ok |
| 20:02.32 | p3nguin | Vonage probably has a video on how to do it. |
| 20:02.56 | Brack10 | I have a pretty small apartment, I might just get a cordless phone |
| 20:02.57 | Brack10 | :) |
| 20:03.14 | p3nguin | That could work, too. |
| 20:03.26 | Brack10 | too bad IP cordless phones cost like 200 dollars |
| 20:03.51 | p3nguin | $35 for a PAP2 and $20 for a cordless phone. |
| 20:04.10 | Brack10 | Yeah |
| 20:04.15 | Brack10 | but that's dissatisfying |
| 20:04.22 | p3nguin | Use more duct tape. |
| 20:04.30 | Brack10 | Now there's an idea |
| 20:04.33 | *** join/#asterisk asphere_ (~ardavis@hologram.homeip.net) |
| 20:04.49 | Brack10 | oh wait I have an iphone |
| 20:05.37 | p3nguin | If you have wifi on your phone and an access point, you already have a very expensive cordless IP phone. |
| 20:05.53 | Brack10 | You're right |
| 20:06.06 | Kobaz | heh |
| 20:06.08 | p3nguin | Now you just need the appropriate SIP app. |
| 20:06.18 | [TK]D-Fender | And you'll watch your battery get eaten up fast. Charge often |
| 20:06.28 | Brack10 | That's true |
| 20:06.43 | Kobaz | but the battery on a smartphone gets eaten up fast anyway |
| 20:06.49 | Kobaz | at least non-blackberrys |
| 20:06.54 | Brack10 | I already have a PAP2 |
| 20:07.04 | [TK]D-Fender | Kobaz: My new Galaxy S is doing good... |
| 20:07.07 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-qgbufobgflqehcya) |
| 20:07.16 | Kobaz | [TK]D-Fender: i got a droid... i get about 30 hours of battery life |
| 20:07.19 | [TK]D-Fender | Kobaz: Just need an H.323 video client for it... |
| 20:07.28 | Brack10 | Maybe a cordless phone wouldn't be such a bad idea just to save battery |
| 20:07.29 | Kobaz | [TK]D-Fender: my blackberry got like 7 days out of one charge |
| 20:07.37 | *** join/#asterisk rdircio (~rdircio@201.137.129.242) |
| 20:07.44 | [TK]D-Fender | Kobaz: And regular use? |
| 20:07.44 | rdircio | hi |
| 20:07.49 | Kobaz | [TK]D-Fender: regular use |
| 20:07.49 | Brack10 | haha my iphone gets like 12 hours with regular use |
| 20:08.01 | Brack10 | my old blackberry got about 24 hours with regular use |
| 20:08.04 | Kobaz | [TK]D-Fender: like, maybe one/two calls a day, minimal surfing, some games |
| 20:08.06 | [TK]D-Fender | Kobaz: Oh well.. probably because you weren't having as much fun with it ;) |
| 20:08.07 | Brack10 | 8330 which is slower than dirt |
| 20:08.22 | [TK]D-Fender | *b00m* |
| 20:08.25 | Brack10 | oh wow netsplit |
| 20:08.27 | Brack10 | that brings me back |
| 20:08.31 | rdircio | i'm having an issue, cannot make outbound calls through a dahdi device until after i receive a call |
| 20:08.32 | citywok | Brack10 i'm pretty sure my colo isn't stripping out the QoS bits |
| 20:08.35 | rdircio | have you seen that? |
| 20:08.42 | Kobaz | [TK]D-Fender: having the droid sit idle, is about 30 hours... if i do anything with it, you can knock off an hour of battery for every 10 minutes of use |
| 20:09.21 | Kobaz | and that's after major tweaking, turning off all kinds of stuff |
| 20:09.42 | Kobaz | i think there's still some stuff i can turn off, like this stupid uninstallable news and weather... although my phone is rooted so i can surely get rid of it |
| 20:09.52 | [TK]D-Fender | Kobaz: I haven't gotten there yet... I charge nightly anyway... which reminds me I need to get some more SUB>MiniUSB cables... |
| 20:10.03 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 20:10.15 | Kobaz | hah yeah. my mini usb's are all useless now... i went and ordered three microusb cables |
| 20:10.49 | Kobaz | just you wait, and we'll have to all switch again to picousb and then the following year nanousb... and then to wireless usb charging |
| 20:10.54 | citywok | i bought some of those adapters and they all broke within a few weeks, cheap pieces fo shit. buy the $1.39 cables off of amazon and call it good lol. |
| 20:11.05 | rdircio | :( |
| 20:11.11 | citywok | Kobaz i hope i dont have to buy a new cable for wireless charging :P |
| 20:11.15 | Kobaz | heh |
| 20:11.38 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 20:11.44 | Kobaz | i'm gonna pick up a solar charger too |
| 20:12.03 | [TK]D-Fender | Kobaz: Mean USB> MicroUSB |
| 20:12.19 | [TK]D-Fender | Kobaz: new standards and all |
| 20:12.27 | Kobaz | heh |
| 20:12.36 | Kobaz | at least they are standardizing now |
| 20:12.48 | Kobaz | there's a push for notebook power standards too |
| 20:13.18 | *** join/#asterisk Faithful (~Faithful@180.194.2.63) |
