| 00:00.10 | p3nguin | I don't really know how to answer that. |
| 00:01.23 | Micc | I just think it might need a change or two to work in my multi-tenant system. |
| 00:13.19 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 01:23.29 | picci | hey |
| 01:25.06 | picci | i tried connecting 2 pc's with xlite to a just installed asterisk server, when I try calling one from the other i get this: |
| 01:25.09 | picci | <PROTECTED> |
| 01:25.13 | picci | any ideas ? |
| 01:25.55 | p3nguin | The one you are calling can't be reached. |
| 01:26.29 | picci | they are both logged on, and whichever i try the call on i get the same result :S |
| 01:27.25 | p3nguin | Show me the output of sip show peers. |
| 01:28.40 | picci | virtual*CLI> sip show peers |
| 01:28.40 | picci | Name/username Host Dyn Nat ACL Port Status |
| 01:28.40 | picci | 333/333 (Unspecified) D N 0 UNKNOWN |
| 01:28.40 | picci | 666/666 (Unspecified) D N 0 UNKNOWN |
| 01:28.40 | picci | 2 sip peers [Monitored: 0 online, 2 offline Unmonitored: 0 online, 0 offline] |
| 01:29.06 | p3nguin | They are not registered. |
| 01:29.18 | picci | if i place a call it shows in the asterisk panel though... |
| 01:29.24 | p3nguin | They are not registered. |
| 01:29.38 | picci | k, gonna figure out how to do that and try again :) thx |
| 01:30.35 | p3nguin | Is asterisk behind NAT? |
| 01:30.49 | picci | asterisk is running on a vps in the us |
| 01:31.01 | picci | i'm stuck behind a large area nat in italy |
| 01:31.02 | p3nguin | Are the phones behind a NAT somewhere else? |
| 01:31.07 | p3nguin | okay |
| 01:35.22 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 01:35.34 | nny | what's the proper way to set user/group for asterisk to run as in 1.8? |
| 01:35.52 | p3nguin | I think I did it in the asterisk.conf. |
| 01:35.57 | picci | p3nguin: i pm'd u if u don't mind :) |
| 01:36.33 | nny | p3nguin: found it thanks |
| 01:36.53 | nny | p3nguin: hmm maybe not |
| 01:37.01 | nny | p3nguin: ps -aux still shows root, one sec |
| 01:37.13 | p3nguin | It's ps aux, not ps -aux. |
| 01:37.30 | nny | i set astctlowner, option for asterisk user not defined in sample asterisk.conf |
| 01:37.35 | nny | p3nguin: both work ^^ |
| 01:37.39 | p3nguin | Not really. |
| 01:37.45 | nny | ok nm |
| 01:37.50 | nny | ignore my question |
| 01:37.52 | nny | and move on |
| 01:37.54 | p3nguin | Warning: bad ps syntax, perhaps a bogus '-'? See http://procps.sf.net/faq.html |
| 01:38.29 | x86 | p3nguin: iPad access yet? |
| 01:38.34 | x86 | :) |
| 01:38.37 | p3nguin | not yet |
| 01:38.45 | x86 | mmk |
| 01:38.53 | p3nguin | Maybe a couple more hours. |
| 01:38.55 | nny | http://pastebin.com/GCKKWUA5 |
| 01:39.08 | nny | so yeah, but it still works ;) |
| 01:39.46 | nny | dammit i forgot my original question, this channel sometimes |
| 01:39.53 | p3nguin | So does rebooting to log off, but you could just log off instead. |
| 01:40.28 | nny | i'll ook at the docs, they're useful and not full of hot air thankfully |
| 01:40.33 | nny | look/ook |
| 01:43.11 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 01:44.41 | picci | yay, now peers are online |
| 01:44.51 | picci | still can't call each other though, says loop detected |
| 01:45.01 | p3nguin | Now you get to show me your dial plan. |
| 01:45.06 | p3nguin | ~pb |
| 01:45.06 | 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.us , 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. |
| 01:45.10 | p3nguin | pastebin it. |
| 01:46.11 | nny | hmm nope, no mention in asterisk.conf of actual user roles. I see how the script expects it, surprised this isn't easier to find online, since it's changed since earlier versions. OTOH my knowledge of the great PS command has been lifted to "exceptional" |
| 01:48.35 | p3nguin | checks asterisk.conf |
| 01:48.50 | p3nguin | Yep, the options are still there. |
| 01:50.48 | nny | would think AST_GROUP=asterisk AST_USER=asterisk in init.d would make if [ $AST_USER ] ; then ASTARGS="-U $AST_USER" true. Odd. |
| 01:50.48 | nny | <PROTECTED> |
| 01:56.29 | picci | http://pastebin.com/EJ6B3vMv |
| 01:56.58 | nny | picci: is that a space between 1, and Dial? |
| 01:57.33 | picci | yep |
| 01:57.51 | p3nguin | Get rid of the spaces. |
| 01:57.55 | p3nguin | Get rid of the register statements. |
| 01:58.24 | picci | if i get rid of the register statements... they don't register, at least they didn't when those two lines weren't there |
| 01:58.28 | [TK]D-Fender | Nice FreePBX leftovers... |
| 01:58.34 | picci | is getting rid of the spaces |
| 01:58.42 | p3nguin | And you didn't specify codecs for 666 but did for 333. |
| 01:58.56 | p3nguin | You clearly have no clue what register statements are for. |
| 01:59.01 | p3nguin | Get rid of them. They are wrong. |
| 01:59.10 | *** join/#asterisk neutr0n777 (~neutr0n77@c-69-181-27-226.hsd1.ca.comcast.net) |
| 01:59.17 | nny | [TK]D-Fender: my favorite |
| 01:59.23 | picci | codecs don't seem to matter atm |
| 01:59.30 | picci | i have no clue what the register statements are for |
| 01:59.34 | p3nguin | I know. |
| 01:59.37 | p3nguin | Get rid of them. |
| 01:59.51 | picci | if i get rid of them what do i put there instead ? if there's nothing there they don't show up as online in sip show peers |
| 01:59.54 | p3nguin | Register statements are for your asterisk to register to another system. |
| 02:00.15 | p3nguin | Tell your phones to register. |
| 02:00.19 | nny | well that's fixed, time for a beer. |
| 02:00.34 | p3nguin | With host=dynamic, that is the setting to allow phones to register to asterisk. |
| 02:00.39 | picci | yeh, that's why i didn't want them there in the first place :) (check pm's from 20 min's ago) |
| 02:00.46 | picci | gonna take them away |
| 02:01.00 | picci | and convince xlite to register... maybe |
| 02:01.20 | p3nguin | I don't accept unsolicited PMs, so anything you sent me went to the bit bucket. |
| 02:01.55 | [TK]D-Fender | Enable SIP debug and go look look at a registration attempt |
| 02:02.49 | picci | k |
| 02:03.01 | picci | anyways i get these often: -- Got SIP response 489 "Bad event" back from ... |
| 02:03.06 | picci | don't know if they're related or not |
| 02:05.04 | SeRi | for some time now only in incoming long distance calls from Puerto Rico I get very choppy sound |
| 02:05.08 | picci | seems to work :) |
| 02:05.14 | picci | thanks guys |
| 02:05.41 | picci | had to select register with domain on xlite |
| 02:05.52 | picci | that was all that was missing |
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| 02:25.24 | *** part/#asterisk picci (~picci@unaffiliated/picci) |
| 02:53.32 | p3nguin | Why is res_smdi.so loaded despite my having noload => res_smdi.so in modules.conf? |
| 02:54.11 | p3nguin | What module provides the features of features.conf? |
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| 03:10.53 | mocker | mental note, postpone cronjobs next time before maintenance window. |
| 03:14.42 | Micc | cronjobs make the world go round. |
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| 04:27.57 | SeRi | p3nguin, you avail? |
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| 04:38.47 | dijib | can someone tell me whats going on in either of these |
| 04:38.50 | dijib | http://pastebin.com/A8wSew2P |
| 04:38.57 | dijib | http://pastebin.com/Wkkrj3bF |
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| 05:27.51 | atan | Silly question but does 1 646-454-3209 ring for anyone? It's Logitech ( http://www.logitech.com/en-us/contact ) |
| 05:28.06 | atan | For me I get dead air but I'm not sure if it's jusy my voip provider |
| 05:28.38 | atan | Err got it |
| 05:28.44 | atan | Strange. |
| 05:29.35 | dijib | atan, i just tried it and it works |
| 05:30.10 | dijib | i had that issue on a new install i did few weeks ago. 3/4 calls were just dead air.. to any number |
| 05:30.12 | atan | I think the dashes were being passed in when they were not needed... must look in to that |
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| 05:30.24 | atan | dijib: interesting that's how I feel right now |
| 05:30.36 | dijib | 3/4 calls work? |
| 05:30.50 | dijib | i mean 1/4 |
| 05:30.52 | dijib | work |
| 05:30.54 | atan | Well I tried 3 or 4 times. The 4th worked. |
| 05:31.03 | dijib | what version? |
| 05:31.11 | atan | I just went back into my recently dialed list to check and redialed one of the failed numbers and it worked this time |
| 05:31.15 | atan | I am on I think 1.8.7 |
| 05:31.25 | dijib | new install |
| 05:31.37 | atan | I just installed it recently, like, day ago or so |
| 05:31.40 | dijib | how did you install it? |
| 05:31.50 | atan | Downloaded source, config, make, make install :D |
| 05:31.55 | dijib | i did a reinstall on centos and it works 100% now |
| 05:32.06 | atan | I'm running Debian |
| 05:32.10 | dijib | yeah thats how i did it the 1st time where i had the issue |
| 05:32.38 | dijib | i used the packet manager the 2nd time and works perfect |
| 05:33.41 | dijib | i upgraded today to 1.8.6 and things are still hunky dory.. except it broke my MOH |
| 05:33.53 | atan | My MOH is messed up right now also. No music, just hold. |
| 05:34.00 | dijib | same |
| 05:34.09 | atan | Did you use 10.x at one point and revert? |
| 05:34.41 | dijib | i suggest uninstalling the source install however thats done and reinstall through a packet manager |
| 05:34.59 | dijib | no never 10 |
| 05:35.47 | zamba | what's wrong with this? exten => _X.,n,GotoIf($["${diff}" > "60"]?900) |
| 05:35.56 | zamba | i want to check if the variable diff is higher than 60 |
| 05:36.10 | zamba | and if it is, go to the priority 900 in the current context |
| 05:36.28 | dijib | :\ im a nuub |
| 05:40.08 | dijib | zamba, http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIf |
| 05:40.19 | zamba | dijib: i've read that |
| 05:41.07 | dijib | yeah but it looks like you need to dial1 dial2 |
| 05:41.09 | zamba | already got it working |
| 05:41.09 | dijib | part |
| 05:41.10 | dijib | no? |
| 05:41.12 | zamba | no |
| 05:41.15 | zamba | you're a nuub :) |
| 05:41.22 | dijib | yeh.. just trying to help |
| 05:42.33 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
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| 05:50.16 | zamba | dijib: yeah, no worries, it's appreciated |
| 05:59.49 | kaldemar | zamba: drop the "'s in the expression |
| 06:00.17 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:00.19 | schmidts | good morning |
| 06:05.42 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 06:09.11 | ChannelZ | Yes, or something. |
| 06:11.37 | *** join/#asterisk irroot (~irroot@196-210-249-226.dynamic.isadsl.co.za) |
| 06:13.59 | irroot | morning folks |
| 06:14.04 | ollii | g'morning |
| 06:16.36 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:16.44 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 06:16.48 | schmidts | morning irroot |
| 06:17.10 | irroot | man i hate this town 600km from home at sea level [1600m below home] and in sub tropics not temperate bush |
| 06:17.12 | joobie | hey guys.. anyone able to help debug a voicemail notificaiotn issue? notify light is not coming on a polycom 321 for some reaosn.. works on other phones, other polycom 321's |
| 06:17.36 | joobie | not sure how to debug in great detail.. confirmed mailbox= is correct in sip.conf |
| 06:18.06 | irroot | joobie mwisubscribe in sip.conf and the mailbox setting on phone what version there were problems in early 1.8.X < 6 |
| 06:18.20 | joobie | irroot, 1.4 |
| 06:19.00 | irroot | joobie been a while i used 1.4 check sugestions above and seriously concider upgrade |
| 06:19.07 | *** join/#asterisk rajiv (~rajiv@gentoo/developer/rajiv) |
| 06:19.26 | joobie | irroot, i am going to upgrade.. but in a few months - in my holidays :P |
| 06:19.26 | tyman | having trouble getting the 'messages' softkey working on polycom phones... my mwi is working fine, just broken softkey |
| 06:19.33 | joobie | need somethign in the interim |
| 06:25.01 | joobie | irroot, any ideas on how to debug on 1.4? |
| 06:25.11 | joobie | it's a sip notify that's send for the wmi right? |
| 06:25.21 | irroot | make sure phone has mailbox setting set |
| 06:25.35 | joobie | done |
| 06:25.36 | irroot | sip set debug ip X.X.X.X |
| 06:25.52 | joobie | i have a few phones coming frmo the same public ip |
| 06:25.57 | joobie | can i do that based on the phone's private ip? |
| 06:25.59 | irroot | also event debug but not sure if thats 1.4 |
| 06:26.20 | joobie | not here |
| 06:26.20 | irroot | joobie sip set debug peer <peername> |
| 06:26.27 | joobie | ty |
| 06:26.35 | joobie | will try now |
| 06:28.01 | joobie | what should i look for in the sip debug irroot ? |
| 06:28.16 | irroot | notify message |
| 06:28.54 | zamba | ok, i've tried using DBdel() but then i get the notice that i should use DB_DELETE() instead |
| 06:29.03 | zamba | [2011-10-05 08:28:33] WARNING[19453]: pbx.c:3675 pbx_extension_helper: No application 'DB_DELETE' for extension |
| 06:29.11 | zamba | what's going on here? |
| 06:29.59 | schmidts | zamba which version do you use? |
| 06:30.08 | zamba | schmidts: 1.6.2 |
| 06:30.15 | zamba | Asterisk 1.6.2.5-0ubuntu1.4 built by buildd @ palmer on a i686 running Linux on 2011-07-12 21:26:25 UTC |
| 06:30.18 | zamba | the ubuntu version |
| 06:30.24 | schmidts | zamba DB_DELETE is a function, DBDEL is an application |
| 06:30.34 | zamba | schmidts: so how is db_delete used? |
| 06:30.39 | irroot | core show applications / functions |
| 06:31.02 | irroot | functions are used via the "SET" application[s] |
| 06:31.12 | zamba | oh |
| 06:31.13 | joobie | irroot, looked ok? http://pastebin.com/52ppwxfY |
| 06:31.14 | irroot | or as varibles |
| 06:31.18 | schmidts | if you want to use db_delete it should look like this: exten => _!,n,Noop(DB_DELETE(SIP/Registry/1234)) or whatever you want to delete |
| 06:31.50 | zamba | ah, sweet.. thanks |
| 06:32.12 | irroot | joobie looks good 7 new 2 old messages |
| 06:33.12 | joobie | irroot, but no LED on the phone |
| 06:33.25 | irroot | mmm |
| 06:35.50 | joobie | Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY |
| 06:36.02 | joobie | it has that too in the OPTIONS (notify specified).. |
| 06:37.02 | irroot | from the trace its all goosd |
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| 06:39.09 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 06:39.45 | joobie | thanks irroot |
| 06:40.01 | joobie | ill have a look into the phone config |
| 06:40.10 | joobie | just doubtful it is this because the same template has been used at a few sites |
| 06:40.14 | joobie | and it works.. |
| 06:40.34 | joobie | but clutching at straws.. were the 1.4 bugs with WMI to do with the NOTIFY msg? |
| 06:40.48 | joobie | ie with that trace looking the way it is, i can presume it's not a 1.4 issue? |
| 06:45.13 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 06:51.08 | irroot | joobie check your mailbox setting in the sip.conf |
| 06:51.32 | irroot | Message-Account: sip:asterisk@210.5.19.5 |
| 06:52.10 | joobie | my mailbox is 3019@default |
| 06:53.02 | joobie | irroot, but that doesnt seem to match up with the mailbox= setting in sip.conf |
| 06:53.24 | joobie | i tried a debug on another phone (same model) and it had asterisk@210.5.19.5 also |
| 06:53.33 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:2c74:de81:da23:593c) |
| 06:54.45 | irroot | ok np just seemed odd |
| 06:55.36 | dijib | how should this be writen? |
| 06:55.37 | dijib | same => n,System("cat /tmp/forecast.txt | text2wave -o /var/lib/asterisk/sounds/forecast.ulaw -otype ulaw"); |
| 06:55.40 | joobie | ya |
| 06:55.56 | zamba | can someone proof read this and maybe come up with some improvements: http://pastebin.com/pcgjjjh6 |
| 06:56.02 | zamba | i'm quite new to dialplan programming, so |
| 06:58.30 | dijib | board is spelt like that not bord |
| 06:58.37 | dijib | nevermind |
| 07:01.10 | joobie | heading home |
| 07:01.13 | joobie | thanks for the help irroot |
| 07:01.17 | joobie | will let you know how i progress |
| 07:01.31 | irroot | pleasure |
| 07:01.51 | irroot | im a bit tied down and not in office so please bear with me |
| 07:03.52 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 07:08.05 | zamba | dijib: wth? :) |
| 07:08.18 | zamba | dijib: yeah, i guess "nevermind" is what you're looking for :) |
| 07:08.28 | dijib | lol |
| 07:11.45 | *** join/#asterisk irroot (~irroot@196-210-249-226.dynamic.isadsl.co.za) |
| 07:17.01 | schmidts | zamba you should use labels instead of context numbers for example exten => _X.,n(firststop),Goto(firststop) |
| 07:17.25 | zamba | schmidts: oh, ok |
| 07:19.06 | schmidts | zamba the problem with fixed numbers is, that the next n will be count up by this and this could have some bad side effects on your dialplan, and it makes it harder to add new stuff cause you allwasy have to keep in mind how many rows you have between the next hop |
| 07:19.26 | zamba | yeah.. but that's also why i did 200 and 900 |
| 07:19.29 | zamba | to have some space in between |
| 07:19.47 | schmidts | zamba and row 4 and row 5 could be easy combined like this: exten => _X.,n,GotoIf(${DB_EXISTS(${key})}?200:sentralbord,bord,1) |
| 07:20.06 | zamba | ah, of course |
| 07:20.07 | zamba | thanks |
| 07:20.45 | kaldemar | the GotoIf is missing $[] |
