| 00:01.50 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 00:25.08 | *** join/#asterisk slim_ (~slim_@41.239.33.60) |
| 00:27.39 | slim_ | hello all |
| 00:39.48 | *** join/#asterisk coppice (~chatzilla@m121-203-228-198.smartone-vodafone.com) |
| 00:42.53 | *** part/#asterisk VoIPBroker (~VoIPBroke@96.176.19.11) |
| 00:48.39 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 00:52.08 | *** part/#asterisk slim_ (~slim_@41.239.33.60) |
| 00:52.37 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 01:00.10 | tzanger | voip.ms++ - had the DID up in under 5 min, and that includes new user registration |
| 01:03.06 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 01:04.24 | QbY | that's cool. but US48 at 0.0105 that's a bit pricey. |
| 01:04.53 | QbY | i'd much rather go through the hassle of e-mail, than pay twice what it costs for non-premium service.. |
| 01:13.35 | *** join/#asterisk radic (~radic@dslb-178-007-134-255.pools.arcor-ip.net) |
| 01:28.26 | *** join/#asterisk weinerk (~irc@unaffiliated/weinerk) |
| 01:28.38 | weinerk | Hi. I have a question about freepbx scenario - |
| 01:28.38 | weinerk | I am trying to do the use a ringgroup from AGI: |
| 01:28.38 | weinerk | 1) agi script accepts call |
| 01:28.38 | weinerk | 2) $AGI->exec("Dial","Local/600@ext-group|55"); -to ring a group |
| 01:28.38 | weinerk | 3) $AGI->get_variable('DIALSTATUS'); |
| 01:28.38 | weinerk | 4) if answered do XXX else do YYY |
| 01:28.39 | weinerk | Problem is that ringgroup does not really return processing back to agi. |
| 01:28.39 | weinerk | How can I get the control back to AGI? |
| 01:30.19 | *** join/#asterisk allan8904 (~allan@unaffiliated/allan8904) |
| 01:35.04 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 01:54.40 | *** join/#asterisk [netman] (~netman@20.Red-80-39-52.staticIP.rima-tde.net) |
| 01:56.16 | godofmonkeys | p3nguin, it's both actually, but i'm only worried about incoming |
| 02:01.27 | *** join/#asterisk Russ (~russ@206.29.188.233) |
| 02:02.06 | *** join/#asterisk tonsofpcs (~tonsofpcs@april/fools/thesixthest/tonsofpcs) |
| 02:02.40 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 02:03.56 | pabelanger | weinerk: you may have better luck in #freepbx |
| 02:05.13 | *** join/#asterisk eugeneoden (~goden@99-62-173-93.lightspeed.austtx.sbcglobal.net) |
| 02:11.41 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 02:25.21 | p3nguin | godofmonkeys: Tell me again what the exact problem is. |
| 02:25.37 | godofmonkeys | my callerid is hosed; shows the proper name but the number is the username of my account with my provider. somebody point me to the proper place? |
| 02:26.09 | godofmonkeys | i can see the right number in the cli |
| 02:46.48 | *** join/#asterisk drunkmaster (~lucifer@armageddon.perm.ru) |
| 02:46.56 | drunkmaster | hi |
| 02:47.57 | drunkmaster | Am I right that code in chan_sip.c (1.6.2 svn) http://pastebin.com/nQ3G5fA6 contains error and must be http://pastebin.com/sH0W9FfM ? |
| 02:50.00 | drunkmaster | And I added to handle_response_invite function 181 response threatment in the same way as 180. Is it right? |
| 02:50.58 | drunkmaster | I've done it because of Broadworks early media signalisation (req invite, resp 100, 181[sdp], 180, 180...) |
| 02:55.25 | p3nguin | godofmonkeys: Can you show me the cli output where it shows the correct data? If you can show me the entire call, that would be ideal. |
| 02:59.41 | godofmonkeys | 1 sec |
| 03:00.03 | drunkmaster | Sorry, first question has been removed. ^+) |
| 03:02.19 | godofmonkeys | p3nguin, http://pastebin.com/86xUpN0X |
| 03:02.41 | godofmonkeys | warnings are fine, those group extensions are not connected right now |
| 03:02.56 | Greek-Boy | what ever happened to [tk]defender ? |
| 03:03.46 | godofmonkeys | on my client i see did@provider |
| 03:04.12 | godofmonkeys | but with the correct name |
| 03:05.56 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 03:06.45 | kaldemar | godofmonkeys: caller id doesn't seem to be hosed in any way, based on what you show. |
| 03:07.01 | godofmonkeys | it's not passing the right number to my extension |
| 03:07.31 | kaldemar | to your device. |
| 03:07.40 | godofmonkeys | err, yes |
| 03:09.17 | kaldemar | extension in asterisk speak is a part of your dialplan. anyway, what is the client supposed to show you? |
| 03:10.00 | godofmonkeys | the phone number of the person calling, not my account with my provider |
| 03:10.55 | godofmonkeys | if i miss a call, i want to be able to call them back, not myself |