| 20:13.39 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 20:14.07 | Kobaz | oh |
| 20:14.12 | Kobaz | polycrum question |
| 20:14.25 | Kobaz | do you guys know how to keep the extension number displaying while you're on a call |
| 20:14.50 | Kobaz | or doing anything else for that matter... as soon as you're off the idle screen you lose your extension number display |
| 20:15.06 | [TK]D-Fender | Kobaz: No, thats just the way it is |
| 20:15.19 | [TK]D-Fender | Kobaz: you should know your buttons anyway.... |
| 20:15.34 | Kobaz | it would save me a lot of trouble of printing out labels to stick on every phone |
| 20:15.36 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 20:15.54 | citywok | people need to know their extension WHILE on the phone, but odnt already know it? wtf? |
| 20:16.09 | Kobaz | the problem is, someone sits at someone elses desk and is on the phone, and gets asked for an extension to reach... and they are like... umm... hmm... i don't know |
| 20:16.25 | Kobaz | so i printed out sticky labels for each phone, with the number on it |
| 20:16.27 | citywok | "the extension i'm talkign to you on right now" :P |
| 20:16.30 | Kobaz | yeah |
| 20:17.10 | Kobaz | the polycom is so configurable, i would be really surprised if you couldn't change some of the displays to show your exten |
| 20:17.42 | *** join/#asterisk bcrisp (~bcrisp@wsip-184-191-141-38.ph.ph.cox.net) |
| 20:17.51 | [TK]D-Fender | Kobaz: This isn't an "option" |
| 20:18.05 | [TK]D-Fender | Kobaz: Feel free to hack up your own ROM's for it if you feel like it.. |
| 20:18.13 | [TK]D-Fender | Kobaz: Or get a label maker... |
| 20:18.15 | Kobaz | yeah maybe not, probably need to modify the bitmaps for the display or something |
| 20:18.18 | citywok | not everything needs to be fixed via technology :P |
| 20:18.22 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:18.22 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:18.25 | bcrisp | have a strange situation: all of a sudden, when we use our out bound SIP provider to make calls out of *, we get a busy tone for only certain numbers |
| 20:18.27 | Kobaz | yeah the label maker is the current solution |
| 20:18.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:18.51 | Kobaz | citywok: it's just a hacky fix |
| 20:19.39 | citywok | bcrisp what's the SIP debug say? talk to your provider about it |
| 20:20.02 | bcrisp | it doesn't have any unusual output |
| 20:20.14 | bcrisp | our company has 2 voip setups |
| 20:20.17 | citywok | is the busy tone coming from the remote provider then? |
| 20:20.25 | bcrisp | yep |
| 20:20.31 | citywok | then you really should talk to them about it |
| 20:20.37 | bcrisp | i think we've had some odd network issues lately |
| 20:20.41 | bcrisp | yeah.. thanks will do |
| 20:20.46 | bcrisp | the problem |
| 20:20.48 | citywok | sounds like they are having issues if they are giving you the busy signal |
| 20:20.50 | bcrisp | is that it only happens when coming from * |
| 20:20.59 | bcrisp | if you dial from cell or other system it does ring through |
| 20:21.41 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 20:23.55 | bcrisp | bizarre |
| 20:24.56 | citywok | bcrisp that could be their upstream provider, i frequently have issues where my cellphone acts differnetly than my ITSP. |
| 20:25.23 | citywok | ITSP can't get their, but i can from my cell. submit ticket w/ provider giving SIP debug output, description of problem, time & phonenumber and they can figure it out / fix it. |
| 20:25.28 | [TK]D-Fender | checkout time, BBIAb |
| 20:26.48 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 20:29.24 | bcrisp | what is weird is this happening just after a major network attack |
| 20:30.05 | tbenson | anyone know if there is a performance difference between meetme() and confbridge()? |
| 20:30.28 | citywok | fail2ban is my new best friend |
| 20:33.51 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:33.51 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:34.20 | jsidhu | hey guys, i setup a Bandwidth.com trunk and am having a small issue. Everything works, but I dont hear a ring when I call their number... The destination rings, but the caller doesnt hear any ringing.. |
| 20:34.47 | citywok | every number you call? |
| 20:35.03 | citywok | check your SIP debug and see if you receive a 180 (or is it 183) RINGING response |
| 20:35.23 | citywok | I had a similar problem with a handful of numbers on flowroute and turns out i wasn't getting the RINGING response in SIP |
| 20:35.33 | jsidhu | i see a 180 |