| 07:20.46 | zamba | anything else? the logic itself seem sound? |
| 07:21.05 | ChannelZ | and your dog just got out |
| 07:21.06 | zamba | added |
| 07:21.15 | zamba | ChannelZ: ok? :) |
| 07:23.19 | zamba | but i have another problem.. if the called party never calls back, then the database entry won't get purged.. |
| 07:25.37 | zamba | how would you suggest i solve that? |
| 07:26.44 | zamba | it's basically a matter of going through every key for a given family in the astdb and check each timeout value and remove the key if the timeout has expired |
| 07:26.55 | zamba | and that has to be done every time i get an incoming call |
| 07:27.21 | zamba | can i put a hash in the database? |
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| 07:29.17 | kaldemar | zamba: doesn't sound too good to do such operations every time you get a call. i'd suggest handling that in some other way. |
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| 07:36.24 | zamba | kaldemar: any suggestions? |
| 07:40.06 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 07:42.10 | eva_02 | Hello, i need help with simple asterisk configuration. I can pay for it. |
| 07:42.22 | ChannelZ | Oh, you'll pay! |
| 07:42.54 | irroot | but it wont cost :P |
| 07:42.57 | ChannelZ | In our mocking! |
| 07:43.04 | ChannelZ | hehe |
| 07:43.14 | atan | This is so true. |
| 07:43.16 | ChannelZ | Try us out first |
| 07:43.36 | atan | eva_02, what did you break? |
| 07:43.50 | irroot | repeats 1000 times as per therapist "be nice to users ... be nice to users .... rm -rf .... " |
| 07:44.33 | eva_02 | But i'm afraid that you will laugh of me, i'm very stupid |
| 07:44.42 | kaldemar | zamba: i don't know what you're trying to achieve. |
| 07:44.46 | atan | Join my club. |
| 07:44.58 | atan | eva_02, the only stupid question is one you know the answer to :D |
| 07:45.00 | kaldemar | ~ask |
| 07:45.00 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 07:45.02 | ChannelZ | We are like bullies, someone more stupid will come in and we'll forget all about you and start mocking someone else ;) |
| 07:45.03 | kaldemar | eva_02: ^ |
| 07:45.19 | irroot | ChannelZ lol |
| 07:45.23 | ChannelZ | I kid, I kid.. |
| 07:45.35 | ChannelZ | most of us at this hour are harmless |
| 07:46.37 | irroot | ChannelZ speak for self im still only on second cuppa coffee |
| 07:47.10 | ChannelZ | well, I said most. |
| 07:47.41 | irroot | add to the fact i flew out this am am 1600m bellow normal this o2 on the coast is hectic plus its in sub tropics .... |
| 07:48.25 | eva_02 | so i have one sip trunk (sip account with login:password), i want forward all incoming calls to another sip login@domain.com |
| 07:48.45 | eva_02 | i don't need any prefixes, IVR's, dialplun's and other |
| 07:49.09 | ChannelZ | so you literally want to just take an incoming call and dial the same thing out someplace else? |
| 07:49.13 | atan | You need to add that to your dialplan I would think :-) match the incoming call to your number and Dial your ither SIP account |
| 07:49.21 | eva_02 | ChannelZ: yes |
| 07:50.06 | ChannelZ | Asterisk might be a bit of overkill for just that, but it's easy enough to do. Do you have the SIP accounts already setup and working or no |
| 07:50.38 | eva_02 | i have successfully connect sip trunk by put in sip.conf string: register=123:secret@domain.com/123 |
| 07:51.20 | eva_02 | sorry my lame, i'm try to read manual but it so huge |
| 07:52.09 | ChannelZ | registering is only half the battle, you need to make a sip peer for them as well |
| 07:52.51 | ChannelZ | actually it's less than half the battle, it really only lets the server you're registering to know your IP so it knows where to send calls. But you need a peer to be able to receive them |
| 07:53.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:53.25 | irroot | and then to make sure RTP passes any NAT/FW |
| 07:54.18 | ChannelZ | cripes I should be in bed |
| 07:54.34 | irroot | ChannelZ but not alone ? |
| 07:55.19 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:56.26 | eva_02 | hold on, i try to paint my scheme |
| 07:59.01 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.181.46) |
| 07:59.05 | devil_evoxxx | hi all :) |
| 07:59.28 | irroot | devil_evoxxx yo there |
| 08:08.15 | devil_evoxxx | good morning :) all fine? |
| 08:10.44 | eva_02 | ChannelZ: http://rghost.ru/24274511/image.png |
| 08:12.50 | eva_02 | so i need forward all incoming calls on +1 123123123 to me@stupid.com |
| 08:13.45 | kaldemar | eva_02: do you have a matching peer for the 123 account? |
| 08:14.27 | eva_02 | kaldemar: i don't understand what you asking |
| 08:15.16 | kaldemar | eva_02: http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#OutsideConnectivity_id36059950 |
| 08:16.20 | kaldemar | eva_02: a peer definition is a way to know where calls are coming from, and the path to defining what to do with them. |
| 08:18.03 | kaldemar | eva_02: add a context parameter to the entry and the calls coming from the provider land in that context in your dialplan, which is in extensions.conf. |
| 08:18.53 | eva_02 | okay, how change this string for me exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) |
| 08:19.00 | eva_02 | i mean _1NXXX |
| 08:19.59 | kaldemar | start with the peer entry. |
| 08:20.50 | kaldemar | you don't need a pattern since the calls come in to the same number every time. |
| 08:21.40 | kaldemar | exten => 123,1,Dial(SIP/me@studip.com) is enough, if the calls come in to number 123. |
| 08:26.01 | eva_02 | kaldemar: can you please show me how must look full extentions.conf |
| 08:28.13 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 08:28.16 | v0lZy | hello |
| 08:28.38 | v0lZy | i hae no idea where to ask, but, does anyone know of any hardware that allows to bridge 3g to lan |
| 08:28.48 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:29.11 | eva_02 | kaldemar: here is my sip.conf http://pastebin.com/raw.php?i=pAWu2reb |
| 08:29.36 | irroot | v0lZy linux with usb port |
| 08:29.36 | v0lZy | basically, I want a 3g access point for mobile phones so I can share LAN with the phones.. so somethin glike phone----3gAccessPoint----LAN--router--internet |
| 08:29.57 | irroot | its pretty plug and play use same box as asterisk |
| 08:30.38 | v0lZy | irroot: im not sure i make myself clear... I dont want to get internet via 3g and share it to other computers... I want my own private mini 3g network |
| 08:30.51 | irroot | yeah 100% |
| 08:31.10 | v0lZy | what kind of hardware do i need? |
| 08:31.19 | v0lZy | i imagine i need some kind of 3g antena |
| 08:31.21 | irroot | google mifi |
| 08:31.38 | irroot | its wifi 3g bridge |
| 08:32.15 | *** join/#asterisk MrSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 08:32.55 | tuxx- | hi guys, how can i debug receivefax? i'm always getting the following error: The call dropped prematurely |
| 08:33.06 | v0lZy | irroot: i dont think thats what i need |
| 08:33.23 | tuxx- | here is my asterisk cli log: http://pastie.org/2642602 |
| 08:33.25 | irroot | tuxx- version ?? |
| 08:33.32 | v0lZy | I need it in reverse.... I want to share my FTTH with a phone, not by wifi, but by 3g |
| 08:33.55 | tuxx- | irroot: Asterisk 1.8.5.0, Copyright (C) 1999 - 2011 Digium, Inc. and others. |
| 08:34.36 | irroot | with res_fax or app_fax |
| 08:34.41 | tuxx- | res_fax |
| 08:35.32 | irroot | ok cool |
| 08:35.55 | kaldemar | v0lZy: just out of curiosity, why do you want to use a 3G base station instead of wifi? |
| 08:36.08 | v0lZy | battery life. |
| 08:36.16 | irroot | cor set debug 1 |
| 08:36.25 | v0lZy | I have a dect central here that needs replacing |
| 08:36.31 | v0lZy | most of us have smartphones |
| 08:36.45 | irroot | s/cor/core/ |
| 08:36.49 | tuxx- | i see receivefax takes a parameter 'd' for debug. shouldi enable that too? |
| 08:36.53 | v0lZy | would be great to connect them to our ip pbx |
| 08:37.01 | v0lZy | and not pay our current providers data transfer rates. |
| 08:37.07 | irroot | tuxx- yeah do so |
| 08:37.21 | v0lZy | I can get rid of desktop phones as well |
| 08:37.37 | kaldemar | v0lZy: is it legal in your country to set up 3G stations just like that? |
| 08:37.37 | tuxx- | okay, gonna get some more logs :) |
| 08:38.03 | v0lZy | kaldemar: no idea, just thought of it an hour ago. |
| 08:38.18 | v0lZy | but if its limited to a certain area, i dont see why it would be a problem |
| 08:41.59 | v0lZy | i think this stuff is called 3g femtocell |
| 08:51.57 | tuxx- | irroot: http://pastie.org/2642660 <- with `core set debug 3` and d parameter for receivefax. Doesnt seem like i get any extra debugging messages :-( |
| 08:52.16 | irroot | tuxx- check logger.conf |
| 08:52.33 | irroot | console => debug,notice,warning,error |
| 08:52.46 | tuxx- | right |
| 08:52.55 | tuxx- | missed the debug param there |
| 08:55.42 | eva_02 | kaldemar: thank you very much, everything work right. I do this http://pastebin.com/raw.php?i=AfBDVB0F |
| 08:56.23 | eva_02 | can you please check the link, maybe something wrong |
| 08:57.26 | tuxx- | http://pastie.org/2642680 there we go |
| 08:58.33 | tuxx- | http://pastie.org/2642684 |
| 08:59.14 | *** join/#asterisk maldous (~root@124-148-129-91.dyn.iinet.net.au) |
| 08:59.29 | maldous | howdy. |
| 09:02.10 | tuxx- | hm, do i need to tweak res_fax.conf for european faxes or something? (my fax knowledge is really low as you can see ;-) |
| 09:02.29 | irroot | Channel 'DAHDI/i4/-10' did not return a frame; tuxx- |
| 09:02.36 | tuxx- | hmkay |
| 09:02.55 | irroot | tuxx- i know nothing about faxing ;) |
| 09:03.10 | tuxx- | :) |
| 09:03.15 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:03.32 | tuxx- | http://forums.digium.com/viewtopic.php?t=73761 someone with the same problem it seems, but no way to solve it :-P |
| 09:04.49 | irroot | http://pastebin.com/ySrREEmX |
| 09:07.03 | coppice | tuxx: oh yes he does |
| 09:07.25 | coppice | everyone point at irroot when FAX questions come up |
| 09:07.30 | irroot | coppice <- is the spandsp aka fax over ip king |
| 09:07.49 | tuxx- | any help is greatly appreciated ;-) |
| 09:08.03 | coppice | I wish people wouldn't keep calling spandsp a FAX machine |
| 09:08.26 | irroot | tuxx- we need to find out why the line is droping |
| 09:08.53 | irroot | coppice yeah lspandsp is such awesome tool and faxing < 1/2 of it |
| 09:09.51 | irroot | <PROTECTED> |
| 09:10.13 | tuxx- | seems like its all the time, gonna ask someone to send me a fax from a different location |
| 09:10.21 | irroot | set up a context for sendfax and then originate call to the faxrec dp |
| 09:10.41 | tuxx- | mkay |
| 09:10.46 | irroot | you using PRI right is it to the telco |
| 09:11.31 | tuxx- | we are using BRI to our telco |
| 09:11.37 | irroot | ah ok |
| 09:11.49 | irroot | b410p then |
| 09:11.57 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:11.58 | tuxx- | B4XXP (PCI) Card 0 Span 1 |
| 09:11.58 | tuxx- | :) |
| 09:12.20 | maldous | anyone know how to match any extension in a context? |
| 09:12.23 | irroot | that should be fine use BRI myself |
| 09:12.46 | irroot | but with misdn [old school |
| 09:12.59 | maldous | I'm getting "chan_sip.c:20276 handle_request_invite: Call from '' to extension 'p12345678' rejected because extension not found in context 'p12345678'." with every variation I can think of.. |
| 09:13.21 | maldous | i thought "s" should match anything. |
| 09:13.54 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:14.27 | irroot | the p is important there |
| 09:14.43 | irroot | maldous if its dahdi use immeadiate=yes |
| 09:15.07 | maldous | it's sip to a broadworks |
| 09:15.49 | irroot | tuxx- try setting the settings in res_fax.conf |
| 09:16.08 | irroot | tuxx- "core show function FAXOPT" |
| 09:16.41 | maldous | the p is important? how so? |
| 09:17.06 | irroot | maldous pb your dialplan your exten needs to match the leading p |
| 09:17.35 | devil_evoxxx | irroot: have you got any hint where i can find in asterisk source the relative section of "rtp set debug ip" |
| 09:17.42 | *** join/#asterisk gego (~quassel@b238085.customer.hansenet.de) |
| 09:18.03 | kaldemar | maldous: s matches when the extension is literally s or there is no extension. it is not a wildcard. |
| 09:18.31 | devil_evoxxx | i'm not a guru of C, but i already worked on C project |
| 09:18.33 | maldous | i'm trying "exten => p12345678,1,Answer" - no luck |
| 09:18.59 | kaldemar | maldous: then you're putting it in the wrong context. |
| 09:20.05 | maldous | the error message (above) shows the context |
| 09:20.14 | maldous | does the context need to be in sip.conf or extensions.conf ? |
| 09:21.33 | irroot | devil_evoxxx prolly in res/res_rtp_engine or main/rtp.... |
| 09:22.34 | maldous | what does "_" mean in an extension? |
| 09:23.14 | irroot | maldous its a "match" indicates to match pattern no explicit extension |
| 09:23.15 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 09:23.19 | v0lZy | hello |
| 09:23.23 | kaldemar | maldous: the context refers to a dialplan context, which is in extensions.conf. "_" denotes a pattern- |
| 09:23.40 | v0lZy | I have found a way to achieve what I want, although in a very different way... |
| 09:24.17 | maldous | oh, would _ by itself match anything? |
| 09:24.31 | v0lZy | just a question, how can i script asterisk so that when a certain number is called, a dialprogram answers and then redirects to a number that is inputed by the user? |
| 09:24.53 | maldous | i just want asterisk to register as a sip client to my softswitch, which it does. i'm now dialing the number, but getting all calls rejected because the extension isn't found. |
| 09:25.05 | irroot | devil_evoxxx you have svn checkout of branches/1.8 ? |
| 09:25.15 | v0lZy | i have 2 phone line providers, and a gsm bridge... now i want my gsm users to call a gsm number, then be able to pick an extension |
| 09:25.35 | v0lZy | but since theres a lot of extensions... id like them to input the extension and press * or something |
| 09:25.54 | kaldemar | maldous: no, it just enables pattern matching. |
| 09:27.20 | maldous | is there a way to debug the extension checking? |
| 09:27.57 | irroot | maldous "core set verbose 3" then "dialplan reload" |
| 09:28.08 | maldous | thx |
| 09:28.33 | irroot | will show you what it does |
| 09:28.44 | kaldemar | maldous: from what you already pasted, your asterisk is looking for "exten => p12345678,..." in [p12345678]. |
| 09:29.59 | maldous | yeah, trying. no luck. |
| 09:31.00 | maldous | in my sip.conf, i have "context=foo", then have [foo] later in sip.conf with "exten => p12345678,.." - i've just renamed the context, restarted, and get the same error. |
| 09:31.14 | maldous | the debug with core set verbose isn't showing anything extra. |
| 09:31.53 | gego | hi there, i've got a problem here with pickups from grandstream phones after changing * 1.4 to 1.6.2.20 |
| 09:31.54 | kaldemar | maldous: dialplan is in extensions.conf. you're modifying the wrong file. |
| 09:32.26 | maldous | ah. goodo. will try now. |
| 09:32.34 | kaldemar | maldous: you should put the [foo] in extensions.conf and the extension under it. |
| 09:32.47 | gego | it's all sip in * , incoming calls come from a patton smartnode. |
| 09:33.04 | gego | internal pickups work, but not from the patton |
| 09:33.32 | maldous | YAY! thankyou |
| 09:34.32 | gego | as far as I understand the internal documentation of the changed app Pickup, i have to set notifycid=ignore-context in sip, if the hints and the pickups are in different contexts, which i did. |
| 09:41.28 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:50.58 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 09:53.19 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 09:54.00 | *** join/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 09:57.05 | tuxx- | ct 5 11:54:54] WARNING[10229]: res_fax_spandsp.c:367 spandsp_log: WARNING T.30 Cannot open source TIFF file '/tmp/fax.tiff' |
| 09:57.09 | tuxx- | eughhh |
| 09:57.11 | tuxx- | :D |
| 10:01.12 | tuxx- | hm, originating a call, and sendfax() a tiff file to my receivefax() seems to work correctly. |
| 10:01.59 | maldous | kaldemar: thank you muchly! working. now to figure why tones aren't being received.. |
| 10:07.49 | maldous | jeeeze, so many configs to work with. |
| 10:09.39 | *** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it) |
| 10:09.48 | tuxx- | :P |
| 10:11.31 | tuxx- | hm weird irroot, just let someone in my town sent a fax to me, and it receives correctly, without me changing anything about my setup :P |
| 10:11.44 | irroot | yeah |
| 10:12.02 | irroot | the fax that was been used earlier is a dog |
| 10:15.10 | tuxx- | hehe |
| 10:15.33 | toresbe | What were you doing to the dog to make it modulate data? |
| 10:15.43 | toresbe | I tought mine to shake hands, but I never got it to handshake. |
| 10:15.52 | toresbe | taught |
| 10:16.01 | coppice | QAM BAM and out comes modem tones |
| 10:18.57 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:19.14 | *** join/#asterisk GreatSUN (~greatsun@91.112.72.178) |
| 10:19.44 | GreatSUN | re |
| 10:22.14 | irroot | tuxx- unfortunately some systems they dont work so well and coppice has done as much as he can to stick to the standard when some one makes up there own standard not everything works with it |
| 10:22.28 | tuxx- | right |
| 10:22.42 | tuxx- | we'll just take it into production, and see how many faxes actually fail |
| 10:23.21 | irroot | suggest limiting modem / minrate /maxrate and ecm options |
| 10:23.34 | tuxx- | mkay |
| 10:23.46 | tuxx- | tnx for all the help so far irroot and coppice :) |
| 10:24.34 | irroot | tuxx- if you want to do FOIP ie T38 there is a branch for it |
| 10:24.50 | irroot | things like linksys/grandstream to a faxmachine |
| 10:35.01 | devil_evoxxx | irroot: yes |
| 10:36.25 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 10:36.28 | *** join/#asterisk salz212 (~chatzilla@182.178.239.134) |
| 10:39.25 | salz212 | I am a bit confused by DIALSTATUS variable, I have 2 different trunks for outbound and I want to do a fail-over when the first one does not work (either gets unreachable or reachable but not routing calls). I have used different dialplans and s extensions, but hav't had much luck. the BUSY, No-Answer Congestion and CHANUNAVAILABLE almost behave same way.. Any suggestion for a proper failover? |
| 10:40.46 | *** join/#asterisk hrolf (~hrolf@unaffiliated/hrolf) |
| 10:41.04 | hrolf | How can I get the time in dial plan? Like the start time of the call? |
| 10:42.01 | irroot | devil_evoxxx cool then grep the code for the error then use "svn blame" to work back to when it had changed |
| 10:44.31 | kaldemar | hrolf: ${EPOCH} and func STRFTIME |
| 10:44.33 | *** join/#asterisk bjhaid (~abejide@41-147.rv.ipnxtelecoms.com) |
| 10:44.51 | bjhaid | hi, I am trunking asterisk with switchvox it worked for a while, and later started getting SIP response 503 "Service Unavailable" on asterisk |
| 10:44.58 | salz212 | any one willing to answer my question? |
| 10:45.21 | hrolf | kaldemar: Doing ${EPOCH} will give system time right? and if I take that as the the call start time, it will be fine, right? |
| 10:46.13 | devil_evoxxx | irroot: thanks :) i think is so hard for me bu i like challange |
| 10:46.40 | bjhaid | hi, I am trunking asterisk with switchvox it worked for a while, and later started getting SIP response 503 "Service Unavailable" on asterisk any suggestions on what the problem is? |
| 10:48.45 | salz212 | I am a bit confused by DIALSTATUS variable, I have 2 different trunks for outbound and I want to do a fail-over when the first one does not work (either gets unreachable or reachable but not routing calls). I have used different dialplans and s extensions, but hav't had much luck. the BUSY, No-Answer Congestion and CHANUNAVAILABLE almost behave same way.. Any suggestion for a proper... |
| 10:48.47 | salz212 | ...failover? The problem is when I use GOTO(s-${DIALSTATUS}) it goes to failover even when no answer or busy. |
| 10:50.27 | kaldemar | hrolf: yes. |
| 10:50.36 | hrolf | kaldemar: Thanks.. |
| 10:52.06 | kaldemar | salz212: where it goes is up to your dialplan. |
| 10:52.55 | salz212 | didn't get you, what? |
| 10:53.31 | salz212 | are you referring to the goto statement? |
| 10:53.33 | hrolf | Can we do this "exten => _XXXX,1,Set(CACA=23123),Set(asdas==123123)" |
| 10:53.39 | hrolf | I mean can we do two Set ? |
| 10:53.49 | hrolf | or do I have to do that in a different line |
| 10:57.53 | *** part/#asterisk bjhaid (~abejide@41-147.rv.ipnxtelecoms.com) |
| 11:01.22 | salz212 | Any one with any idea about failover, using dialstatus as a jump? |
| 11:01.29 | kaldemar | salz212: i'm referring to the rest of your dialplan. |
| 11:01.59 | kaldemar | hrolf: you can't call two applications in one priority. you have to use separate priorities. |
| 11:03.10 | salz212 | actually its a macro with first line .. Dial(---- OUTBOUND trunk 1) line 2 goto(s-${DIALSTATUS}) and after that there are s extensions like s-Answer s-BUSY etc which dial from the secondary trunk |
| 11:03.31 | kaldemar | hrolf: application MSet on the other hand can be used to set multiple variables in a single command. |
| 11:05.00 | hrolf | kaldemar: I see but voip-info says not to use it. So I guess I'll stick with Set |
| 11:05.13 | salz212 | Do you want me to paste the dialpan? |
| 11:05.22 | kaldemar | salz212: your goto syntax is not what you expect. Goto([[context,]extension,]priority) |
| 11:05.43 | kaldemar | hrolf: i wouldn't trust voip-info. |
| 11:06.02 | hrolf | kaldemar: Why? Is there a better source? |
| 11:07.12 | kaldemar | hrolf: voip-info is generally the last place to use for documentation. if you want a wiki, use the official one: https://wiki.asterisk.org/wiki/display/AST/Application_MSet |
| 11:07.30 | salz212 | I hope I am not offending the channel, here is the macro |
| 11:07.37 | salz212 | [macro-cell] |
| 11:07.39 | salz212 | exten => s,1,Dial(${ARG2},,t) |
| 11:07.39 | atan | Anyone here from Canada dealing with US ITSPs? |
| 11:07.40 | salz212 | exten => s,n,AGI(agi-sal.agi,${DIALSTATUS}) |
| 11:07.42 | salz212 | exten => s,n,Goto(s-${DIALSTATUS},1) |
| 11:07.44 | salz212 | exten => s-NOANSWER,1,Set(CALLERID(All)=LHR-PBX) |
| 11:07.45 | salz212 | ;exten => s-NOANSWER,n,Dial(${ARG3},30)615969 |
| 11:07.47 | salz212 | exten => s-NOANSWER,n,Dial(${iplivr}/${ARG2:15},30) |
| 11:07.48 | salz212 | exten => s-NOANSWER,n,Playback(all-outgoing-lines-unavailable) |
| 11:07.50 | salz212 | exten => s-NOANSWER,n,Voicemail(${ARG1}@default) |
| 11:07.51 | salz212 | exten => s-NOANSWER,n,Hangup() |
| 11:07.53 | salz212 | ;exten => s-CHANUNAVAIL,1,Set(CALLERID(All)=LHR-PBX) |
| 11:07.54 | salz212 | ;exten => s-CHANUNAVAIL,n,Dial(${iplivr}/${ARG2:15},30) |
| 11:07.56 | salz212 | ;exten => s-CHANUNAVAIL,n,Playback(all-outgoing-lines-unavailable) |
| 11:07.57 | salz212 | ;exten => s-CHANUNAVAIL,n,Voicemail(${ARG1}@default) |
| 11:07.59 | salz212 | ;exten => s-CHANUNAVAIL,n,Hangup() |
| 11:08.01 | salz212 | exten => s-BUSY,1,Set(CALLERID(All)=LHR-PBX) |
| 11:08.02 | salz212 | ;exten => s-BUSY,n,Dial(${iplivr}/${ARG2:20},30) |
| 11:08.04 | salz212 | exten => s-BUSY,n,Dial(${iplivr}/${ARG2:15},30) |
| 11:08.05 | salz212 | exten => s-BUSY,n,Playback(all-outgoing-lines-unavailable) |
| 11:08.07 | salz212 | exten => s-BUSY,n,Hangup() |
| 11:08.08 | atan | Salz: pastebin.com my man |
| 11:08.08 | salz212 | exten => s-BUSY,101,Playback(invalid) |
| 11:08.10 | salz212 | exten => s-BUSY,102,Hangup() |
| 11:08.12 | salz212 | exten => _s-.,1,Goto(s-NOANSWER,1) |
| 11:08.13 | salz212 | exten => a,1,VoicemailMain(${ARG1}) |
| 11:08.15 | salz212 | exten => h,1,Hangup() |
| 11:08.16 | salz212 | exten => t,1,Hangup() |
| 11:08.36 | atan | Or pastie.org :-) |
| 11:08.40 | salz212 | apologies |
| 11:08.45 | kaldemar | hrolf: it seems to say the same, but if it works for you... |
| 11:08.46 | kaldemar | salz212: don't paste here, use a pastebin. |
| 11:08.46 | kaldemar | ~pb |
| 11:08.46 | 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.us , 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. |
| 11:08.49 | hrolf | kaldemar: How do we get the unique call ID for a call? ${UNIQUEID} changes (but I don't know when) |
| 11:09.12 | hrolf | kaldemar: and it is usually the same minus the . 'dot' part |
| 11:09.46 | salz212 | here http://pastecode.com/a1 |
| 11:11.55 | salz212 | so the problem is even when the number is busy or no answer or even congested with truck one, it goes to failover which I do not want. |
| 11:11.58 | kaldemar | hrolf: what id are you talking about? |
| 11:12.29 | hrolf | kaldemar: Is there any sort of identifier which uniquely identifies the call? |
| 11:13.49 | kaldemar | hrolf: if you don't find UNIQUEID usable, see what function CHANNEL offers. |
| 11:16.00 | hrolf | kaldemar: ${CHANNEL} is the name of the current channel. Do you suggest I combine these two? |
| 11:17.22 | kaldemar | hrolf: not the CHANNEL variable, but CHANNEL() function. |
| 11:18.02 | kaldemar | howver, UNIQUE should be unique, i don't see why you couldn't use it. |
| 11:18.14 | *** part/#asterisk maldous (~root@124-148-129-91.dyn.iinet.net.au) |
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| 11:21.21 | *** part/#asterisk m_tadeu (~quassel@89-180-136-234.net.novis.pt) |
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| 11:23.31 | *** join/#asterisk maldous (~root@124-148-129-91.dyn.iinet.net.au) |
| 11:25.52 | maldous | can someone tell me if http://pastebin.ca/2087094 should work (extensions.conf) - I don't seem to be able to get tones to be recognised/received. |
| 11:29.59 | maldous | ah. dtmfmode=rfc2833 fixed it. |
| 11:33.42 | *** join/#asterisk imox (~imox@p4FC5C79A.dip0.t-ipconnect.de) |
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| 11:58.52 | zamba | which distro do you guys recommend for running asterisk? |
| 11:59.01 | zamba | without letting this turn into a flame war :) |
| 11:59.13 | zamba | but basically what i'm looking for is updated packages |
| 12:01.12 | ollii | why not using asterisk from source? |
| 12:01.40 | leifmadsen | zamba: ubuntu or centos |
| 12:02.47 | zamba | leifmadsen: which has the most updated version? |
| 12:02.51 | zamba | ollii: nah, too much hassle |
| 12:03.47 | zamba | running asterisk is just one small bit of running the whole it infrastructure.. we can't be experts in every field all the time and follow every mailing list looking for new bugs and then upgrading.. better to just upgrade asterisk with the rest of the system |
| 12:04.33 | zamba | but of course, in an ideal world :) |
| 12:10.12 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:16.57 | maldous | does ringing() take arguments? |
| 12:17.05 | maldous | I want to ring for 30 seconds and then answer.. |
| 12:18.03 | atheos | maldous - ringing() followed by wait(30) |
| 12:19.00 | maldous | ah |
| 12:19.38 | maldous | thx |
| 12:20.12 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:30.39 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:34.32 | russellb | zamba: asterisk.org provides repos for both ubuntu and centos (5 at least) |
| 12:34.51 | russellb | zamba: though if you use centos 6, asterisk is in the EPEL repo, and stays up to date there |
| 12:37.11 | zamba | russellb: oh! sweet |
| 12:38.10 | zamba | will the upgrade from 1.6 to 1.8 be seamless? |
| 12:39.07 | russellb | you'll need to read the UPGRADE.txt file in 1.8 |
| 12:39.42 | [TK]D-Fender | zamba, seams have been deprecated and replaced with zippers but will be maintained through the next version... |
| 12:43.57 | maldous | is there a "keep-alive" option somewhere I can enable? |
| 12:44.18 | maldous | my asterisk-as-a-sip-client works for a while, but then stops receiving events some time later. |
| 12:45.02 | [TK]D-Fender | maldous, "qualify" |
| 12:46.50 | maldous | thx |
| 12:49.20 | *** join/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 12:49.30 | *** part/#asterisk robl^laptop (~robl@pdpc/supporter/active/robl) |
| 12:53.27 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:54.17 | maldous | no luck |
| 12:54.33 | maldous | sip debug enabled. i don't see asterisk doing anything to keep it alive. |
| 13:05.23 | *** join/#asterisk serafie (~erin@nat/digium/x-ajcaaxbokoowbrex) |
| 13:11.49 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 13:14.00 | Katty | drags in |
| 13:14.20 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:15.34 | [TK]D-Fender | maldous, Show us what you've done |
| 13:16.18 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-219-172.w86-204.abo.wanadoo.fr) |
| 13:17.08 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 13:17.14 | merlin8282 | Hello. Do you know if a *list* of asterisk sounds (not the sounds themselves, but a list of them) does exist for 1.8 ? |
| 13:21.11 | kaldemar | merlin8282: what kind of a list are you looking for? |
| 13:21.32 | merlin8282 | something like this http://web.archive.org/web/20100126041204/http://www.nathanpralle.com/software/ast_soundlist.html |
| 13:21.42 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 13:30.36 | merlin8282 | kaldemar: maybe I should have a look at the scripts here https://wiki.asterisk.org/wiki/display/AST/About+the+Sounds+Tools ? |
| 13:35.57 | merlin8282 | kaldemar: mmm, nothing in the scripts. |
| 13:38.10 | *** join/#asterisk VoipForces (~Adium@69.165.199.195) |
| 13:39.30 | VoipForces | Hi all, on linksys phones SPA941 phones, any way to know if the phone is in local DND mode ? |
| 13:39.55 | p3nguin | Look at the display. |
| 13:41.00 | VoipForces | I mean remotly… Have freaking remote support agent that put themself in DND… Looks like a BUSY extension as far as asterisk is concerned. |
| 13:42.17 | leifmadsen | yep, that's how it would show up |
| 13:42.47 | leifmadsen | unless the phone has some sort of API you can interact with, Asterisk won't be able to tell if it is in DND mode, or what. The response coming back from the phoen would just be a 486 Busy Here |
| 13:42.54 | *** join/#asterisk wonderworld (~ww@port-92-201-254-65.dynamic.qsc.de) |
| 13:43.00 | leifmadsen | so the only way is to look at the phone |
| 13:43.13 | cusco | hi |
| 13:43.17 | cusco | I just compiled asterisk |
| 13:43.21 | leifmadsen | \o/ |
| 13:43.26 | DND | someone called ma? |
| 13:43.26 | cusco | and starting it says: illegal instruction |
| 13:43.28 | DND | *me |
| 13:43.32 | cusco | :( |
| 13:43.32 | DND | lol |
| 13:43.36 | VoipForces | Ok, then we will ask each remote agent to open a port so we can disable the DND in their phone... |
| 13:43.43 | DND | another one |
| 13:43.48 | DND | :D |
| 13:44.09 | singler | :D |
| 13:44.10 | leifmadsen | DND: that's almost as bad a moniker as file :) |
| 13:44.49 | DND | btw are you configuring spa941 manually? |
| 13:45.16 | DND | oh this is the asterisk channel.. all command lines here :D |
| 13:45.32 | VoipForces | DND: Unfortunatly yes, this was before I came on board and added central provisionning... |
| 13:46.04 | DND | im using asterisknow with endpoint manager module but hell it seems its not being updated. |
| 13:46.27 | cusco | root@velhadas:~# asterisk -vvvvvvvvvvvvvvvvc |
| 13:46.28 | cusco | Illegal instruction |
| 13:46.40 | VoipForces | DND: I build my own asterisk with freePBX works great. What version of freePBX do you have? |
| 13:46.40 | merlin8282 | mmm, so nobody has an idea of how I could even generate myself a list of prompts that are effectively used on my * ? |
| 13:46.41 | cusco | a clean install... |
| 13:46.53 | cusco | merlin8282: dialplan show |
| 13:47.13 | DND | the module can detect the phones but it wont change the settings |
| 13:47.16 | merlin8282 | cusco: ok, but what about all prompts used by for example voicemail ? |
| 13:47.19 | DND | im using *now 1.5 |
| 13:47.26 | DND | but its updated frequently |
| 13:47.39 | VoipForces | DND: you mean it does not update the files in /tftpboot ? |
| 13:50.38 | cusco | merlin8282: do you have a context for voicemail? |
| 13:50.43 | cusco | if yes, dialplan show voicemail |
| 13:51.13 | cusco | check in your dialplan where do you use the voicemail feature |
| 13:52.00 | [TK]D-Fender | cusco, No. He wants a list of the sounds files with the text of what is in them |
| 13:52.17 | *** join/#asterisk jeffspeff (~jeffspeff@173-11-144-149-houston.txt.hfc.comcastbusiness.net) |
| 13:52.31 | cusco | ow... speech2text ! |
| 13:52.47 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 13:52.55 | [TK]D-Fender | cusco, No, he wasn't asking for a conversion. He just wants a list. |
| 13:52.58 | merlin8282 | [TK]D-Fender: or at least a list of the files that are used at all. |
| 13:53.08 | merlin8282 | (with the text it's a plus) |
| 13:53.12 | devil_evoxxx | hi to all, again :9 |
| 13:53.15 | devil_evoxxx | hi to all, again :) |
| 13:53.53 | cusco | hi |
| 13:54.15 | cusco | I just compiled 1.8.7.0 and I get Illegal instruction |
| 13:55.36 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 13:55.36 | zamba | i believe i've seen that list somewhere |
| 13:55.37 | jeffspeff | i'm having a horrible brain fart trying to get this to work. it keeps trying to go to the 's' extension rather than receive the extension typed in. http://pastebin.com/zvLkr4Wb |
| 13:56.23 | cusco | normally that happens if the extension it tries to go does not exist in the context |
| 13:56.32 | [TK]D-Fender | ... |
| 13:56.33 | [TK]D-Fender | no |
| 13:56.57 | zamba | merlin8282: http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 13:57.12 | cusco | ok don't listen to me... |
| 13:57.13 | [TK]D-Fender | jeffspeff, You showed us a context with only 1 exten in it. We don't see what you are expecting to have land on that however |
| 13:57.45 | [TK]D-Fender | "s" is not a catch-all |
| 13:58.00 | jeffspeff | my apologies, i posted the wrong one, i've been messing with for trial and error. http://pastebin.com/SGfZ47m5 explains more like what i'm trying. |
| 13:58.36 | [TK]D-Fender | jeffspeff, exten=s,n,Voicemail(${EXTEN}@star) <-- ${EXTEN} is "s" because that is the line you are in. |
| 13:58.47 | jeffspeff | i'm just wanting a user to be able to type *123 on a phone and it go to this context, then they can type an extension number and go straight to that persons vmail like they unavailable |
| 13:59.06 | [TK]D-Fender | jeffspeff, It is never meant to hold "some number that may have been involved at the start of the call" |
| 13:59.27 | jeffspeff | [TK]D-Fender, so how would i accomplish this? |
| 13:59.43 | [TK]D-Fender | jeffspeff, ${CALLERID(dnid)} |
| 13:59.58 | cusco | ${EXTEN} doesn't seem valid ('s') ? |