| 20:35.38 | jsidhu | SIP/2.0 180 Ringing |
| 20:38.31 | p3nguin | You could fake it. |
| 20:38.39 | jsidhu | how |
| 20:38.51 | p3nguin | Add the 'r' option on your outbound Dial(). |
| 20:39.18 | p3nguin | core show application Dial |
| 20:40.08 | jsidhu | interesting, how would i go about getting this setp |
| 20:40.20 | p3nguin | <p3nguin> core show application Dial <--- clue |
| 20:42.23 | jsidhu | yeah thanks, exten = _+1555555555,n,Dial(Sip/700,30,r) |
| 20:42.27 | jsidhu | that would do it? |
| 20:42.39 | p3nguin | That's not an outbound route. |
| 20:42.55 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:42.59 | jsidhu | no, its not a problem outbound, its inbound. If i cann that number, i dont hear ringtone |
| 20:43.08 | jsidhu | unless im looking at it wrong |
| 20:43.08 | p3nguin | Well, I guess it could be. Forgive my presumption. |
| 20:43.29 | p3nguin | That's the correct way to fake ringing, though. |
| 20:44.29 | p3nguin | You mean if I pick up my phone and call your DID number, I won't hear ringing, but you will and you might answer? |
| 20:44.37 | jsidhu | right |
| 20:44.38 | jsidhu | yes |
| 20:44.40 | jsidhu | exactly |
| 20:45.08 | p3nguin | What do you have before that Dial() command in the extension? |
| 20:45.28 | jsidhu | exten = _+15555551212,1,Answer |
| 20:45.35 | p3nguin | Remove that. |
| 20:45.41 | jsidhu | o |
| 20:45.47 | p3nguin | and change the n in the next line to 1 |
| 20:46.34 | p3nguin | There's really no good reason to explicitly bring up the channel immediately before Dial()ing a phone. |
| 20:47.03 | p3nguin | If you were going to bring it up, do some other fun stuff, THEN Dial() the phone, that might be different. |
| 20:47.14 | jsidhu | oh ok, understoof |
| 20:47.17 | jsidhu | thank you, that worked |
| 20:47.32 | p3nguin | without the r or with it? |
| 20:47.51 | jsidhu | yes, without the r |
| 20:48.05 | p3nguin | Good. Always go without it when possible. People will laugh at you if you use it when you don't need to. |
| 20:48.14 | jsidhu | ok cool |
| 20:48.27 | mmlj4 | heh |
| 20:49.14 | mmlj4 | hrm? timtoady hangs out here? go, larry! |
| 20:49.20 | jsidhu | another question, my caller ids come with a +1, so +15555551212, and when looking at the list of missed calls, we cant dial directly, i guess i need to change the dialplan to allow for the +1? |
| 20:50.06 | p3nguin | ExecIf($["${CALLERID(num):0:2}" = "+1"]|Set|CALLERID(num)=${CALLERID(num):2}) |
| 20:50.12 | citywok | set the callerid and cut the + off of it |
| 20:50.40 | citywok | what he said :P |
| 20:50.49 | jsidhu | cool, thanks |
| 20:50.52 | p3nguin | I probably wouldn't change the extension matching for outbound -- I would modify the caller ID incoming. |
| 20:51.14 | citywok | agreed, one line is easier. |
| 20:51.39 | citywok | though you can always use a goto and cut the + and send it back in to the same logic as if it didn't have the + on it, but you'd also have to modify your phone to allow + dialing |
| 20:52.02 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 20:52.02 | citywok | thats how i handle 12535551212, 2535551212, 5551212, i just use Goto() and use one set of logic |
| 20:53.02 | *** join/#asterisk talntid (~talntid@c-76-104-157-191.hsd1.wa.comcast.net) |
| 20:53.08 | jsidhu | this should be done *before the dial command right? |
| 20:53.12 | p3nguin | right |
| 20:53.25 | talntid | Can't seem to find any information... anyone have a way to make it so when a certain number calls incoming, i can play them a message automatically? |
| 20:53.41 | talntid | if ($incomingnumber == XXXXXXXXXX) { do this } kind of thing.... |
| 20:53.54 | p3nguin | caller id matching |
| 20:53.56 | citywok | exten => 12535551212,1,Playback(you-suck) |
| 20:54.07 | p3nguin | That would play for ALL. |
| 20:54.37 | citywok | oh then you need to use an if and check the callerid :P -- i was thinking all calls to a certain number, not from a certain number |
| 20:54.43 | leifmadsen | exten => 4165551234/9055551212,1,... |
| 20:55.10 | citywok | wait, leifmadsen is that legit? is the 905551212 the callerid to be checked? |
| 20:55.18 | mmlj4 | hrm? that's valid syntax now? the second number being the outgoing CID? |
| 20:55.19 | leifmadsen | citywok: yes, and it is legit |
| 20:55.25 | leifmadsen | that's been valid syntax for a long time |
| 20:55.28 | citywok | holy crap i never knew that, thank you for that trick! |
| 20:55.30 | p3nguin | That's incoming cid. |
| 20:55.34 | leifmadsen | that's for incoming CID matching |
| 20:55.36 | leifmadsen | not outgoing |
| 20:55.43 | talntid | i am looking for incoming. |
| 20:55.49 | citywok | gotcha, yea it makes sense i just didnt' know you could do that. |
| 20:55.52 | mmlj4 | ok, that makes less sense |
| 20:56.04 | citywok | so many little tricks that you can do that i dont know about lol. |
| 20:56.16 | WIMPy | BTW: Does exten/ work again to match anonymous calls? |
| 20:56.42 | WIMPy | hasn't tried for ages. |
| 20:58.16 | talntid | hmm... does not appear to work for me.. |
| 20:58.23 | citywok | leif on processing does it process the exten/callerid,1 before exten,1 ? |
| 20:58.33 | leifmadsen | yes |
| 20:58.37 | leifmadsen | it is more specific |
| 20:58.38 | p3nguin | talntid: http://pastebin.com/7VAhWUMg |
| 20:58.45 | talntid | exten => numberbeingcalled/numbercallingfrom,1,Playback(hello-world) |
| 20:59.04 | citywok | slick, gracias senor. i figured it would but figured i'd ask. thanks |
| 20:59.39 | jsidhu | application 'ExecIf' for extension |
| 20:59.55 | jsidhu | No Application, is there a module which needs to be enabled? |
| 21:01.12 | p3nguin | My example is kind of crude, but it should work. |
| 21:01.19 | jsidhu | maybe the spaces in it |
| 21:01.22 | jsidhu | let me look again |
| 21:01.30 | citywok | yup, it gives you the idea of how it works |
| 21:01.56 | p3nguin | The ExecIf command I put above was EXACT. |
| 21:06.48 | p3nguin | jsidhu: http://pastebin.com/Bq1mccPe |
| 21:06.51 | talntid | beautiful. |
| 21:06.57 | talntid | i have successfully implemented it. |
| 21:07.06 | talntid | thanks leifmadsen =] |
| 21:07.44 | citywok | core show application execif |
| 21:07.56 | thehar | ExecIf = lovely |
| 21:08.01 | citywok | mine gives not available for al the info, but it does tell me the command exists |
| 21:08.05 | citywok | all* |
| 21:08.17 | jsidhu | thanks penguin, appreciate it |
| 21:08.37 | citywok | penguin in 1.6 is says to use ?Application(params) after the execif[] |
| 21:08.41 | citywok | http://www.voip-info.org/wiki/view/Asterisk+cmd+ExecIf |
| 21:09.43 | jsidhu | p3nguin: pbx_extension_helper: No application 'ExecIf' for extension |
| 21:10.03 | jsidhu | im on a OpenWRT box, so perhaps theres a module that needs to be enabled? |
| 21:10.03 | p3nguin | jsidhu: Did you use it just like I put in the pastebin? |
| 21:10.07 | jsidhu | yes i did |
| 21:10.07 | p3nguin | oh |
| 21:10.16 | *** join/#asterisk Faithful (~Faithful@180.194.0.9) |
| 21:10.37 | jsidhu | honestly, my callerid is always going to be that format, so i dont really need the ExecIF |
| 21:10.46 | talntid | leifmadsen, maybe you also know... instead of exten => XXXxxxXXXX/numbertomatch,.... |
| 21:11.01 | talntid | can Instead of numbertomatch, can i make a variable with multiple numbers? |
| 21:11.21 | WIMPy | DrawÉ |
| 21:11.21 | p3nguin | Alright, then Just change it. Set(CALLERID(num)=${CALLERID(num):2}) |
| 21:11.25 | jsidhu | p3nguin: ,1,Set(CALLERID(num)=${CALLERID(num):1:12}) |
| 21:11.30 | jsidhu | yeah that works for me |
| 21:11.47 | WIMPy | Ooops. Sorry. |
| 21:11.52 | p3nguin | I use the ExecIf because of having it change at times. |
| 21:12.07 | p3nguin | Using ${EXTEN:0:2} would get the first two numbers of ${EXTEN}... is there a way to get just the LAST two numbers from ${EXTEN} ? |
| 21:12.25 | WIMPy | p3nguin: -2 |
| 21:13.02 | citywok | http://www.voip-info.org/wiki/view/Asterisk+variables#Substrings |
| 21:13.11 | leifmadsen | talntid: no, in that case you should build some dialplan to utilize ${CALLERID(num)} and perhaps a loop |
| 21:13.19 | citywok | WIMPy is correct, ${exten:-2} |
| 21:13.23 | leifmadsen | then you can check all the numbers in whatever list you have |
| 21:13.40 | talntid | roger that. :) |
| 21:13.58 | talntid | i figured that solution, but didn't know if you had a genius built-in way of doing it :) |
| 21:14.48 | p3nguin | talntid: If you're trying to build a blacklist, there's a prescribed way to handle it already. |
| 21:15.17 | bmoraca_work | astdb could do it pretty easily |
| 21:15.32 | talntid | i am, p3nguin. :) |
| 21:15.38 | p3nguin | BLACKLIST() |
| 21:15.51 | p3nguin | GotoIf($[${BLACKLIST()}]?misc,blocked,1) |
| 21:16.53 | p3nguin | You just need to add your numbers to the DB. |
| 21:18.52 | p3nguin | I set the last calling number for each incoming call, then use it to blacklist the last caller with this dialplan: http://pastebin.com/ipPdse09 |