| 14:00.02 | [TK]D-Fender | jeffspeff, or pust it into another var at an earlier point |
| 14:00.49 | jeffspeff | [TK]D-Fender, so will the callerid dnid be set to the *123 that the user typed in to get in that context or would it be set to what the user presses once in that context? |
| 14:02.18 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:02.18 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:06.47 | *** join/#asterisk master_of_master (~master_of@p57B54035.dip.t-dialin.net) |
| 14:06.59 | *** join/#asterisk dirkD (~dirk@84-245-20-6.dsl.cambrium.nl) |
| 14:08.07 | [TK]D-Fender | jeffspeff, it isn't "what thy typed to get into the context". it's what they dialed only at the very start of the call |
| 14:08.49 | [TK]D-Fender | jeffspeff, and once "in" the context? So far that doesn't look like an IVR where they should by typing anything |
| 14:09.05 | [TK]D-Fender | jeffspeff, So I'm not sure what you are tying to refer to there... |
| 14:11.24 | jeffspeff | here's what i'm trying to accomplish... call starts by user dialing *123 (or whatever exten is) -> call goes to [somecontext] -> caller hears a beep -> caller dials some numbers -> caller is then directed to the voicemailbox (to leave a message) of the corresponding user to the numbers typed |
| 14:12.31 | VoipForces | jeffspeff: Maybe you should use the READ function to get the user extension |
| 14:14.16 | jeffspeff | what's happening is our calls go straight to a receptionist who then tries to call the desired employee and do a warm transfer (or whatever you want to call it) if the desired person isn't there and the caller wants to leave a voicemail, then the receptionist must then transfer the caller to the users exten and they have to wait for the hole ringing time-out part before opportunity to leave vm. |
| 14:14.47 | [TK]D-Fender | jeffspeff, Nothing you have shown takes any kind of input. |
| 14:15.32 | [TK]D-Fender | " caller dials some numbers -" <- too vague |
| 14:15.41 | TheCompWiz | so... looking @ asterisk 10... I just noticed that 10 has t.38 gateway functionality.... anyone know how well this works in production? |
| 14:16.13 | *** join/#asterisk jerware (~jerryg@c-71-58-179-44.hsd1.pa.comcast.net) |
| 14:16.16 | jerware | folks. |
| 14:16.30 | [TK]D-Fender | jeffspeff, If you want them to simply choose a box to leave a message in, then yes, use READ() |
| 14:16.42 | [TK]D-Fender | jeffspeff, and pass what they entered to VoiceMail() |
| 14:16.44 | jerware | Is it possible to configure the configuration files and get something up and running and be oblivious to the protocols? |
| 14:16.49 | jeffspeff | [TK]D-Fender, thanks, that's what i'm reading up on now. |
| 14:16.58 | [TK]D-Fender | jerware, .... huh? |
| 14:17.15 | jeffspeff | i'd never heard of Read() before, much appreciated. |
| 14:17.20 | jerware | Is there a layer of ecapsulation from the protocols? |
| 14:17.21 | [TK]D-Fender | jerware, sip.conf is intimately aware that it is for the SIP protocol. |
| 14:17.42 | [TK]D-Fender | jerware, Which config files? Oblivious how? Versus what? |
| 14:18.11 | jerware | Well I just got the asterisk oreilly book. It's not til chapter 8 that they start talking about protocols. |
| 14:18.14 | jerware | and I hate reading. |
| 14:18.27 | VoipForces | jeffspeff: You may want to look into freePBX, would save you a LOT of headake for simple corporate environement. |
| 14:18.37 | [TK]D-Fender | jerware, So go read another guide. |
| 14:18.51 | WIMPy | There are lots of things that behae differently on different channeltypes, unfortunately. |
| 14:19.00 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:19.01 | VoipForces | jeffspeff: All those functions are 'built-in' freePBX. |
| 14:19.09 | [TK]D-Fender | VoipForces, What he was asking for is pretty much manual in there too.. plus the fight of integrating it. |
| 14:19.23 | [TK]D-Fender | VoipForces, His specific request isn't "built -in" |
| 14:19.25 | jerware | freepbx looks like a what I too could use. thanks VoipForces |
| 14:19.32 | jeffspeff | VoipForces, thanks for the info, but i'm not converting to freePBX. |
| 14:20.14 | VoipForces | [TK]D-Fender: Well, yes freePBX had transfer direct to VM and interactive voicemail (*97 vs *98) |
| 14:20.34 | [TK]D-Fender | VoipForces, Not where the person being dumped gets to choose the box... |
| 14:21.01 | [TK]D-Fender | VoipForces, read his request again |
| 14:21.05 | VoipForces | True… |
| 14:21.23 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 14:21.49 | [TK]D-Fender | jerware, What is it you are trying to make "oblivious to protocol"? |
| 14:23.00 | Qwell | [TK]D-Fender: himself. I think his question is "I don't know the SIP RFCs. Do I have to?" |
| 14:23.01 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:23.07 | Qwell | To which the answer is no, you don't |
| 14:23.25 | [TK]D-Fender | Qwell, I'm not thinking that is it.... |
| 14:23.34 | WIMPy | Sounds optimistic |
| 14:23.59 | [TK]D-Fender | Qwell, His entire question, direction, and goal are so generic as to not mean anything as worded... |
| 14:24.02 | Qwell | [TK]D-Fender: <jerware> Is it possible [for me] to configure the configuration files and get something up and running and be oblivious to the protocols? |
| 14:24.11 | Qwell | that's how I read it |
| 14:24.56 | [TK]D-Fender | Qwell, Hard to configure sip.conf and not know it's for SIP :) Now does it mean that he has to understand everything about even SIP RFC to use it? Oh hell no :p |
| 14:25.06 | Qwell | yes, I think that's his question |
| 14:25.39 | r0m|u | [TK]D-Fender, whats going on. |
| 14:26.20 | [TK]D-Fender | r0m|u, Breakfast. Somewhere. By someone. I swear it. Disclaimer: that "someone" would not be me. But it is happening. |
| 14:26.59 | r0m|u | lmao |
| 14:27.01 | r0m|u | nice |
| 14:27.02 | Qwell | that's a hell of an assertion |
| 14:27.17 | jerware | [TK]D-Fender: I just wanna set my phones up. |
| 14:27.26 | r0m|u | by the way I am SeRi just with my nick from work lol |
| 14:28.04 | [TK]D-Fender | jerware, Go right ahead. sip.conf takes about a dozen and a half lines. |
| 14:28.22 | [TK]D-Fender | jerware, then a phone should be able to register. then you work on your dialplan |
| 14:28.44 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-27-NewEngland.hfc.comcastbusiness.net) |
| 14:36.23 | [TK]D-Fender | The dialplan for a simple PBX might have all of 2 occurances of the word "SIP" at all using SIP phones and an ITSP that also uses SIP. |
| 14:36.52 | [TK]D-Fender | Hope that isn't too much considering you don't really have to specify any paramters reall in there even... |
| 14:39.08 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.247.170.dsl.dyn.forthnet.gr) |
| 14:39.17 | *** join/#asterisk celord (~celord@201.191.130.196) |
| 14:40.15 | p3nguin | How can I reopen an issue on jira after it was erroneously closed? |
| 14:40.38 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 14:40.58 | irroot | p3nguin #asterisk-bugs post ASTERISK-XXXX and message to open |
| 14:41.29 | p3nguin | does not understand what irroot is saying. |
| 14:41.54 | eva_02 | Guys, can you advice sip to skype gateway? |
| 14:42.34 | irroot | eva_02 dont bother with Skype i for one have canned skype due to there licencing |
| 14:42.44 | Katty | GUESS WHO"S AWAKE!!!!!11 |
| 14:42.50 | *** join/#asterisk leroybuckingham (43350083@gateway/web/freenode/ip.67.53.0.131) |
| 14:44.44 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:44.50 | coppice | irroot: I still haven't figured out anything you can do with the skype developers kit |
| 14:45.07 | merlin8282 | zamba: thanks, that should be enough :) |
| 14:45.54 | leroybuckingham | Hey guys, I'm having a bizarre problem on this site. When my provider sends a call to the site, it rings my softphone as it should, but for some reason my provider is seeing a congestion and sending a busy signal to the caller. I set up another asterisk system as a trunk so I could watch sip debug from both ends, and I'm seeing something I really don't understand. Immediately after the provider sends the invite, it claims |
| 14:46.42 | [TK]D-Fender | leyJust show us the debug from the call involving the softphone |
| 14:46.48 | [TK]D-Fender | leroybuckingham, Just show us the debug from the call involving the softphone |
| 14:47.12 | leroybuckingham | site with the softphone: http://pastebin.com/LwPr7vdR ... providing trunk: http://pastebin.com/beDvSxZ0 |
| 14:47.14 | Katty | coppice |
| 14:47.16 | Katty | pokes coppice |
| 14:49.22 | irroot | just told a customer to put on a stupid hat and go run round in circles in the street ... network cable broken and he is trying to tftp flash the phone claiming asterisk/tftp is broken |
| 14:49.32 | [TK]D-Fender | leroybuckingham, You only debugged have of one leg on that. |
| 14:50.15 | [TK]D-Fender | leroybuckingham, enable global SIP debug and look at the whole call. |
| 14:50.23 | leroybuckingham | okay |
| 14:51.03 | jeffspeff | [TK]D-Fender, hey thanks again for pointing me in the direction of Read() that did the trick perfectly. |
| 14:51.46 | [TK]D-Fender | leroybuckingham, Also you have configured your peers without specifying a clean list of allowed codecs. Which all things considered should only be 1 each. |
| 14:52.23 | Katty | fender bender. |
| 14:52.29 | *** join/#asterisk CrossWired (~chatzilla@65.210.186.34) |
| 14:52.29 | [TK]D-Fender | Katty, Mew. |
| 14:52.35 | Katty | i'm bored. |
| 14:52.37 | Katty | let's go have drinks. |
| 14:52.47 | [TK]D-Fender | Katty, Commute would suck... hard |
| 14:53.01 | [TK]D-Fender | Katty, Get your ass out of Misery... |
| 14:53.10 | Katty | butbut |
| 14:53.11 | [TK]D-Fender | Katty, err ... Missouri :p |
| 14:53.24 | Katty | i will bribe you wif hugs! |
| 14:53.27 | Katty | and scotch! |
| 14:53.29 | jeffspeff | lol, no Missouri is a Misery |
| 14:53.38 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 14:54.14 | coppice | They could have a campaign to attract new businesses with the slogan "Missouri loves company" |
| 14:54.22 | Katty | well someone needs to come get their Party On with me |
| 14:54.27 | jeffspeff | lmao! |
| 14:54.32 | Katty | today is suckin |
| 14:54.45 | coppice | today is a holiday |
| 14:54.49 | Katty | imma need to drown myself in bubbles with a margarita later |
| 14:54.52 | jeffspeff | it's 10am, too early |
| 14:54.57 | irroot | coppice hate those dev kits/packs/.... that are meant to get you excited but often waste of time |
| 14:54.57 | Katty | pffff |
| 14:55.02 | Katty | it's never too early! |
| 14:55.18 | Qwell | Katty: road trip to Huntsville. go! |
| 14:55.22 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 14:55.32 | Katty | but..that's like...4hours away |
| 14:55.39 | coppice | irroot: the kit itself isn't too bad, but once you read the conditions they seem to ban everything you might want to do |
| 14:56.00 | Qwell | exactly! only 4 hours |
| 14:56.06 | jeffspeff | lol |
| 14:56.14 | Katty | and how am i supposed to drive back home after shenanigans |
| 14:56.24 | Qwell | who says you're allowed to leave? |
| 14:56.26 | jeffspeff | that's what planes are fo |
| 14:56.29 | jeffspeff | *for |
| 14:56.31 | irroot | lol yeah this really cool dev thing can do ....... all denied in t&c in fine print a "apple licence" |
| 14:56.39 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:56.39 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:56.40 | Katty | okay fine, but i'm taking my dog |
| 14:57.47 | Katty | 100lb lap dog ftw |
| 14:58.01 | Qwell | Katty: there will be beer and shenanigans at astricon |
| 14:58.06 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:58.21 | Katty | The_Boy_Wonder: wondderrr boyy, what is the secret of your powaarrr |
| 14:58.26 | leroybuckingham | [TK]D-Fender: Here's the site with all sip debugging enabled: http://pastebin.com/zt5kLESg |
| 14:58.30 | The_Boy_Wonder | pancakes |
| 14:58.47 | Katty | i LOVE pancakes |
| 14:59.13 | Katty | astricon totally needs pancake shenanigans. |
| 14:59.47 | leroybuckingham | And the provider: http://pastebin.com/v9mSLHvV ... this one is kinda noisey though |
| 15:00.13 | Katty | Qwell: i'mma be all grown up friday :> |
| 15:00.19 | Qwell | howso? |
| 15:00.26 | Katty | it's my bifday |
| 15:00.35 | Qwell | oic |
| 15:01.12 | Katty | you're still older than me |
| 15:02.25 | jaytee | he's older than dirt |
| 15:02.35 | [TK]D-Fender | leroybuckingham, This time 101 isn't answering.... |
| 15:02.47 | Katty | no he's not!! |
| 15:02.47 | Qwell | Katty: I wouldn't know. Your facebook doesn't give a year :p |
| 15:03.02 | Katty | hugs jaytee |
| 15:03.05 | leroybuckingham | I know it looks that way |
| 15:03.06 | Katty | jaytee: how're you dear? |
| 15:03.11 | irroot | greetings peeps from durban by the sea |
| 15:03.11 | leroybuckingham | but I get a busy signal immediately when dialing |
| 15:03.18 | [TK]D-Fender | leroybuckingham, And you seemt o be showing 2 kinds fo calls. Please jsut concentrate on a single scenario to fix |
| 15:03.19 | jaytee | I'm havin a 'splodey day |
| 15:03.19 | leroybuckingham | not long enough for 101 to not answer |
| 15:03.24 | jaytee | how're you? |
| 15:03.30 | jeffspeff | so, i've noticed that when my phones qualify with the server it shows their ping times in ms; however what asterisk shows as the ping time is always a very high number and does not reflect the same results when you just ping that same IP. any ideas? |
| 15:03.52 | [TK]D-Fender | jeffspeff, Perfectly normal |
| 15:03.57 | leroybuckingham | I could call in to an announcement or something else and it would still fail, should I do that instead to narrow tihngs down? |
| 15:03.59 | [TK]D-Fender | jeffspeff, Qualify != ping |
| 15:04.28 | jeffspeff | [TK]D-Fender, then where does it get the qualify time from? time between each register? |
| 15:04.43 | Katty | jaytee: good now that i got some caffeine down me. what made your day all splodey? |
| 15:04.44 | p3nguin | Ping is an ICMP Echo. Qualify is SIP OPTIONS. |
| 15:04.47 | [TK]D-Fender | jeffspeff, a Qualify packet is a layer 7 communication and can get prioritized very differently with no reflection on the time to respond to a base ICMP ping |
| 15:04.57 | jaytee | Katty, Hyper-V |
| 15:05.12 | Qwell | jaytee: eww |
| 15:05.17 | [TK]D-Fender | jeffspeff, Polycom's are notably slower in responding to those vs many other phones on the same switch |
| 15:05.58 | [TK]D-Fender | leroybuckingham, I don't see a call being placed to it. |
| 15:06.11 | Katty | jaytee: ohhhh you poor dear |
| 15:06.47 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 15:06.52 | jaytee | Katty, yeah .... but someone's gotta do it :-) |
| 15:07.00 | [TK]D-Fender | leroybuckingham, http://pastebin.com/zt5kLESg <- There is no SIP debug for the call out to SIP/101 |
| 15:07.15 | Katty | jaytee to the rescue. why does that not surprise me. |
| 15:07.24 | Katty | jaytee: cause you're EPIC!!! <3 |
| 15:07.28 | jaytee | I end up wearing too many hats |
| 15:07.35 | Katty | me too |
| 15:07.40 | jaytee | a jackoff of all trades and a master of none :-) |
| 15:07.41 | Katty | but they're usually knit ones, with cute little ears from spencers. |
| 15:07.43 | Naikrovek | hyper-v isn't so bad... |
| 15:07.59 | Naikrovek | it's no vmware, surely, but it suits many |
| 15:08.34 | jaytee | Naikrovek, no not really. just takes a bit getting used to and the virtual networking components are a tad thorny to "conjure" |
| 15:09.25 | Naikrovek | there are sharp edges, yes. but it's not worthy of immediate dismissal anymore |
| 15:09.34 | Naikrovek | windows server8 will make it even better |
| 15:09.40 | jaytee | Qwell, you still smokin or did you finally quit? |
| 15:09.51 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 15:09.57 | Naikrovek | if he doesn't respond right away, he's probably out smoking |
| 15:10.01 | Naikrovek | so there's your answer, maybe |
| 15:10.04 | jaytee | haha |
| 15:10.07 | Qwell | pretty much that |
| 15:10.09 | Qwell | brb, smoke |
| 15:10.14 | Naikrovek | knew it. |
| 15:10.16 | jaytee | I'll join ya! |
| 15:10.41 | jaytee | we've actually taken a smoke break together in real life |
| 15:11.21 | Naikrovek | lol |
| 15:11.42 | Naikrovek | "I smoked with Qwell! SQUEEE!" |
| 15:11.50 | jerware | I want to make sure I have working phones before touching asterisk (So I know where a problem resides if one occurs). Can two sip phones communicate with each other with out a PBX if on the same lan? |
| 15:11.51 | Naikrovek | sorry, hehehe |
| 15:12.16 | Naikrovek | jerware: they often do, but they need a SIP server in between to arrange the communication. |
| 15:12.47 | Naikrovek | the SIP stuff always happens with a SIP server like Asterisk, but the real time audio usually goes directly from phone to phone |
| 15:13.11 | leroybuckingham | [TK]D-Fender: This should be a bit more simple. Forget the softphone, it's calling into an announcement http://pastebin.com/14c1aZrN |
| 15:13.29 | jerware | Naikrovek: Is that how the phone finds its peers. By looking up addressing information via sip ? |
| 15:13.37 | leroybuckingham | so the site is accepting the call, and the provider is still seeing congestion |
| 15:14.02 | Naikrovek | the phones don't find anything; the sip server arranges all the communication. the phones are not peer to peer devices at all, except for call audio, and only then, sometimes. |