| 21:28.15 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:29.18 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 21:29.28 | *** join/#asterisk f0ner00t (~f0ner00t@99-90-61-14.lightspeed.frokca.sbcglobal.net) |
| 21:33.43 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 21:33.53 | *** part/#asterisk f0ner00t (~f0ner00t@99-90-61-14.lightspeed.frokca.sbcglobal.net) |
| 21:43.35 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 22:05.13 | talntid | not finding a command to write to a file... using the asterisk commands... should I use system() ? |
| 22:07.55 | leifmadsen | talntid: try STAT() |
| 22:08.20 | leifmadsen | I can't remember if that is used for writing, or just checking permissions thought |
| 22:08.23 | leifmadsen | though* |
| 22:08.36 | leifmadsen | ah just for checking perms |
| 22:09.00 | leifmadsen | core show function FILE |
| 22:09.03 | leifmadsen | I think that might be what you want |
| 22:22.37 | talntid | checking |
| 22:23.04 | talntid | using too old of a version of *... |
| 22:23.07 | *** join/#asterisk dan__t (~dant@vpn.withparity.net) |
| 22:23.11 | dan__t | 'afternoon. |
| 22:23.24 | talntid | Asterisk 1.4.17~dfsg-2ubuntu1 |
| 22:23.34 | talntid | scared to upgrade because it can break all my zaptel stuff. |
| 22:24.44 | dan__t | Kind of related, maybe a bit OT, but I was hoping someone was familiar with one- or two-party recorded calls. As a service provider are you obligated to disclose that the call may be recorded to the called party, or is it the calling party's responsibility to inform the called party? Can a provider that records a call on behalf of the calling party have any liability for the recording of that call, if any? |
| 22:25.22 | talntid | first, we are not attorneys... and this varies from state to state... |
| 22:25.24 | WIMPy | What country? |
| 22:25.54 | talntid | so, although I am not going to give the answer, because I don't know... please be advised that anyone who DOES give an answer, may give me a wrong answer... or one that applies to THEM.. |
| 22:26.15 | dan__t | I know it varies from state to state. I've read up on it. However, nothing that I've read addresses a provider's role in the matter. |
| 22:26.19 | bmoraca_work | dan__t: in general, in the US, BOTH parties need to be made aware that a call will be recorded. |
| 22:26.29 | citywok | dan_t check which state, in most states it's two party, some states are one. |
| 22:26.45 | dan__t | Federal law dictates that AT LEAST the calling party is informed. 16 states mandate that both the calling and the called party are informed. |
| 22:26.51 | citywok | i used to have a list of two party vs one party states in a text file, broken down by area code per state but i dont know where it went |
| 22:26.54 | talntid | citywok, but phone providers record for "debugging" reasons, and they don't inform... |
| 22:26.57 | dan__t | I've done my homework. I promise. Again, nothing addresses a provider's role/liability. |
| 22:26.58 | Defraz | For example Idaho is one but, once of the two parties needs to say it will be recorded. |
| 22:27.45 | citywok | talntid none of my providers record calls, working with Qwest, Bandwidth.com and Flowroute. If you believe they are recording... please explain why. |
| 22:27.47 | Defraz | if you record it then record it at the request of the yoru client but they should be the one to say it is being recorded. It is an option of the service you give them. |
| 22:28.08 | talntid | citywok, mine does.. when I call them with an issue, they pull recordings to investigate the issue... |
| 22:28.22 | talntid | Accessline Communications |
| 22:28.25 | citywok | hrmmm that seems like a problem to me. lol. |
| 22:28.27 | dan__t | Right. "should" be the one to say it is being recorded. Now, as a provider, must you play some automated audio when the call connects to disclose this etc etc. |
| 22:28.37 | talntid | also... *waves to Defraz* hi neighbor! |
| 22:29.03 | citywok | are you sure they pull up the actual audio from the call, or just the SIP logging? |
| 22:29.17 | citywok | my providers all log all of the SIP traffic, but not the RTP/media streams. |
| 22:29.18 | talntid | they have played recordings |
| 22:29.35 | Defraz | waves back! |
| 22:29.52 | citywok | wow, they must not have many customers, or they have wasted a ton of resources on recording all those calls. lol |
| 22:30.19 | citywok | call recording at an ITSP level can't possibly be cheap |
| 22:30.29 | talntid | yeah, i connect a -lot- of calls daily... |
| 22:30.33 | talntid | 25kish.. |