| 15:14.47 | jerware | How do I know what my phones need. |
| 15:15.02 | Qwell | ~book |
| 15:15.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 15:15.07 | Qwell | jerware: You aught to start there ^. |
| 15:15.17 | jerware | Yeah I have that book. and it sucks. |
| 15:15.28 | leroybuckingham | [TK]D-Fender: Here's the provider side if it helps: http://pastebin.com/jJxN9NT8 |
| 15:15.39 | [TK]D-Fender | leroybuckingham, Doesn't look like the packet is making it back to the caller.... |
| 15:15.41 | Naikrovek | the sip server knows where the phones are and how to reach them, etc. phone A calls phone B, if they're on the same server, the sip server tells B that a call is incoming, the phone rings, sip server tells A to play the remote ring sound, B answers, sip server tells A that B answered and tells A how to talk to B |
| 15:16.03 | Naikrovek | jerware: your opinion of that book explains your lack of understanding in the matter. it's explained clearly in that book. |
| 15:16.04 | leroybuckingham | thats what i thought too |
| 15:16.05 | [TK]D-Fender | leroybuckingham, Check all of yoru firewalls and routing |
| 15:16.08 | leroybuckingham | but the caller is still seeing "<--- SIP read from UDP:64.105.229.19:5060 --->" |
| 15:16.25 | jaytee | <PROTECTED> |
| 15:16.28 | Qwell | jerware: I'll be sure to let the authors know you think it sucks. |
| 15:16.30 | leroybuckingham | like it's coming in too late or something? |
| 15:16.37 | Qwell | leifmadsen, russellb: ^^^ |
| 15:16.43 | Naikrovek | Qwell: they'll see on their own |
| 15:17.01 | Naikrovek | "it sucks" is a valid opinion. shame it's not based on fact. |
| 15:17.03 | [TK]D-Fender | jerware, The book doesn't suck. You provide no information at all. |
| 15:17.05 | Naikrovek | :P |
| 15:17.18 | leifmadsen | it does suck -- and mighty good I might add |
| 15:17.28 | [TK]D-Fender | jerware, You ask immensely vague questions and give us nothing to help you with |
| 15:17.31 | jaytee | perhaps Asterisk For Dummies would be more his speed? |
| 15:17.35 | Qwell | Proof the book doesn't suck: It uses me in an example. |
| 15:17.46 | jaytee | yay! |
| 15:17.54 | Naikrovek | proof that the book sucks: it does not use me in an example |
| 15:17.54 | [TK]D-Fender | jerware, You ask how to make your phones do something we don;t even normally attempt. And in doing so you still never bothered to even tell us what you have. |
| 15:18.31 | [TK]D-Fender | jerware, Every SIP phone can be different. |
| 15:18.39 | jerware | Oh I have SoundPoint IP501 |
| 15:18.40 | [TK]D-Fender | jerware, Details matter and we have none to go on. |
| 15:18.50 | Naikrovek | sweet |
| 15:18.53 | Naikrovek | polycom phones rule |
| 15:18.59 | jeffspeff | lol |
| 15:19.17 | jaytee | I <3 my Polycoms |
| 15:19.27 | Naikrovek | agreed |
| 15:19.36 | Naikrovek | well I don't <3 them but I like them as friends |
| 15:19.43 | leifmadsen | ~polycomsong |
| 15:19.48 | leifmadsen | ~polycom |
| 15:19.48 | infobot | methinks polycom is The Polycom Song by dialing sip:polycom@leifmadsen.com or ISN 7659*460. Polycom phone are devices that are favoured by much of the community and range in price from under $100 and upwards. |
| 15:19.48 | irroot | hates polycoms not as flexi as snom |
| 15:19.53 | leifmadsen | ya that ;) |
| 15:19.56 | [TK]D-Fender | jerware, if both phones are running then just dial the IP of the other on the phone. |
| 15:20.21 | Naikrovek | that'll be fun once ipv6 takes off |
| 15:20.31 | p3nguin | You don't have to enable IP dialing on the phones? |
| 15:20.35 | Naikrovek | "you've been on the phone for 5 minutes!" "I'm still dialing!" |
| 15:20.43 | jeffspeff | lmao |
| 15:20.47 | irroot | seriously folks in the US it seems polycoms are "normal" way here in africa/europe seems snoms are more natural |
| 15:20.49 | jaytee | and I like using the FTP provisioning I setup with bash scripts to quickly provision a phone. I was able to setup the provisioning easily by reading a "book that sucks" I'd found. |
| 15:20.52 | [TK]D-Fender | p3nguin, I suspect he's done little on them |
| 15:21.02 | Nugget | heh |
| 15:21.11 | p3nguin | That's my point. I thought you had to enable IP dialing. |
| 15:21.13 | jaytee | telnet |
| 15:21.31 | [TK]D-Fender | p3nguin, IIRC it is possible in stock condition.... |
| 15:21.43 | Naikrovek | it is |
| 15:21.46 | jaytee | hmmm, looks like Nugget turned off his autoresponder |
| 15:21.47 | p3nguin | Good to know. |
| 15:21.58 | [TK]D-Fender | jayNo, it's timed so people can't abuse him |
| 15:21.58 | Katty | NUGGET |
| 15:22.03 | Katty | glomps Nugget |
| 15:22.05 | [TK]D-Fender | jaytee, ^^ |
| 15:22.06 | leroybuckingham | [TK]D-Fender: As for firewall, the box at the site is connected directly to the ISP. The site I'm using for the trunk I'm using to test a lot of other sites so I know it works |
| 15:22.22 | p3nguin | People still use telnet? |
| 15:22.30 | Naikrovek | lol yep |
| 15:22.44 | [TK]D-Fender | leroybuckingham, that does not sound like the kind of conclusive research I would consider stopping for |
| 15:23.20 | jaytee | I could see some using telnet in an isolated network like Livermore Labs |
| 15:23.38 | Naikrovek | That's Lawrence Livermore Labs to you! |
| 15:23.43 | jaytee | any place with an internet connection is just begging for digital sodomy |
| 15:25.50 | Katty | dances |
| 15:26.26 | GreatSUN | re |
| 15:26.33 | [TK]D-Fender | jerware, Also this test of yours will likely not prove of any value in getting them working with your PBX. |
| 15:26.51 | Naikrovek | watches that latest futurama with the different animation styles and laughs derisively at anyone that doesn't like it |
| 15:26.59 | Nugget | hihi katty |
| 15:27.00 | [TK]D-Fender | ~jerjerguide |
| 15:27.00 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 15:27.12 | [TK]D-Fender | jerware, ^^^^ |
| 15:27.14 | Naikrovek | [TK]D-Fender: he wants to make sure the phones work. cart before the horse in my eyes, but okay. |
| 15:27.18 | Katty | yay Nugget <3 |
| 15:27.21 | Katty | Nugget: how're you dear |
| 15:27.21 | [TK]D-Fender | jerware, a small and relatively complete sample |
| 15:27.25 | Nugget | life is good |
| 15:27.29 | jeffspeff | if i comma seperate values defined in a variable and then later call that variable in a GotoIf() will the variable see the commas as like an 'or' statment? |
| 15:27.30 | Katty | woo!!! |
| 15:27.35 | Katty | dances with Nugget |
| 15:28.11 | [TK]D-Fender | jerware, Clearly a little dated, but 95% syntax-correct |
| 15:28.41 | [TK]D-Fender | jeffspeff, "," is not "or" |
| 15:29.22 | [TK]D-Fender | unless you're referring to the post ? |
| 15:29.24 | [TK]D-Fender | "?" |
| 15:29.35 | jeffspeff | [TK]D-Fender, just thought it might be due to how it handles '&' |
| 15:31.43 | *** join/#asterisk Freeaqingme (~dolf@83.232.96.217) |
| 15:32.04 | jeffspeff | [TK]D-Fender, what im trying to accomplish is to have phones from locationA to set callerid(num)=somenumber and locationB to set callerid(num)=someothernumber when dialing outbound, both locations use the same outbound sip context. |
| 15:32.30 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-wlehcouaekhnqzsx) |
| 15:32.30 | *** mode/#asterisk [+o mnicholson] by ChanServ |
| 15:32.58 | p3nguin | That shouldn't be too hard. |
| 15:33.34 | p3nguin | You could give each location its own "outbound" context where you Set(CALLERID(num)=123) before the Dial(). |
| 15:34.16 | p3nguin | outbound-a and outbound-b, for example. |
| 15:34.23 | carrar | w00t! |
| 15:35.29 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 15:35.37 | leroybuckingham | [TK]D-Fender: Okay, I used netcat to confirm that either site can connect & write to either other sites 5060 and RTP ranges |
| 15:35.38 | jeffspeff | how would i change the outbound context on those certain extensions? here's my dialplan showing how everything is currently related http://pastebin.com/ye7q8rJY |
| 15:35.42 | leroybuckingham | does that rule out firewall issues? |
| 15:39.00 | leroybuckingham | It's succeeding at the TRYING message but for some reason the 200 OK message isn't going through. does that sound like packet filtering? They have a pretty shitty ISP with their own voip service--maybe that's just paranoia talking though |
| 15:39.04 | p3nguin | You change the context of extensions by putting the extensions within a context which is different. |
| 15:40.20 | p3nguin | That extensions.conf actually works? Amazing! |
| 15:41.13 | jeffspeff | p3nguin, it's a bit unorganized, and it's a work in progress, but yes it works perfectly |
| 15:41.20 | Madkiss | hm. I am seeing a funny effect on my calls. Whenever I start a phone call, the call has a decent audio quality at the beginning of the call, and after a certain time frame (which varies), sound start to be choppy |
| 15:41.26 | p3nguin | Totally amazing. |
| 15:41.46 | *** join/#asterisk fobus912 (~fobus912@41.141.248.71) |
| 15:41.47 | jeffspeff | p3nguin, why is that so amazing? |
| 15:41.56 | fobus912 | Hi All |
| 15:42.10 | p3nguin | I guess writer of pbx_config included a certain level of forgiveness in the code. |
| 15:42.29 | fobus912 | Can please someone help in the setup of directmedia between two endpoint on the same localnetwork |
| 15:43.17 | fobus912 | Is it enough to have directmediapermit=yes |
| 15:43.20 | fobus912 | ? |
| 15:43.32 | p3nguin | Your exten=s,1,NoOp() vs. a normal exten => s,1,NoOp() |
| 15:43.37 | Freeaqingme | if one was asked to give an indication of the stability of the beta1 of 10.0, what would it be? |
| 15:43.48 | p3nguin | Your include=something vs. a normal include => something |
| 15:44.20 | fobus912 | can please someone advice if possible ? |
| 15:44.25 | jeffspeff | using "=" instead of "=>" works and makes it a lot easier to read |
| 15:44.25 | *** join/#asterisk bis0n (~56416@css35-2-78-238-86-105.fbx.proxad.net) |
| 15:44.49 | [TK]D-Fender | p3nguin, "=" and "=>" have always been interchangeable |
| 15:44.53 | p3nguin | Be glad there is an apparent level of forgiveness. |
| 15:44.57 | p3nguin | And those numbered priorities are asking for grief later on when you start changing things. |
| 15:45.10 | [TK]D-Fender | fobus912, directmedia=yes |
| 15:45.31 | fobus912 | I have tried it without any luck [TK]D-Fender, Thank your input |
| 15:45.46 | bis0n | hi, have an asterisk, with an minimal sip.conf file, bue I sip client on local network (or not...) no success |
| 15:46.03 | bis0n | I don't understand the problem |
| 15:46.09 | p3nguin | I also don't see how squishing it all together could possibly make it "a lot easier to read." |
| 15:46.18 | [TK]D-Fender | bis0n, Neither do we. what action is failing? |
| 15:46.42 | [TK]D-Fender | jeffspeff, there is much that could be done to clean that all up. |
| 15:47.08 | bis0n | [TK]D-Fender, all action, I always get "reuest timeout" |
| 15:47.29 | p3nguin | So... back to your question of how to change the context of extensions... just move the extensions to another context. |
| 15:47.49 | leroybuckingham | alright, you're going to laugh at this, but I pushed the OK message to the provider through netcat while dialing from the provider, and it managed to complete the call, once out of several attempts |
| 15:47.51 | [TK]D-Fender | bis0n, Please provide a proper and complete description of the networking involved in the things * is supposed to be talking to and a description of what you installed it on. |
| 15:48.11 | jeffspeff | [TK]D-Fender, any ideas on how to get my 911 outbound to maintain the same functionality but coded much cleaner? |
| 15:48.17 | bis0n | ok, I will make patebin |
| 15:48.44 | fobus912 | [TK]D-Fender After reviewing my sip conf i was doing something stupid i had both directmedia = yes and directmedia = no |
| 15:48.47 | p3nguin | For what you've given me so far, I'd duplicate the "outbound" extensions into a new context, and append -a and -b respective to your locations. I'd set callerid num accordingly for each context. |
| 15:48.56 | fobus912 | Really Stupid Thank you |
| 15:49.23 | [TK]D-Fender | jeffspeff, You are effectively blocking only 7 people from 911. yet you specified Gotoif for tons more |
| 15:49.39 | catphish | are "one way audio" problems common with a nat'd sip client and an unnatted asterisk server (assuming nat=yes and canreinvite=nevereverever)? |
| 15:49.53 | bis0n | this is my sip.conf file: http://pastebin.com/whHM7x6K |
| 15:50.03 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-219-172.w86-204.abo.wanadoo.fr) |
| 15:50.06 | p3nguin | Or, alternatively, I'd set an explicit caller id number for every single phone in both locations using setvar, then refer to that in the outbound extensions. |
| 15:50.07 | [TK]D-Fender | catphish, Never allow reinvites to the outside world |
| 15:50.13 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:50.15 | jeffspeff | [TK]D-Fender, the purpose of that wasn't to block anybody, but to direct 2 different groups of extensions to dial 911 out through 2 different contexts |
| 15:50.21 | bis0n | the server adress on local network is 192.168.0.4 |
| 15:50.30 | [TK]D-Fender | jeffspeff, out different trunks. |
| 15:50.31 | catphish | [TK]D-Fender: i never allow reinvites at all |
| 15:50.41 | jeffspeff | [TK]D-Fender, yes |
| 15:50.53 | bis0n | when i try to connect with an sip client on local network |
| 15:50.58 | bis0n | he can't connect |
| 15:51.05 | p3nguin | As an alternative to my alternative, I'd use a template for each location and then apply the template to phones as needed. |
| 15:51.36 | [TK]D-Fender | jeffspeff, SetVar=trunkfor911=teliax-admin-911 |
| 15:51.56 | p3nguin | But what do I know? |
| 15:51.58 | [TK]D-Fender | jeffspeff, in your sip peers |
| 15:52.10 | [TK]D-Fender | jeffspeff, jeffspeff, Dial(SIP/${trunkfor911}/911) |
| 15:52.41 | [TK]D-Fender | jeffspeff, And yes, you probably should be putting them in different contexts to begin with |
| 15:52.49 | p3nguin | Too easy! Give him something to make it complicated. |
| 15:53.03 | [TK]D-Fender | jeffspeff, these are different sites I'm suspecting. They should be in separate contexts, not all jumbled together |
| 15:53.21 | p3nguin | That's what I said. |
| 15:53.24 | [TK]D-Fender | p3nguin, I compromised... an easy way to do it the hard way :p |
| 15:54.04 | p3nguin | Some people prefer it hard. |
| 15:54.08 | [TK]D-Fender | bis0n, Check your firewalls, and enable SIP debug from * CLI and look at the attempt |
| 15:54.09 | p3nguin | looks around and waits |
| 15:54.40 | jeffspeff | if i were to put them in seperate context the includes that would be necessary to allow extension dialling between the locations would eventually bridge the contexts to behave as 1 wouldn't it? |
| 15:54.47 | [TK]D-Fender | p3nguin .... |
| 15:54.52 | [TK]D-Fender | That's what SHE said :p |
| 15:54.56 | p3nguin | :D |
| 15:54.59 | bis0n | [TK]D-Fender, i have CLI but nothing appears on... my firewall is ok, iptables, I open all possibilities on 5060. |
| 15:55.23 | [TK]D-Fender | jeffspeff, "include" is your friend and a concept you really need to understand about the dialplan. |
| 15:56.09 | [TK]D-Fender | bis0n, enable sip debug. "sip set debug on" and dump the firewall. I am not trusting that it is set up right on that description alone |
| 15:58.12 | bis0n | o such command 'sip set debug on' (type 'core show help sip set' for other possible commands) |
| 15:58.43 | p3nguin | Someone forgot to load chan_sip.so |
| 15:59.36 | [TK]D-Fender | p3nguin, Horrible possibility |
| 15:59.44 | [TK]D-Fender | bis0n, "core show modules" |
| 15:59.54 | p3nguin | module show like sip |
| 16:00.58 | bis0n | No such command 'core show modules' :/ |
| 16:01.06 | leifmadsen | modules show |
| 16:01.08 | p3nguin | (1059.53) <p3nguin> module show like sip |
| 16:01.08 | leifmadsen | not 'core show modules' |
| 16:01.18 | Katty | hi leif |
| 16:01.21 | leifmadsen | Katty: ohai |
| 16:01.47 | bis0n | http://pastebin.com/mWPVbMnJ |
| 16:02.06 | p3nguin | (1101.08) <p3nguin> (1059.53) <p3nguin> module show like sip |
| 16:02.21 | [TK]D-Fender | bis0n, Looking like chan_sip.so isn't even loaded as p3nguinsuspected |
| 16:02.27 | Madkiss | what happened to dahdi_dummy? |
| 16:02.32 | [TK]D-Fender | bis0n, module load chan_sip.so |
| 16:02.52 | Madkiss | is that part of dahdi now? also, is there something I need to pay special attention to when running asterisk inside a VM? |
| 16:02.54 | p3nguin | madkiss: It is gone, and not dahdi provides the timing. |
| 16:02.58 | bis0n | with lsmod? |
| 16:03.03 | p3nguin | s/not/now/ |
| 16:03.04 | [TK]D-Fender | bi* CLI |
| 16:03.08 | [TK]D-Fender | bis0n, * CLI |
| 16:03.09 | Madkiss | okay |
| 16:03.23 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 16:03.41 | bis0n | [TK]D-Fender, ok but I don't find the rifht command for list modules |
| 16:03.49 | p3nguin | lsmod lists kernel modules. If you can't run module show like sip, how on earth are you going to be able to figure out what lsmod is doing? |
| 16:03.51 | [TK]D-Fender | biYou did, you gave it to us |
| 16:03.58 | [TK]D-Fender | bis0n, Now run the command I just gave you |
| 16:04.18 | Madkiss | I am really having this strange asterisk problem. Sound on meetme conferences (but also on normal SIP calls) is quite choppy, and I wonder what might be the reason for that. |
| 16:04.41 | Freeaqingme | define 'choppy'? |
| 16:04.58 | p3nguin | Mr. Roboto? |
| 16:05.16 | Madkiss | well, i hear a little bit of sound, then silence, then a little bit of sound, then silence, but the other party is totally ununderstandable that way |
| 16:05.19 | bis0n | module load sip_chan Unable to load module sip_chan Command 'module load sip_chan' failed |
| 16:05.29 | Madkiss | bis0n: it's chan_sip |
| 16:05.38 | p3nguin | (1102.32) <[TK]D-Fender> bis0n, module load chan_sip.so |
| 16:05.56 | bis0n | oops thx i try |
| 16:06.23 | bis0n | == Parsing '/etc/asterisk/sip.conf': == Found :) okay I make an test |
| 16:06.25 | Madkiss | Freeaqingme: it seems to be the worst of all in MeetMe conferences, appears to be a little bit better in direct sip calls |
| 16:06.44 | [TK]D-Fender | bis0n, apstebin your modules.conf |
| 16:08.09 | bis0n | http://pastebin.com/2qpFEDTD |
| 16:09.44 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 16:11.26 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176001555.dsl.bell.ca) |
| 16:13.01 | bis0n | logger.c:675 reload_logger: Unable to create queue log: Permission denied |
| 16:13.24 | bis0n | when I make "module reload" |
| 16:13.39 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:15.24 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 16:17.12 | bis0n | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) //// wtf |
| 16:17.26 | TheCompWiz | asterisk isn't running. |
| 16:17.37 | fobus912 | bis0n try asterisk -c |
| 16:17.53 | p3nguin | Try starting asterisk before trying to connect to it. |
| 16:19.11 | bis0n | http://pastebin.com/fYk588N9 |
| 16:19.27 | bis0n | p3nguin, thx but not start... ^^ |
| 16:26.44 | leroybuckingham | [TK]D-Fender: I'm sure you're sick of me by now ;) But can you think of anything at the network level that would cause this same call to succeed one out of 30ish attempts? |
| 16:27.03 | [TK]D-Fender | leroybuckingham, Firewalls |
| 16:35.02 | bis0n | http://pastebin.com/c4tpuysx |
| 16:35.05 | jaytee | OMG! new research into Alzhiemer's indicates it may be transmissible and caused by an infection that is similar to prions like with Mad Cow |
| 16:35.30 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:36.07 | [TK]D-Fender | jaytee, Phew! We're safe. Men can't catch mad-cow disease .... because we're all PIGS :p |
| 16:36.14 | jaytee | hahaha |
| 16:36.29 | jaytee | "Denny Crane!" |
| 16:41.14 | fobus912 | bis0n were you able to start asterisk ?!! |
| 16:41.53 | bis0n | no :/ |
| 16:42.05 | fobus912 | Even with " asterisk -c " |
| 16:42.19 | fobus912 | yesterday i had the same issue i was able to connect using asterisk -c |
| 16:43.26 | Madkiss | Freeaqingme: Are these timing issues possibly? I am running this thing inside a KVM virtual machine, I wonder whether ztdummy will work appropriately in there |
| 16:43.37 | bis0n | ok, have add noload app_voicemail.so and app_voicemai_odbc.so to modules, asterisk start now... |
| 16:46.10 | [TK]D-Fender | bis0n, Keep the one VM module that you will be using |
| 16:47.08 | bis0n | for the moment always have 408 Request Timeout vwith my sip client |
| 16:47.21 | bis0n | voicemail is not my priority... |
| 16:48.18 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 16:48.27 | fobus912 | That's Great binbash_ |
| 16:48.29 | anonymouz666 | leifmadsen: 1.8.8.0-rc1 this week? |
| 16:48.36 | fobus912 | thats great bis0n |
| 16:48.47 | leifmadsen | anonymouz666: might be -- it'll be created whenever I get some scripts updated |
| 16:48.56 | bis0n | when i call cli doesnt see anything, and i have an timeout |
| 16:49.21 | anonymouz666 | updates the blitzrage's script |
| 16:49.25 | bis0n | can we make any call test on localhost? |
| 16:50.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:51.39 | [TK]D-Fender | bis0n, dume your firewall |
| 16:52.17 | [TK]D-Fender | dump* |
| 16:52.21 | bis0n | i can't |
| 16:52.28 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 16:52.36 | *** join/#asterisk slidesinger-lt (~jtatum@173-161-172-121-Philadelphia.hfc.comcastbusiness.net) |
| 16:55.51 | jeffspeff | [TK]D-Fender, so, i've changed the dialplan from earlier so that each location has it's own context. I am curious if the way i have the includes will allow them to dial each others extensions, but only go to their assigned outbound context. http://pastebin.com/ccDFsWs7 |
| 16:56.52 | jeffspeff | i guess the real question is whether or not the includes from contextA are inherited by contextB if contextB includes contextA. |
| 16:56.57 | [TK]D-Fender | jeffspeff, None of those Gotoifs make any sense |
| 16:57.06 | [TK]D-Fender | jeffspeff, star-intnl-approved-1=1000 |
| 16:57.07 | bis0n | that's ok!!! |
| 16:57.12 | bis0n | thx u very |
| 16:57.13 | [TK]D-Fender | jeffspeff, exten=_011XXXXXXX!,1,GotoIf($[${CALLERID(num)} = ${star-intnl-approved-1}]?${EXTEN:0},5:${EXTEN:0},2) |
| 16:57.23 | WIMPy | yes, they are |
| 16:57.46 | [TK]D-Fender | jeffspeff, You don't even need any |
| 16:58.07 | Freeaqingme | Madkiss, I've never experienced timing issues in KVM |
| 16:58.19 | [TK]D-Fender | jeffspeff, the point of separate contexts is so that all the ones going one way point to one, and al the other to another. There is nothing "conditional" about your dialplan at that point |
| 16:58.21 | jeffspeff | [TK]D-Fender, that's my way of blocking international calling unless i explicitly allow you to by adding your CIDnum to the var star-intnl-approved. |
| 16:58.24 | Freeaqingme | Madkiss, although I cant tell for sure, I'd put my bet at codec conversions first |
| 16:58.34 | dijib | hey guys. i have an issue where when i call someone the callee hears a beep on the phone every minute or so. here is the pertinent dialplan. http://pastebin.com/3yvgTdvE |
| 16:58.39 | dijib | what would be causing that? |
| 16:58.46 | [TK]D-Fender | jeffspeff, No, you have a constant in there, not a variable. |
| 16:58.52 | [TK]D-Fender | jeffspeff, entirely wrong idea. |
| 16:59.04 | atheos | dijib - a wiretap |
| 16:59.17 | [TK]D-Fender | jeffspeff, remove the setvar, trash the gotoifs and just set up a tiny context for each that only dials out their respective peers |
| 16:59.23 | dijib | its anybody i call. |
| 16:59.37 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:00.05 | jeffspeff | [TK]D-Fender, ok, but what about down towards the bottom where the location contexts are. does that logic work the way i think it does? |
| 17:01.11 | bis0n | :) :) good!!! |
| 17:01.30 | Rroet | [TK]D-Fender: thanks for the callerid hint. It w0rks as I hoped it would. |
| 17:01.54 | [TK]D-Fender | jeffspeff, You still have a ton of checks you should not have |
| 17:02.30 | jeffspeff | [TK]D-Fender, a ton of checks? |
| 17:02.44 | [TK]D-Fender | jeffspeff, exten=_011XXXXXXX!,1,GotoIf($[${CALLERID(num)} = ${star-intnl-approved-1}]?${EXTEN:0},5:${EXTEN:0},2) <- 17-20 |
| 17:02.46 | [TK]D-Fender | for starters... |
| 17:03.12 | zamba | i'm trying to upgrade asterisk and it's now stuck at the following: Removing all DKMS Modules |
| 17:03.18 | zamba | how long should it be stuck there? |
| 17:03.28 | Qwell | zamba: That isn't something Asterisk does. |
| 17:03.45 | zamba | Qwell: i guess it's ubuntu specific |
| 17:03.45 | [TK]D-Fender | jeffspeff, [star-admin-outbound] should not contain international at all. This is the start of the heirarchial error |
| 17:04.03 | Qwell | zamba: Asterisk doesn't have kernel modules. You're not just upgrading Asterisk. |
| 17:04.10 | [TK]D-Fender | jeffspeff, And repeats with the outbound context below it as well |
| 17:04.40 | jeffspeff | [TK]D-Fender, those are 3 different outbound contexts for 3 different locations |
| 17:05.05 | [TK]D-Fender | jeffspeff, Ok, the idea is not coming acroos.. Break them up. You should not have those levels mixed together |
| 17:05.17 | *** join/#asterisk dms (~dms@nat/digium/x-vrozieyfcxfxcgfg) |
| 17:05.19 | [TK]D-Fender | jeffspeff, geernal outbound is differnt from international. break them up |
| 17:06.18 | bis0n | Unable to create channel of type 'DAHDI' |
| 17:06.21 | bis0n | ?? |
| 17:06.25 | jeffspeff | ok, so internation should be specefied once instead of repeated in each one. |
| 17:06.33 | jeffspeff | *international |
| 17:07.12 | Madkiss | Freeaqingme: ahum. is there a chance to make sure that doesn't happen? i.e. can I make asterisk enforce the usage of a certain codec? |
| 17:07.37 | [TK]D-Fender | jeffspeff, all of the conditional bits in those contexts should be in their own without condition. |
| 17:07.58 | dijib | anybody have any clue what this beep the callee is hearing? |
| 17:09.03 | jeffspeff | are the includes from contextA are inherited by contextB if contextB includes contextA? |
| 17:09.23 | jeffspeff | *are the includes from contextA inherited by contextB if contextB includes contextA? |
| 17:09.23 | [TK]D-Fender | jeffspeff, yes |
| 17:09.32 | [TK]D-Fender | jeffspeff, it is transitive |
| 17:09.38 | [TK]D-Fender | jeffspeff, that is the point. |
| 17:10.20 | [TK]D-Fender | jeffspeff, so no more gotoif's. break apart the pieces and make composite INCLUDE-only ones for the combinations that matter and assign those to your phones. |
| 17:10.29 | Freeaqingme | Madkiss, sorry for the slow responses. If you configure the same codec everywhere, and configure one only, all devices should be forced to use the same |
| 17:11.12 | dijib | atheos, what did you mean a wiretap? |
| 17:11.39 | atheos | dijib FBI? |
| 17:11.40 | zamba | after upgrading to 1.8 i get the following error: [2011-10-05 19:11:19] WARNING[13498]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (privat, 8150, 5) |
| 17:11.51 | jeffspeff | so, i have [star-metairie], [star-admin] and [star-main]. how do i include the extensions of each of those between themselves, but specify different outbound contexts? |
| 17:12.15 | [TK]D-Fender | jeffspeff, You need more contexts |
| 17:13.30 | dijib | im in canada so it would be Cesis but i doubt they have a bug on this sip did |
| 17:13.49 | dijib | did anybody see anything funny in the dialplan that would cause the callee the have a beep? |
| 17:13.49 | Madkiss | Freeaqingme: i iwll give that a try |
| 17:13.57 | leifmadsen | dijib: CSIS |
| 17:13.58 | [TK]D-Fender | dijib, CSIS |
| 17:14.01 | leifmadsen | :) |
| 17:14.04 | dijib | thats the one |
| 17:14.16 | Madkiss | Freeaqingme: I don't actually see high sysload on the system when the problems occur, but I will try that anyway |
| 17:14.28 | [TK]D-Fender | Before backtracking I was thinking you just butchered "celcius" |
| 17:15.04 | zamba | where have the meetme application gone? |
| 17:15.07 | zamba | has* |
| 17:15.24 | [TK]D-Fender | Next... why kind of crazy person thinks you'd get a beep on a bugged line? The whole idea is for your spying to not be noticable. |
| 17:16.14 | Rroet | frack me, sip dialout works ;) |
| 17:16.26 | r0m|u | I am with [TK]D-Fender |
| 17:17.08 | [TK]D-Fender | jeffspeff, Your entire dialplan could be chopped by more than half from what I've seen |
| 17:17.33 | bis0n | need put something on dahdi_channels conf for use an x100p card? |
| 17:17.43 | jeffspeff | [TK]D-Fender, how do you chop in half by adding more contexts? |
| 17:18.13 | [TK]D-Fender | jeffspeff, making proper macros out of redundant dialplan on a boatlod of yoru extens |
| 17:18.25 | zamba | noone knows where the meetme application has disappeared in 1.8? |
| 17:18.29 | [TK]D-Fender | jeffspeff, and also ditiching all the conditional checks you don't need |
| 17:18.37 | Madkiss | zamba: take a look at ConfBridge I guess |
| 17:18.47 | [TK]D-Fender | zamba, You are lacking DAHDI for it no doubt and it has not been compiled |
| 17:19.00 | zamba | Madkiss: has it been renamed? |
| 17:19.29 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 17:19.32 | jeffspeff | zamba, i agree with [TK]D-Fender, i'm using 1.8.5 and meetme is there |
| 17:19.39 | [TK]D-Fender | zamba, no, confbridge is another solution entirely |
| 17:19.51 | nny | stupid question, how do you define # in a context as an extension? |
| 17:19.51 | Madkiss | zamba: I think ConfBridge is something totally different, which sort of does the same thing. |
| 17:19.52 | zamba | jeffspeff: but i'm installing from a debian package |
| 17:19.58 | Rroet | [TK]D-Fender: I'm having 2 lines in my logfile which look odd / suspissious. If I look at the last line, is somebody trying to call via my asterisk? http://pastebin.com/8WMqrLag |
| 17:19.58 | leifmadsen | ConfBridge() probably won't do what you want unless you're using Asterisk 10 |
| 17:19.58 | zamba | Madkiss: what is "best"? :) |
| 17:20.04 | zamba | ah, ok |
| 17:20.13 | WIMPy | nny: Just like any other. |
| 17:20.24 | nny | WIMPy: as I thought, ok the issue is somewhere else, thanks |
| 17:20.25 | paulc | nny: exten => #,1,Command.. |
| 17:20.28 | leifmadsen | exten => #,... |
| 17:20.45 | nny | paulc: leifmadsen yeah heh, just wasn't working, assuming the issue is mine somewhere else |
| 17:21.01 | Katty | sooooo full |
| 17:21.04 | Katty | sprawls |
| 17:21.08 | zamba | installing asterisk-dahdi now, hopefully that will amend the problem |
| 17:21.16 | Katty | food coma imminent |
| 17:21.16 | WIMPy | nny: On may sip devices, # doesn't work as it functions as a send key. |
| 17:21.46 | nny | WIMPy: interesting, actually I am getting Invalid extension '#' in context 'mainmenu' on SIP/vitel-inbound-0000001d so the issue really is something in my dialplan |
| 17:21.56 | Rroet | ohh, before I forget: congrats leifmadsen. (Unesco's world teachers day)... the book is a good learning tool ;) |
| 17:22.04 | WIMPy | nny: yes |
| 17:22.09 | leifmadsen | Rroet: glad you find it useful :) |
| 17:22.14 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 17:22.38 | nny | WIMPy: oddly the # extension is (in it's entirety) exten => #,1,Goto(s,1) in [mainmenu], so I must have goofed seomthing trivial up |
| 17:22.56 | p3nguin | Okay, this is stupid. I ran core set debug channel all to find out what information it gave me... but now I can't turn it back off. |
| 17:23.20 | Rroet | leifmadsen: about the little pastebin I just tossed up. was I right about somebody trying to call out through my asterisk? |
| 17:23.27 | Rroet | http://pastebin.com/8WMqrLag |
| 17:23.27 | leifmadsen | p3nguin: core set debug off |
| 17:23.35 | leifmadsen | Rroet: I didn't look sorry |
| 17:24.13 | Rroet | np, just asking. It's odd as my sip.conf entries all have passwords on it, and I know for sure it wasn't me trying to call at that hour. |
| 17:24.28 | cerienjean | Hello - I am having a weird issue... Asterisk no longer displays error messages when a user tries to register with incorrect credentials... so f2b is not seeing anything... I've made sure via ngrep that the sip dialog is taking place, good credential go to asterisk, so what would cause bad credentials not to be logged anymore (verbose=3) |
| 17:24.57 | nny | Rroet: have you defined the guest context in sip.conf? If so, does your [incoming] context have a way for someone to enter those digits? |
| 17:25.41 | Rroet | nny: I'm not sure.. and second, I wouldn't like a guest to have a way to call international phonecalls through my bpx ;) |
| 17:25.49 | nny | Rroet: guest context/default context. You can set it with context=something in [general] in sip.conf or you can allowguest=no to prevent non defined peers from connecting via sip |
| 17:25.49 | WIMPy | p3nguin: Did you have to mention that? I see things I don't like, now. |
| 17:25.50 | Rroet | allowguest = yes |
| 17:26.06 | bis0n | Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented) |
| 17:26.08 | p3nguin | leifmadsen: Yeah, tried that and tried setting to 0. Doesn't disable the new channel debugging, though. |
| 17:26.20 | nny | Rroet: if that's desired, your guest hit your incoming context and was able to request that "extension" which looks like an international number |
| 17:26.50 | Rroet | I have a allowguest set as yes. I hope, or my intention is, to open up my asterisk bpx for people to reach SIP phones connected to asterisk. but only inbound. not relay. |
| 17:26.59 | WIMPy | bis0n: s/channel/channel type/ |
| 17:27.19 | bis0n | Dial(DAHDI/1/${EXTEN:1},20,r) |
| 17:27.20 | nny | Rroet: yeah increasingly common. As long as that context is secure you're fine I assume |
| 17:27.29 | Rroet | so I got the picture. I know where the context is defined and I need to make sure that incoming will always be terminated at the extensions within the bpx without having an option to route to other numbers. |
| 17:27.39 | bis0n | exten => _6.,1,Dial(DAHDI/1/${EXTEN:1},20,r) |
| 17:27.41 | nny | Rroet: yeah pretty much |
| 17:28.09 | Rroet | sadly it'll lead to increased log entries of people trying to get around that. |
| 17:28.25 | Katty | hi bison |
| 17:28.50 | bis0n | o/ |
| 17:29.39 | Katty | how're you hunny |
| 17:29.40 | [TK]D-Fender | <Rroet> [TK]D-Fender: I'm having 2 lines in my logfile which look odd / suspissious. If I look at the last line, is somebody trying to call via my asterisk? http://pastebin.com/8WMqrLag <- could be a hack attempt |
| 17:29.47 | p3nguin | rroet: Typically, if you allow anonymous calls (allowguest=yes in sip.conf), you should be very careful with what context you set in the general section of sip.conf, because that's where all anonymous calls will go. |