| 22:31.16 | citywok | yea lol, i only do 5k, recording my 5k is very easy but to record all of their customers hundreds of thousands if not millions of calls per day, and encode/store all the recordings... |
| 22:31.26 | talntid | indeed |
| 22:31.33 | talntid | i record my 25k and even that's easy |
| 22:31.46 | citywok | but i'm pretty sure they shouldn't be doing that. remember the verizon wiretapping government stuff? lol |
| 22:31.49 | talntid | but i agree, if they are recording all, they are in for a problem.. :) |
| 22:31.50 | *** join/#asterisk MuskyHusky (Smegma@unaffiliated/coil) |
| 22:31.51 | MuskyHusky | [Oct 7 22:30:08] NOTICE[24160]: chan_sip.c:20063 handle_request_invite: Call from '' to extension 's' rejected because extension not found. |
| 22:32.09 | dan__t | Alright, well, guess I've got more research to do. |
| 22:32.13 | dan__t | Thanks for your help. |
| 22:32.18 | talntid | dan__t, again.. |
| 22:32.20 | talntid | lawyer ;) |
| 22:32.23 | dan__t | Yep. |
| 22:32.35 | dan__t | Figured this would be a good starting point for information, regardless. |
| 22:33.03 | MuskyHusky | im trying to recieve calls from my sip2sip.info account |
| 22:33.18 | citywok | i'd say your status as provider doesnt alter the fact that one/both parties need to be informed by state law. |
| 22:33.32 | dan__t | right. |
| 22:35.06 | bmoraca_work | i'm having a hard time thinking of a reason why a provider would be recording calls |
| 22:35.14 | bmoraca_work | (of customers) |
| 22:35.53 | dan__t | Because.... as a provider.. they asked me to? |
| 22:35.54 | bmoraca_work | the facility should be there to record if required by law (with accompanying warrant), but providers should not otherwise be listening in to customer calls |
| 22:36.22 | dan__t | bmoraca_work, I know about the "in general" laws that you mentioned earlier. I was inquiring about the specifics. But, as I've told talntid, I'm pretty sure its just lawyer time. |
| 22:37.21 | talntid | nods |
| 22:38.13 | bmoraca_work | dan__t: if a customer has asked you to record all calls, that customer needs to make its callers aware that their calls may be recorded |
| 22:39.18 | WIMPy | wouldn't offer to record generally. Only on a per call basis. That way you can be sure, your custormer is away of the fact ant that he needs to inform the other party. |
| 22:39.51 | dan__t | While I appreciate your input, it doesn't sound definitive. Lawyer time. heh. |
| 22:40.42 | dan__t | But when I learn, I'll pop back in here and let you guys know, if anyone cares. |
| 22:40.53 | bmoraca_work | dan__t: whether the call is being recorded by the client or by you (the provider), if anyone is going to be accessing it, callers need to be made aware that the call is being recorded |
| 22:41.37 | bmoraca_work | having the service provider record for you to avoid having to tell customers is a loophole that the FCC isn't stupid enough to have left open |
| 22:41.41 | dan__t | According to federal law, only one party needs to be implicitly informed. The laws of 16 states, however, mandate that both parties need to know heh. |
| 22:42.14 | dan__t | I don't know how else to explain this. You seem to think "a provider" would do it for malicious purposes. That's not the case. So... |
| 22:42.34 | bmoraca_work | a provider could do it for malicious purposes. |
| 22:42.46 | dan__t | Like the whole "to avoid having to tell customers" thing is, well, not even part of the equation. You made that up. THat's not what I'm getting at. |
| 22:42.49 | bmoraca_work | that's the rub. "would" doesn't matter. "could" is all that matters. |
| 22:42.56 | dan__t | Sure. |
| 22:43.08 | dan__t | Ok. Well thanks again for the time. I'll chime back in when I learn for sure. |
| 22:43.29 | bmoraca_work | dan__t: yours, and your customer's, intentions may be good. the laws, though, need to be written to accomodate people who aren't so scrupulous. |
| 22:43.43 | dan__t | (again: lawyer time) |
| 22:51.29 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 22:52.22 | talntid | dan |
| 22:52.30 | talntid | mind shooting me an email when you find out? |
| 22:52.39 | talntid | eric@spokanepcrepair.com -- i'm curious |
| 22:53.10 | citywok | spokane wa? ewwwww |
| 22:53.16 | talntid | yeah :( |
| 22:53.23 | citywok | sux2beu |
| 22:53.27 | talntid | lol yup |
| 22:53.36 | citywok | i didnt even know htey had internet out there |
| 22:53.50 | p3nguin | muskyhusky: They obviously are not sending calls to extension 's' on your system. |