| 17:30.07 | p3nguin | rroet: You should never have the ability to dial back out, at least not directly, in that general context. |
| 17:30.08 | Rroet | context was set to an empty incoming for now |
| 17:30.40 | p3nguin | wimpy: Now if there were only some way to turn that back off, we'd be in business. |
| 17:30.51 | Rroet | I'm going to rename it to www-incoming to know it's coming from the web and refer it to [internal] which terminates at my 6XXX extensions or voicemailbox |
| 17:30.52 | p3nguin | It's kind of odd that you can turn on debug but not turn it back off. |
| 17:31.12 | [TK]D-Fender | nny, PB your dialplan along with the error |
| 17:31.26 | p3nguin | rroet: Does internal have the ability to dial anywhere else? |
| 17:31.45 | WIMPy | p3nguin: I can't, either. |
| 17:31.58 | [TK]D-Fender | <Rroet> context was set to an empty incoming for now <- excellent start |
| 17:32.10 | Rroet | no. only to the internal softDevices I have in the sip.conf or VoiceMailMain |
| 17:32.29 | nny | [TK]D-Fender: i figured it out, thanks though. |
| 17:34.58 | cerienjean | Hello - I am having a weird issue... Asterisk no longer displays error messages when a user tries to register with incorrect credentials... so f2b is not seeing anything... I've made sure via ngrep that the sip dialog is taking place, good credential go to asterisk, so what would cause bad credentials not to be logged anymore (verbose=3) |
| 17:36.57 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377528.home.otenet.gr) |
| 17:37.06 | p3nguin | I've never before seen a debug that can be turned on but not turned off. |
| 17:37.26 | cerienjean | well - it can not be turned on actually... |
| 17:37.57 | p3nguin | Hmm? |
| 17:38.19 | saxa | p3nguin: you know what ? Today it works again :) |
| 17:38.20 | leifmadsen | p3nguin: should work and if not, could just be a bug |
| 17:38.29 | leifmadsen | the intention obviously not being to not be able to turn it off |
| 17:38.49 | p3nguin | So I should file this as an issue? |
| 17:38.57 | dijib | p3nguin, whats causing the callee to hear a beep on outbound calls? something in that voipms-outbound context |
| 17:39.06 | cerienjean | when a users tries to log in with wrong credential, i no longer get Registration from 'yyyy failed for xxxx - No matching peer found |
| 17:39.06 | WIMPy | tab behaves strangely there as well. |
| 17:39.07 | dijib | im thinking its mixmonitor. |
| 17:39.18 | p3nguin | I'd still like to know how to reopen an issue on jira that was closed erroneously. |
| 17:39.27 | saxa | p3nguin: for some reason i messed something with my adsl modem and it seems that this solved it. |
| 17:39.30 | WIMPy | And setting it to a specific channel doesn't seem to do anything, either. |
| 17:39.39 | p3nguin | dijib: I wouldn't know. MixMonitor() does not beep. |
| 17:39.51 | *** join/#asterisk celord (~celord@201.191.135.209) |
| 17:39.56 | dijib | then what could it be |
| 17:40.21 | p3nguin | wimpy: None of the stuff that it says it can debug actually work. Everything I tried said no such channel. The only thing that did work was "all" which is now on forever. |
| 17:40.24 | dijib | i think the mosad is manipulating my yum package, so its bridges them in |
| 17:40.50 | p3nguin | I have no idea what you just tried to say. |
| 17:41.28 | WIMPy | it does accept channel names for me, but it doesn't make a difference. It still debugs all. |
| 17:41.48 | [TK]D-Fender | dijib, "Mossad". You seem to have a real problem remembering your intelligence organizations... |
| 17:41.48 | p3nguin | wimpy: What channel name did it accept for you? |
| 17:42.05 | WIMPy | I used an LCR channel. |
| 17:42.14 | p3nguin | I tried sip. No such channel chan_sip.so |
| 17:42.42 | *** part/#asterisk analogkid (~analogkid@ip-178-202-132-139.unitymediagroup.de) |
| 17:42.42 | WIMPy | No, the name of an actually open channel. |
| 17:42.52 | bis0n | nedd to go thx u all |
| 17:43.28 | WIMPy | Using tab does no good there. |
| 17:44.32 | dijib | [TK]D-Fender, im a phonetical thinker |
| 17:45.16 | [TK]D-Fender | dijib, "I'm" ;) |
| 17:45.36 | [TK]D-Fender | p3nguin, You're passing up a gold-mine here! |
| 17:46.51 | dijib | dude i just figure its a bit of a pain in the rear to punctuate and act like you have a boner put somewhere, not implying you have a boner anywhere. as you seem to be a cool guy |
| 17:47.24 | dijib | , & . & ; is enough |
| 17:48.58 | [TK]D-Fender | dijib, Actually that last one was more up p3nguin's ally.. but he's not biting today. I was just amused about the intelligence-agency specific side :) |
| 17:50.51 | dijib | :D happy, things need to be known = actual intelligence |
| 17:51.02 | dijib | mossad and the idf have phone tech on lock down |
| 17:51.51 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-qkyleksiqluvynfu) |
| 17:51.56 | dijib | ive heard a lot of the r&d of the bigger platforms, ie alcatel. , , |
| 17:52.12 | dijib | oh is done by them |
| 17:58.29 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 18:04.44 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 18:05.15 | nny | I have a request to integrate a phone system with a TAPI compliant application. The wiki has some info, anyone care to add their .02? |
| 18:06.45 | dijib | is /usr/bin/text2wave -F 8000 -otype ulaw - /tmp/forecast.txt /var/lib/asterisk/sounds/forecast.ulaw, that -> SYNOPSIS |
| 18:06.45 | dijib | <PROTECTED> |
| 18:07.41 | Gugge | echo "text" |text2wave -someoption > file.wav |
| 18:07.49 | nny | i see some TAPI driver options, seems straightforward, nm |
| 18:24.26 | Katty | i'll tapi your driver in a minute. |
| 18:24.37 | Katty | DOUBLETAPi. |
| 18:24.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:32.38 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 18:36.56 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:42.31 | nny | :) |
| 18:54.41 | mocker | Does anyone know of a resource to see what the changes are between different revisions of Digium cards? |
| 18:56.10 | mocker | i.e. how different is a TE410P Rev. C compared to a 5th gen card? |
| 18:56.44 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v002-063.mobile.uci.edu) |
| 18:57.06 | puzzled | mocker: cheaper parts :) |
| 18:57.23 | zamba | anyone used jabra pro 9400 with asterisk? |
| 18:57.39 | mocker | puzzled: Yeah, but I think it's more than that. |
| 18:57.40 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 18:57.55 | mocker | I know at one point they changed the architecture, fixed IRQ issues, etc.. |
| 18:58.05 | mocker | Need card revision changelog! |
| 18:58.15 | puzzled | mocker: makes sense. that was in serious need of fixing |
| 18:59.54 | VoipForces | Question on Pause Queue member… |
| 19:00.00 | VoipForces | If my queue shows members like: |
| 19:00.01 | VoipForces | Callers |
| 19:00.16 | VoipForces | Local/7003@agent_call with penalty 3 (dynamic) (Not in use) has taken no calls yet |
| 19:00.55 | VoipForces | should I do PauseQueueMember(,Local/${agent}@agent_call) ? |
| 19:01.39 | [TK]D-Fender | VoipForces, as it appears |
| 19:02.14 | VoipForces | [TK]D-Fender: ok, lets try that |
| 19:03.20 | VoipForces | Yup works thanks. |
| 19:05.02 | mocker | asks the list. |
| 19:09.56 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 19:15.59 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 19:16.36 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053216.ics.uci.edu) |
| 19:27.42 | *** join/#asterisk jimbo_uk (~IceChat77@84.12.253.146) |
| 19:27.59 | jimbo_uk | is there an asterisk function that i can test if a queuemember is paused? |
| 19:28.26 | jimbo_uk | i need to test on an outbound call if a queue member is in a paused state and if so then unpause them |
| 19:28.27 | *** join/#asterisk oliver1 (~oliver@manz-590f301a.pool.mediaWays.net) |
| 19:29.01 | *** join/#asterisk celord (~celord@201.202.104.220) |
| 19:29.01 | billmania | jimbo_uk: Does your dialplan logic Pause() the member or is it from an "auto pause"? |
| 19:30.34 | *** part/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 19:31.59 | r0m|u | [TK]D-Fender, you avail? |
| 19:32.02 | billmania | jimbo_uk: If your requirement is simply to ensure that the member isn't paused before making the outbound call, call UnpauseQueueMember(), regardless of the paused-ness of the member. |
| 19:32.10 | [TK]D-Fender | r0m|u, Possibly |
| 19:32.46 | r0m|u | quick question I have my fax setup for incoming calls as follow exten => mynumberhere,1,Goto(fax,fax-rx,1) |
| 19:33.08 | jimbo_uk | bo i can't do that |
| 19:33.11 | r0m|u | but when the fax comes in I get Channel 'SIP/voipms-000000a6' sent into invalid extension 'fax' in context 'trunk-provider', but no invalid handler |
| 19:33.26 | jimbo_uk | actually what happens is when a member calls out they pause and when they hangup they unpause. |
| 19:33.45 | jimbo_uk | but the problem is that is they were alrerady paused and should remain paused then we dont weant to unpause them!! |
| 19:34.02 | r0m|u | [TK]D-Fender, what could I be doing wrong? |
| 19:34.38 | billmania | jimbo_uk: Why must the member be unpaused when calling out? Being in a paused state doesn't have any impact on outbound calls. |
| 19:35.14 | jimbo_uk | they must be paused when calling out (so they don't get new queue calls when on an outbound call) then once the call hangs up then unpaused |
| 19:35.14 | billmania | I intentionally Pause() a member everytime they make an outbound call, while a queue member, in order to prevent them being offered an inbound call while they're on an outbound call. |
| 19:35.25 | jimbo_uk | BUT if they were already paused then it's wrong to unpause them |
| 19:35.40 | jimbo_uk | otherwise we throw them back in the queue when they should be legitimately paused. |
| 19:35.53 | jimbo_uk | i need to test whether the member is paused |
| 19:35.54 | billmania | jimbo_uk: Got it. |
| 19:35.58 | jimbo_uk | :-) |
| 19:36.25 | billmania | jimbo_uk: Back to my earlier question: Did your dialplan logic Pause() the member or were they "auto paused"? |
| 19:36.34 | billmania | What caused the member to be paused in the first place? |
| 19:36.46 | jimbo_uk | they paused it using queuemetrics (so an ami call) |
| 19:36.51 | jimbo_uk | auto pause? |
| 19:37.20 | jimbo_uk | ah no, manual |
| 19:37.32 | jimbo_uk | auto pause is if we pause them if they fail to answer a call. |
| 19:37.55 | billmania | OK. When your dialplan calls PauseQueueMember(), add an entry to the asterisk database. |
| 19:38.07 | jimbo_uk | ouch |
| 19:38.16 | jimbo_uk | so there is no function that can test this? |
| 19:38.35 | billmania | I'm not aware of any function to test the paused state. |
| 19:38.46 | jimbo_uk | no me either - that's my issue lol! |
| 19:39.17 | billmania | Through AMI you can obviously parse the paused state from "queue show". |
| 19:39.33 | jimbo_uk | yes, but i'm trying to keep this pure dialplan |
| 19:39.42 | jimbo_uk | i guess it's going to have to go to an agi |
| 19:39.52 | billmania | Then I know of no better way than to add an entry to the database. |
| 19:40.06 | billmania | Adding entries to the database, testing for them and deleting them is nearly trivial. |
| 19:40.25 | jimbo_uk | sure, but the pausing is done with queuemetrics so it's not easy to modify that |
| 19:40.32 | jimbo_uk | in fact let me check |
| 19:40.33 | jimbo_uk | ... |
| 19:41.07 | billmania | jimbo_uk: Check to see if queuemetrics is putting something in the asterisk DB for its own use. |
| 19:41.16 | jimbo_uk | what about agent_status function |
| 19:41.33 | jimbo_uk | but i think this is for agent callback login |
| 19:44.16 | billmania | There is a function named "QUEUE_MEMBER_LIST()" but I don't know the details of what it returns: https://wiki.asterisk.org/wiki/display/AST/Function_QUEUE_MEMBER_LIST |
| 19:45.10 | [TK]D-Fender | <r0m|u> but when the fax comes in I get Channel 'SIP/voipms-000000a6' sent into invalid extension 'fax' in context 'trunk-provider', but no invalid handler <- a fax was detected and * tried jumping to that extension and it did not exist just as it says |
| 19:49.03 | r0m|u | [TK]D-Fender, Yea I figured it out. I had the whole damn context backwards :) |
| 19:49.06 | r0m|u | Thanks for the help! |
| 19:49.21 | jimbo_uk | hmm |
| 19:50.31 | dijib | what do you guys think of this???? |
| 19:50.32 | dijib | http://pastebin.com/Ru0Py6SL |
| 19:51.05 | dijib | how would i Set(variable=/tmp/textfile.txt) & echo $variable ???? |
| 19:51.15 | dijib | in the text2wave |
| 19:54.09 | Gugge | dijib: system("text2wave < ${filename} > something.wav"); |
| 19:54.21 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 19:54.42 | dijib | i havn't been able to get that working Gugge |
| 19:54.54 | Gugge | then watch the verbose cli output while it runs |
| 19:57.12 | [TK]D-Fender | dijib, same => n,System("echo "${variable}") |
| 19:57.27 | dijib | need to quote it? |
| 19:57.37 | [TK]D-Fender | no, I just grabbed that from your sample |
| 19:57.56 | dijib | same => n,System("echo "${variable}"") <-double quote? |
| 19:58.00 | Gugge | wouldnt it fuck up with all those "'s ? |
| 19:58.04 | [TK]D-Fender | Quote = unimportant |
| 19:58.09 | dijib | ok |
| 19:58.19 | [TK]D-Fender | ${} <- variable reference. shove it where it deserves |
| 19:58.32 | p3nguin | I'd think quoting what System() is supposed to run would be bad. |
| 19:58.51 | Qwell | > bash: unexpected EOF while looking for matching `"' |
| 19:59.54 | p3nguin | Quote what you want System() to echo, but don't quote the echo itself. |
| 20:00.35 | dijib | like this ? |
| 20:00.36 | dijib | same => n,Set(forecast=$/tmp/forecast.txt)); |
| 20:00.36 | dijib | same => n,System("echo ${forecast} | text2wave -F 8000 -o /var/lib/asterisk/sounds/en/forecast.ulaw -otype ulaw"); |
| 20:01.17 | Gugge | remove $ from the Set() |
| 20:01.32 | Gugge | and do you realy want it to echo the filename, and not the content from the file? |
| 20:01.46 | dijib | centents |
| 20:01.51 | dijib | contents |
| 20:02.01 | Gugge | then you are doing it wrong :) |
| 20:02.14 | dijib | ive noticed as its not working |
| 20:02.48 | Gugge | that would be because you are doing it wrong :) |
| 20:02.57 | [TK]D-Fender | <PROTECTED> |
| 20:02.59 | Tim_Toady | u can use somethign like ReadFile(MYTEXT=/path/file) and then try echo ${MYFILE} |
| 20:03.17 | dijib | yeah i cleaned those up after i sent that |
| 20:03.32 | dijib | thanks Tim_Toady ill try that |
| 20:03.40 | Tim_Toady | but it might be better to use an agi script instead of calling system() |
| 20:03.41 | Gugge | why do you want the content in a variable? |
| 20:03.53 | Gugge | just do as i wrote |
| 20:04.00 | Tim_Toady | or an existing text2speek app that can work within the dialplan |
| 20:04.01 | Gugge | text2wave < ${filename} |
| 20:04.08 | Tim_Toady | text2speech* |
| 20:04.10 | dijib | the text from /forecast.txt |
| 20:04.30 | *** join/#asterisk Russ (~russ@206.29.182.161) |
| 20:05.12 | dijib | readfile doesnt seem to work. |
| 20:07.30 | [TK]D-Fender | dijib, this isn't bash.... |
| 20:07.40 | [TK]D-Fender | * vars are char limited in a nasty way |
| 20:07.52 | [TK]D-Fender | dijib, You are using the wrong tool for the job |
| 20:08.25 | dijib | like whats my limit? ive set 65535 in the readfile options |
| 20:08.54 | dijib | [TK]D-Fender, your thinking agi script aswell? |
| 20:09.30 | Gugge | or just run text2wave < ${filename} :P |
| 20:09.35 | [TK]D-Fender | dijib, or just a system call. |
| 20:11.23 | dijib | so this? same => n,System("text2wave -F 8000 -o /var/lib/asterisk/sounds/en/forecast.ulaw -otype ulaw > /tmp/forecast.txt"); |
| 20:11.37 | Gugge | i give up :) |
| 20:11.38 | dijib | nope |
| 20:11.46 | dijib | oi < |
| 20:12.29 | *** join/#asterisk nix8n82 (~nate@24.143.28.16) |
| 20:12.35 | *** join/#asterisk wonderworld (~ww@port-92-201-254-65.dynamic.qsc.de) |
| 20:12.38 | dijib | Gugge, how did i deviate? |
| 20:13.40 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:13.40 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:14.32 | Qwell | remove the quotes.. |
| 20:15.06 | Madkiss | dammit. |
| 20:15.19 | Madkiss | i would love to find out where these damm choppysoun-issues are coming fromg |
| 20:15.57 | Madkiss | apparently, straight after initiating a call, sound is fine |
| 20:16.00 | Gugge | bad network, bad virtualizations system (if its a vm), bad phones |
| 20:16.06 | Madkiss | and then, after some seconds, sound starts to be choppy |
| 20:16.09 | Madkiss | Gugge: KVM |
| 20:16.16 | Gugge | try on real iron :) |
| 20:16.21 | Gugge | just to be sure |
| 20:16.40 | Madkiss | well. I would rather like to know what's the reason for this. vere scire est per causas scire. |
| 20:16.53 | Gugge | eliminate one think at a time |
| 20:17.01 | Gugge | thing |
| 20:17.23 | Freeaqingme | ~thebook |
| 20:17.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 20:17.41 | Freeaqingme | thank you leifmadsen & Rufus ! |
| 20:17.50 | Freeaqingme | * russellb |
| 20:17.51 | Madkiss | the problem seems to be much more apparent on MeetMe conferences |
| 20:18.24 | navaismo | <PROTECTED> |
| 20:18.32 | Gugge | Madkiss: what dahdi hardware do you use? |
| 20:18.42 | Madkiss | inside the KVM? |
| 20:18.47 | Gugge | yes |
| 20:18.56 | Madkiss | dahdi as such, providing dahdi_dummy these days |
| 20:19.07 | Gugge | dahdi_dummy uses the system timer |
| 20:19.13 | Gugge | which often suck in a vm |
| 20:19.16 | Gugge | use real iron :) |
| 20:19.29 | navaismo | madkiss your kvm support usb control? |
| 20:19.51 | Madkiss | navaismo: what exactly do you mean? whether i have passed through a USB device to theVM? |
| 20:20.07 | navaismo | yep like virtualbox |
| 20:20.10 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-73-BusName-Richmond.hfc.comcastbusiness.net) |
| 20:20.14 | Madkiss | I don't think so |
| 20:20.25 | Madkiss | but that can change |
| 20:20.42 | Qwell | What does USB have to do with anything? |
| 20:20.53 | navaismo | i ask because sangoma sell usb keys for provide a timing source |
| 20:21.09 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:21.18 | Qwell | USB is less accurate than what dahdi uses. |
| 20:21.18 | Madkiss | I have read that dahdi uses USB as clock source, too |
| 20:21.52 | navaismo | http://wiki.sangoma.com/sangoma-wanpipe-voicetime |
| 20:22.27 | [TK]D-Fender | Madkiss, in ancient times. |
| 20:22.31 | Madkiss | ah, I see |
| 20:22.35 | Madkiss | so that's a dead end, too |
| 20:23.05 | [TK]D-Fender | checkout time, BBL |
| 20:37.25 | *** part/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 20:43.43 | LittleFool | Hello, i always see incoming calls through my first Sip Trunk but its impossible that they come throuw this trunk. Is this a bug in asterisk or in freepbx? |
| 20:47.30 | wdoekes2 | if you're using freepbx, you should be in #freepbx |
| 20:48.05 | LittleFool | but its not a bug in asterisk? |
| 20:48.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:48.43 | wdoekes2 | what you said doesn't make any sense to me, so I'm assuming user error |
| 20:49.22 | LittleFool | well it doesnt make any to me too. but i cant write my own dialplans and stuff thats the main reason why i use freepbx |
| 20:50.09 | wdoekes2 | please make sense before you write anything then. I don't know what your "first sip trunk" is or why calls should or should not go throuw [sic] it |
| 20:51.22 | LittleFool | ok sorry that i asked man :/ |
| 20:51.46 | *** join/#asterisk caveat- (~false@gateway/shell/bshellz.net/x-fjxuvsakuoivjtct) |
| 20:52.02 | wdoekes2 | please heed my advice and look in #freepbx. they will probably point you to some configuration issue |
| 20:52.22 | LittleFool | i asked there |
| 20:53.21 | wdoekes2 | perhaps you need to rephrase the question. maybe they don't know what your saying either? (kind advice, no insult) |
| 20:53.29 | wdoekes2 | *you're |
| 20:58.22 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 21:12.29 | *** join/#asterisk vbman2 (~WildSide@207.251.82.226) |
| 21:13.22 | vbman2 | how can i setup multiple companies in asterisk |
| 21:13.30 | vbman2 | such as company A has extention 200 |
| 21:13.30 | Katty | ohaii |
| 21:13.35 | vbman2 | company b has exten 200 |
| 21:13.53 | vbman2 | how can i keep the calls between them seperate |
| 21:14.03 | vbman2 | so if company a calls 201 they wont get company b 201 |
| 21:14.10 | wdoekes2 | vbman2: (a) have them land in different [context]s, (b) use other means like db queries to distinguish who is who |
| 21:14.30 | wdoekes2 | option (a) is simplest, option (b) scales the best |
| 21:14.58 | vbman2 | so if user company b extention 201 calls 200 it will ring only company b extn 200 right? |
| 21:15.15 | wdoekes2 | you already said that |
| 21:15.28 | vbman2 | will asterisk let me make multiple instances of same exten #? |
| 21:15.38 | vbman2 | never done asterisk this way before |
| 21:16.02 | wdoekes2 | if you have [companyA] context in your extensions.conf, and all sip accounts belonging to that company have context=companyA |
| 21:16.17 | vbman2 | ok cool |
| 21:16.27 | vbman2 | makes sense so for each company on the pbx |
| 21:16.30 | vbman2 | setup a context right? |
| 21:16.35 | *** join/#asterisk jerware (~jerryg@c-71-58-179-44.hsd1.pa.comcast.net) |
| 21:16.37 | wdoekes2 | then they all start out in that context.. so you can have exten => 200 there do something other than in [companyB] |
| 21:16.41 | [TK]D-Fender | vbman2: 1st step : understand that a SIP device name and a dialed extension # have no inherent relationship to one another |
| 21:17.11 | vbman2 | whats more user friendly workign with asterisk directly |
| 21:17.16 | vbman2 | or using trixbox or elastix? |
| 21:17.16 | [TK]D-Fender | vbman2: Don't just use numbers for device names |
| 21:17.30 | [TK]D-Fender | vbman2: NEITHER of those are multi-tennent <- |
| 21:17.45 | [TK]D-Fender | vbman2: Just saving you a lot ogrief putting it out there.. |
| 21:17.54 | vbman2 | so just do a fresh standalone plain asterisk setup right? |
| 21:18.09 | [TK]D-Fender | vbman2: for a little extra background, both use FreePBX (trixbox is using a forked version that is older) |
| 21:18.13 | wdoekes2 | for fresh standalone plain vanilla asterisk, you're in the right place |
| 21:18.32 | [TK]D-Fender | vbman2: I'm just saying that those 2 won't do it |
| 21:18.42 | vbman2 | i've used elastix before |
| 21:18.46 | vbman2 | for a single company pbx |
| 21:18.58 | wdoekes2 | ~book |
| 21:18.58 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 21:19.13 | vbman2 | my objective is i want to run a single asterisk pbx as a hosted platform for multiple companies |
| 21:22.25 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-17-5.w86-196.abo.wanadoo.fr) |
| 21:26.37 | wonderworld | is using asterisk 10 in a production environment suicide or should i expect minor problems only? |
| 21:27.22 | wdoekes2 | http://images.memegenerator.net/instances/400x/9689481.jpg |
| 21:27.24 | [TK]D-Fender | wonderworld: "beta". Best blocked by a few feet of water... |
| 21:28.56 | vbman2 | so in short setup contexts for each company right? |
| 21:29.52 | _Corey_ | wdoekes2: Nice |
| 21:30.01 | [TK]D-Fender | vbman2: that is the dialplan part just for starting |
| 21:30.03 | wonderworld | [TK]D-Fender: sry, my english isn't good enough to understand the comment. what would "Best blocked by a few feet of water" mean? |
| 21:30.47 | [TK]D-Fender | wonderworld: Beta.. like the radiation. Make good use of that 3rd flipper you're trying to grow... |
| 21:31.02 | vbman2 | just for starting? |
| 21:31.20 | vbman2 | are they anything like elastix already setup for multi-tenant? |
| 21:31.58 | [TK]D-Fender | vbman2: Nothing free. |
| 21:32.37 | vbman2 | ok |
| 21:32.53 | vbman2 | so what all do i have to do for multi-tenant enviorment |
| 21:33.30 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 21:34.00 | [TK]D-Fender | vbman2: "nothng based on FreePBX" |
| 21:35.19 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 21:35.25 | vbman2 | huh |
| 21:36.01 | [TK]D-Fender | vbman2: Roll your own, or find some other solution that accomodates it. Nothing based on FreePBX does |
| 21:36.03 | _Corey_ | vbman2: You could look into the 2600hz project, that might be what you want... |
| 21:36.07 | jwiggins | Is the "pri" command still part of Dahdi in 2.5 and Asterisk 1.8? |
| 21:37.14 | Madkiss | abstract_jb.c:429 jb_get_and_deliver: JB_IMPL_NOFRAME is returned from the adaptive jb when now=121460 >= next=121452, jbnext=121452! |
| 21:37.17 | Madkiss | ahum |
| 21:37.29 | [TK]D-Fender | jwiggins: Yes |
| 21:38.07 | wonderworld | vbman2: i'd think about scalability first. how do you want to trunk? |
| 21:38.31 | jwiggins | I have "dahdi show status" working and it is showing my Xorcom XPD device and the "Dynamic 'ethmf' span" (fonebridge2) but the "pri" command says it does not exist |
| 21:38.38 | luckman212 | anyone know if it is possible to issue a Dial() command *WITHOUT* having Asterisk update the CONNECTEDLINE() info? I have an Asterisk 1.8.7 setup and I am setting CallerID info from a database -- this works great and the name appears on the phones display but as soon as the dialplan hits the Dial() command it gets overwritten |
| 21:38.46 | jwiggins | any assistance as to what may cause this? "dahdi_tool" shows both as "OK" |
| 21:39.16 | _Corey_ | jwiggins: In my experience it happens when something is wrong with your dahdi-related conf files |
| 21:39.44 | [TK]D-Fender | jwiggins: Nowhere do I see you confirming having installed libpri |
| 21:40.24 | jwiggins | [TK]D-Fender, wow... is that it... /smack |
| 21:41.15 | jwiggins | luckman212, are you seeing it actually being overwritten in the logs, or you just see the CID display differently? |
| 21:43.19 | mocker | vbman2: Sounds like you should avoid the dialplan and setup multiple servers for each company. |
| 21:43.28 | mocker | er, one server per company. |
| 21:44.38 | wonderworld | i couldn't sleep very well with one server handling all my customers |
| 21:44.41 | luckman212 | jwiggins: I put a Wait() command in my dialplan |
| 21:45.01 | luckman212 | jwiggins: everything up to the Wait() is perfect, the display shows the correct CID name |
| 21:45.55 | luckman212 | jwiggins: the next step in the dialplan is the Dial() command, and at that moment the display changes back to e.g. "device <700>" which is the callerid set in that particular SIP PEER |
| 21:47.31 | jwiggins | so your Noop right before the Dial command states the proper CID? |
| 21:49.32 | luckman212 | yes. but It seems that the Dial() command does a new lookup of the "callerid=" value from the SIP peer (these are local extension->local extension calls) and overwrites whatever was set menually via CONNECTEDLINE |
| 21:51.16 | luckman212 | ah, I think I just figured it out |
| 21:53.41 | jwiggins | what was it? |
| 21:54.11 | luckman212 | the "I" option needs to be set for the Dial() |
| 21:54.18 | luckman212 | e.g. Dial(SIP/123,20,I) |
| 21:54.29 | luckman212 | without that, the CONNECTEDLINE info gets clobbered |
| 21:59.29 | navaismo | Hey rdegges your chanspy issue gone after downgrade asterisk? |
| 22:01.54 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 22:08.31 | *** join/#asterisk f2knight (~ben@c-76-115-43-21.hsd1.or.comcast.net) |
| 22:09.28 | Freeaqingme | It's been a while that I last did something with asterisk, but I'm missing here something. I have a client that is able to connect, but whenever it tries to make a call, the request times out (408). Nothing about the call is shown with any kind of debugging on in asterisk. What is a likely problem? |
| 22:10.18 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 22:12.00 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.247.170.dsl.dyn.forthnet.gr) |
| 22:15.46 | [TK]D-Fender | Freeaqingme: If you've enabled SIP debug and still see nothing then packets aren't even making it to your box. |
| 22:16.09 | [TK]D-Fender | Freeaqingme: Firewalls, routing, client config. One or multiple. |
| 22:16.38 | Freeaqingme | good point |
| 22:17.15 | Freeaqingme | It's on a vm, asterisk is listening on the right port. iptables accepts all. wireshark shows packets are sent, but none returned (although registration works) |
| 22:20.15 | [TK]D-Fender | Freeaqingme: Wireshark on the client side? |
| 22:20.30 | Freeaqingme | yes |
| 22:21.51 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 22:23.57 | Freeaqingme | [TK]D-Fender, tnx for the attention. past midnight here; will give it a try tomorrow |
| 22:29.10 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053216.ics.uci.edu) |
| 22:46.22 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 22:46.22 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 22:54.05 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:57.34 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:58.34 | *** part/#asterisk fireman_biff (~biff@65.48.133.103) |
| 23:05.21 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 23:07.43 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 23:11.28 | *** join/#asterisk fireman_biff (~biff@65.48.133.103) |
| 23:16.12 | fireman_biff | I am setting up dundi between two PBXs, one running asterisk 1.4 and the other 1.6. With identical setups I can call from the 1.6 to the 1.4 fine, but when I try to call from the 1.4 to the 1.6 the call fails because it is "circuit busy" (Everyone is busy/congested at this time) although dundi finds the extension with no problem. Any idea where I should look? Could it be a syntax difference between 1.4 and 1.6 causing this? |
| 23:17.34 | WIMPy | dundi doesn't transport calls it only lets you discover a route to the destination. |
| 23:17.57 | WIMPy | So you should look at whatever transports the calls. |
| 23:21.25 | fireman_biff | WIMPy: alright, let me turn on IAX debugging and see what happens |
| 23:25.50 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 23:31.18 | fireman_biff | with IAX debugging I'm seeing an Rx-Frame with subclass REJECT on the 1.4 side, but nothing on that stands out on the 1.6 side and nothing that actually explains whats wrong |
| 23:32.07 | navaismo | what its the advantage using dundi of iax2/sip direct connection? im never use dundi only iax2 because i dont understand dundi. |
| 23:32.11 | SeRi | p3nguin, you avail? |
| 23:32.42 | navaismo | fireman_biff maybe the calltoken in the 1.6 its the problem. do you disabled it? |
| 23:33.04 | fireman_biff | navaismo: i'm only now starting to check it out, but it seems very useful for routing calls between offices when you have many offices |
| 23:33.13 | fireman_biff | hmm... let me check that out... |
| 23:35.21 | WIMPy | Try the normal verbose/debug instead of iax debug. |
| 23:35.47 | SeRi | guys I am getting handle_response_invite: Received response: "Forbidden" |
| 23:35.48 | fireman_biff | navaismo: that was it, thanks a lot |
| 23:36.00 | SeRi | when I am calling out |
| 23:36.07 | fireman_biff | are there any security implications of using requirecalltoken=no ? |
| 23:36.53 | WIMPy | There is a readme for exactely those questions. |
| 23:37.48 | navaismo | https://wiki.asterisk.org/wiki/display/AST/IAX2+Security |
| 23:38.23 | fireman_biff | thanks again |
| 23:38.28 | navaismo | np |
| 23:39.13 | SeRi | navaismo, you think you can help me out with an error I am getting? |
| 23:39.35 | SeRi | when I amke outbound calls I get handle_response_invite: Received response: "Forbidden" |
| 23:40.01 | navaismo | using a ITSP ("sip trunk") |
| 23:40.49 | SeRi | yes sr |
| 23:41.13 | navaismo | SeRi when i see the forbidden string its cause wrong password fromdoain its wrong or wrong user |
| 23:41.34 | SeRi | I can receive calls just fine thoug :? I am also registerd |
| 23:42.15 | navaismo | can you pb the complete line of the forbidden message |
| 23:42.19 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 23:42.21 | SeRi | sure one sec |
| 23:42.23 | p3nguin | seri: yep |
| 23:43.04 | *** part/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 23:43.14 | p3nguin | If you want to know if you are registered, sip show registry is the command for that. |
| 23:44.04 | SeRi | http://pastebin.com/CjXphPGR |
| 23:44.45 | SeRi | p3nguin, Yes its showing me registered |
| 23:45.33 | SeRi | I am puzzled... :/ |
| 23:46.13 | p3nguin | So far I haven't seen anything useful. |
| 23:47.11 | navaismo | I guess you are sending the extension cidnum instead the peer, ill suggest check the fromuser and fromdomain and if apply the sendrpid |
| 23:47.18 | navaismo | options |
| 23:49.49 | SeRi | http://pastebin.com/StYzngVm |
| 23:49.56 | SeRi | thats my sip.conf |
| 23:51.38 | navaismo | add to callcentric peer the trystrpid and sendrpid, then sip relod in the cli and try again |
| 23:52.42 | SeRi | ok |
| 23:53.22 | DrDigital | Steve jobs died http://www.apple.com/stevejobs/ |
| 23:53.55 | p3nguin | CallCentric does not say they need trustrpid/sendrpid. |
| 23:53.59 | SeRi | navaismo, do I set them to yes? |
| 23:54.51 | SeRi | p3nguin, thats why i dont have them define... the funny part is that it use to work and I cancel the account and had to repopen it because of family :/ |
| 23:54.57 | SeRi | I am willing to give it a try |
| 23:55.19 | p3nguin | For what reason does "family" make you reopen the account? |
| 23:56.00 | navaismo | DrDigital seriously?? |
| 23:56.06 | p3nguin | Seriously. |
| 23:56.10 | SeRi | IP calling for free from Puerto Rico... ugh... as much as I hate CC I have to :( |
| 23:56.10 | DrDigital | www.apple.com |
| 23:56.19 | DrDigital | its on the bottom of like every channel of my tv |
| 23:56.26 | navaismo | o_O whaat he was fine right |
| 23:56.55 | DrDigital | he just stepped down not to long ago as CEO |
| 23:56.58 | DrDigital | it was coming |
| 23:57.03 | dym | navaismo: no - he seemed ill for some time |
| 23:57.04 | *** join/#asterisk nighty- (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 23:57.13 | navaismo | llevame a miiii |
| 23:57.19 | navaismo | sorry wrong window |
| 23:57.21 | dym | you could tell from WWDC to WWDC |
| 23:57.27 | SeRi | navaismo, lol |
| 23:57.28 | dym | getting more slim etc |
| 23:58.01 | p3nguin | He's had some health issues over the past several years. |
| 23:58.13 | p3nguin | I think that was part of what made him step down from CEO. |
| 23:58.31 | dym | yupp |
| 23:58.32 | dym | agreed |
| 23:58.32 | navaismo | RIP |
| 23:58.47 | SeRi | navaismo, still the same error |
| 23:59.17 | WIMPy | Will the iHype survive the iDeath? |
| 23:59.28 | dym | one would wonder |
| 23:59.38 | dym | the shared went down after the iphone presentation already |
| 23:59.41 | dym | shares* |