| 22:53.54 | citywok | i guess it's only fair they have it in spokane if we were able to have a call center in Colville. |
| 22:54.58 | talntid | colville?!!? |
| 22:55.07 | talntid | now you are not telling the truth. |
| 22:55.32 | citywok | lolol |
| 22:56.02 | talntid | what could possibly be in colville? |
| 22:56.08 | drmessano | spams eric |
| 22:56.08 | p3nguin | nothing |
| 22:56.16 | MuskyHusky | p3nguin, its the same sip provider trying to route it through asterisk |
| 22:56.56 | p3nguin | muskyhusky: The same as what? |
| 22:57.06 | drmessano | Colville is a swamp, in the middle of a swamp, surrounded by swamp.. in the swamp area of the state |
| 22:57.10 | MuskyHusky | the person calling me is using the same sip provider as i am |
| 22:57.12 | citywok | talntid, not a whole lot. there's a roundabout and a stop light. also "the mall" (walmart) |
| 22:57.29 | drmessano | Last time I was in Colville, I didn't even know it |
| 22:57.30 | p3nguin | muskyhusky: What you showed says that the call is being send to the s extension. But it says that you don't have an s extension. |
| 22:57.43 | MuskyHusky | yeah whats that mean |
| 22:57.54 | MuskyHusky | i need to make a 's' extension? |
| 22:57.54 | citywok | i wonder how many asterisk users there are in Redmond/Seattle. we should have an asterisk drinking group. lol |
| 22:58.03 | talntid | and there was a... call center there? |
| 22:58.12 | talntid | this facinates me. |
| 22:58.37 | p3nguin | muskyhusky: Typically you're going to tell the ITSP what extension to send calls to, and it is rarely going to be the s extension. Then you would configure the real extension to accept calls. |
| 22:58.43 | citywok | yea we had a small call center that had as many as 70 people at one point |
| 22:58.45 | talntid | did they hook horses to a roll of copper wire, to generate electricity for the call center? |
| 22:58.56 | talntid | wow, what did you guys do? |
| 22:59.08 | talntid | i'm currently expanding mine to 70... |
| 22:59.10 | citywok | hah, close. every couple months some jackass with a backhoe somewehre between spokane and colville would cut the fibre. god that was annoying. |
| 22:59.28 | citywok | www.csgchannels.com |
| 23:00.23 | talntid | but started in colville.... huh.... |
| 23:00.32 | talntid | maybe there is hope for my call center.. lol ;) |
| 23:00.38 | citywok | no, started in seattle area, we did have a location in colville though |
| 23:00.53 | talntid | ah |
| 23:02.04 | talntid | with the 70-seat call center, was the IT load a one-person job? |
| 23:02.41 | citywok | all the core equipment was in redmond, so yes never more than 1 IT person in cv |
| 23:02.45 | *** part/#asterisk sekil (~Ognjen@78.24.104.79) |
| 23:02.58 | talntid | ah, i see. |
| 23:03.16 | talntid | I use LTSP, and the core equip is onsite.. |
| 23:03.24 | talntid | LTSP doesn't like going over distances ;) |
| 23:03.31 | citywok | what's LTSP? lol |
| 23:03.37 | MuskyHusky | p3nguin, i try and call my friend and i get this ARNING[24160]: chan_sip.c:17791 handle_response_invite: Received response: "Forbidden" from '"Anonymous" <sip:Anonymous@anonymous.invalid>;tag=as3 |
| 23:04.48 | talntid | thin client environment |
| 23:04.50 | drmessano | Lightweight Telephony Shuttling Pidgeons? |
| 23:04.56 | talntid | so close. |
| 23:05.04 | citywok | lol @drmes |
| 23:05.51 | citywok | we use atom's as our desktops now running win7 & softphones. the softphones probably won't last too much longer though. |
| 23:06.15 | talntid | polycoms here... |
| 23:06.19 | drmessano | Nothing like waiting 3 days for a phone to boot |
| 23:06.32 | citywok | lol, no kidding. we use aastra's. |
| 23:07.00 | talntid | you referring to the poly boot time? |
| 23:07.17 | drmessano | They call it "7" because that's the amount of RAM it takes to make it boot in under 5 minutes |
| 23:07.42 | talntid | i tried it. my android phone boots faster than my polycom |
| 23:07.52 | citywok | lol, it actually runs very well on 1gb on my netbook, i've got a 2gb stick but i'm too lazy to put it in. |
| 23:08.17 | citywok | my droid takes so long to boot that it makes me want to smash it to bits. if the polycom takes longer dear god what is it doing, building an arc? |
| 23:08.20 | citywok | ark* |
| 23:08.25 | drmessano | I've got 3GB on a year-old Dell notebook and it runs like XP did on a PIII.. |
| 23:08.30 | talntid | i have samsung captivate.. |
| 23:08.35 | talntid | so it's not terrible... |
| 23:09.04 | talntid | dell d830, 4gb ram, c2d.. :) runs 7 great. |
| 23:09.21 | talntid | except for graphics. the quadro video card chokes.. |
| 23:10.55 | drmessano | Well, I don't care what anyone says.. 7 is buggy as hell. There's way too many "glitchy" little things that make it intolerable at times |
| 23:11.15 | citywok | i haven't found anything i truly dislike about 7 |
| 23:11.43 | drmessano | I'm probably going to go back to XP at work until Wine runs all my specialized apps |
| 23:11.45 | citywok | the biggest annoyance i have is how the remote desktop windows snap and fullscreen if they aren't set to your max screen res. |
| 23:12.12 | *** join/#asterisk drudge` (tacos@unaffiliated/drudge/x-837452) |
| 23:12.12 | talntid | my desktop is way overpowered for windows 7... and my experience has been close to flawless... |
| 23:12.15 | citywok | i think it acts the way it does because of the new win7 snap feature, which is pretty freakin handy |
| 23:12.31 | drmessano | I knew Windows would finally snap |
| 23:13.20 | drmessano | I guess now would be a bad time to mention that snapping Windows have been in alternative OS'es for a while now |
| 23:13.32 | drmessano | No, no it wouldn't |
| 23:14.24 | citywok | lol, i can't really use anything else. i live in a microsoft world. |
| 23:14.41 | citywok | being a sysadmin for a windows network, using linux is like cutting my productivity in half to avoid the devil. lol. |
| 23:15.24 | drmessano | Same here, to a point.. I have a file server that I RDC into and handle my AD tasks |
| 23:15.46 | citywok | lol, 50 windows servers here. if i had to remote for everything i'd kill myself. |
| 23:15.47 | talntid | geeks. |
| 23:16.03 | citywok | once you figure out how totally awesome RemoteApp is... life will never be the same again |
| 23:16.58 | drmessano | Well, I have 40 machines in my automation system running Win2k, and if I didn't remote into them, I would spend a lot of time walking, driving, and standing in front of KVMs |
| 23:17.03 | drmessano | So you learn to remote |
| 23:17.11 | talntid | fuck. i forgot about a dinner date... and just ate a sandwich... |
| 23:17.25 | citywok | lol drmes i feel so bad for you with win2k. lol. |
| 23:17.40 | talntid | yeah, i'd slit my wrists. |
| 23:17.45 | drmessano | I have |
| 23:17.48 | drmessano | Several times |
| 23:17.49 | citywok | i generally have 3-8 RDP windows open, but remoteapp has drastically cut that down. Having to remote in to exchange to admin an exchange server is so 90's |
| 23:18.01 | drmessano | But they keep saving me, because they need me to keep the stations on the air |
| 23:18.24 | talntid | so tell them to upgrade....? |
| 23:18.34 | talntid | tell them y2k is coming, and win2k won't work. |
| 23:18.38 | citywok | yea unlessy ou work for free, upgrading is probably cheaper |
| 23:18.55 | drmessano | Not that easy.. it's like $4000 per machine |
| 23:19.31 | citywok | what the hell do they do? |
| 23:19.42 | drmessano | Run a bunch of a radio stations |
| 23:19.43 | citywok | if they are 10+ years old, the tech in them can't cost that much to upgrade |
| 23:19.58 | drmessano | Yeah it can, thanks to M$ |
| 23:20.25 | drmessano | New machines are $2100.. and because of Windows 7, all my $1800 sound cards are obsolete |
| 23:20.47 | talntid | write drivers :) |
| 23:20.47 | drmessano | Can't run XP on the machines, because Dell doesn't have drivers |
| 23:20.54 | drmessano | lol |
| 23:21.04 | citywok | Then buy HP's ;) |
| 23:21.11 | drmessano | Can't do that either |
| 23:21.38 | citywok | White box time then! |
| 23:21.44 | drmessano | I wish |
| 23:21.59 | drmessano | I could build something so much cheaper... Just isn't an option |
| 23:23.30 | drmessano | It's "turnkey by mandate" Approved hardware for the specific application.. hardware that requires Win7, that necessitates new sound cards.. |
| 23:23.48 | drmessano | brb |
| 23:28.00 | dmz | uhh why are you running xp then :) |
| 23:28.44 | citywok | b/c the $1,800 cards dont run on 7... |
| 23:30.09 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 23:34.30 | talntid | woot. a bug marshall :P |
| 23:37.13 | bmoraca_work | woohoo! texas won and the yankees are winning! |
| 23:38.01 | dan__t | No clue how the Rangers are pulling that off. |
| 23:38.19 | dan__t | They hadn't won a post season game in over 10 years or something ridiculous like that right? |
| 23:38.51 | bmoraca_work | today was their third ever post season win |
| 23:39.02 | dan__t | heh |
| 23:39.04 | *** join/#asterisk Faithful (~Faithful@180.194.0.54) |