| 03:11.53 | kaldemar | enable sip debug and see what gets sent to the client. |
| 03:13.22 | *** join/#asterisk neurosys (~neurosys@c-71-196-30-85.hsd1.fl.comcast.net) |
| 03:15.08 | godofmonkeys | http://pastebin.com/dyQFHRab |
| 03:16.21 | godofmonkeys | well... hold on a sec... |
| 03:17.05 | *** join/#asterisk Russ (~russ@206.29.188.183) |
| 03:18.37 | godofmonkeys | i fixed it, is there a way to strip off the @blah part? shows on my ..device as rightnumber@asteriskprivateip |
| 03:20.54 | godofmonkeys | perhaps fromdomain= |
| 03:20.58 | godofmonkeys | then nothing |
| 03:21.04 | godofmonkeys | in my sip.conf per extension |
| 03:23.34 | *** join/#asterisk TheRealJeanLuc (~jcrawford@dhcp-west-78.resnet.nmt.edu) |
| 03:24.02 | TheRealJeanLuc | Hey, I'm using Asterisk-GUI and it's listing status "120" for a trunk I'm trying to set up, and it doesn't seem to be connected. Any idea what that means? |
| 03:24.22 | *** join/#asterisk Maxus2 (~Maxus@59.191.225.49) |
| 03:24.25 | godofmonkeys | never trust a gui? |
| 03:24.42 | kaldemar | godofmonkeys: see x-lite settings first, making a domain empty is quite ugly. |
| 03:25.22 | TheRealJeanLuc | haha. probably true. I'm brand new to asterisk and I thought using a gui might be easier, but after dealing with FreePBX and now trying Asterisk-gui I'm wondering if I ought to give up on them. |
| 03:25.50 | godofmonkeys | i had to; there were options i had to have that werent displayed for my provider |
| 03:25.55 | kaldemar | #asterisk-gui |
| 03:26.14 | Maxus2 | Hi People, if im pass calls between asterisk boxes using IAX, is it possible to pass varibles along too? |
| 03:26.32 | TheRealJeanLuc | ah. thanks, kaldemar. |
| 03:26.35 | Maxus2 | seems as if the iax varibles are empty |
| 03:26.43 | kaldemar | Maxus2: yes. with function IAXVAR. |
| 03:26.45 | Maxus2 | ont he ecieving side |
| 03:26.56 | Maxus2 | Thanks kaldemar! |
| 03:27.06 | Ean | Does it matter where the DID server hosted vs PBX server distance? |
| 03:27.07 | Maxus2 | dont know how i missed that. |
| 03:28.14 | Maxus2 | what about if im also using SIP? |
| 03:28.48 | Maxus2 | do i use the set command for sipa nd the iaxvar for iax? |
| 03:30.08 | p3nguin | godofmonkeys: I have a feeling the problem is within your sip peer entry for your phone. Pastebin that. |
| 03:30.12 | kaldemar | sip has nothing to do with this. and you acnt set any variables with set. |
| 03:30.31 | Maxus2 | so if i have a box peered using sip i can pass varibles to it? |
| 03:30.39 | kaldemar | IAXVAR is a function, not an application like Set. |
| 03:31.25 | kaldemar | Maxus2: not like with IAX. you can use X-headers in SIP. |
| 03:32.20 | godofmonkeys | p3nguin, http://pastebin.com/W2p1dmbq |
| 03:32.41 | godofmonkeys | i have fixed the number, you were corect, in sip.conf for the device |
| 03:32.50 | Maxus2 | cool, thanks kaldemar, so i take it there is no generic way to pass varibles to both? |
| 03:33.10 | Maxus2 | as we have some clients peering boxes using iax and some using sip. |
| 03:33.16 | kaldemar | Maxus2: no, not a common way. |
| 03:33.30 | Maxus2 | ouch, im really not winning. |
| 03:35.08 | Maxus2 | trying to find a way to see if a set of records in the CDR all belong together as one call, just cant find anyway to tie them together |
| 03:35.36 | Maxus2 | as we have a call bounce through a couple servers all outputing to the same CDR |
| 03:36.02 | Maxus2 | but we end up with two or three records, and the unique ID is different on all of them |
| 03:38.21 | Maxus2 | are you aware of a way to tie them together? |
| 03:38.52 | p3nguin | godofmonkeys: So it's fixed, or you still have something not right? |
| 03:42.13 | kaldemar | Maxus2: pass an account code to each box and use it. |
| 03:44.28 | godofmonkeys | p3nguin, it's passing the private ip of the asterisk server, on my device i see 'RIGHT NAME - RIGHT NUMBER@ASTERISKINTERNALIP' |
| 03:44.49 | godofmonkeys | which down to it, i've got bigger fires. |
| 03:45.16 | godofmonkeys | still need to setup basic auto attendent and provision this polycom |
| 03:50.18 | godofmonkeys | speaking of, the book has no info on polycom phone provisioning |
| 03:50.26 | godofmonkeys | any good place to look? |
| 03:51.13 | *** join/#asterisk pig (~pig@bb220-255-91-16.singnet.com.sg) |
| 03:51.31 | p3nguin | godofmonkeys: Let me show you one of my sip phone entries: http://pastebin.com/knpya8uW |
| 03:52.16 | *** part/#asterisk pig (~pig@bb220-255-91-16.singnet.com.sg) |
| 03:53.07 | p3nguin | godofmonkeys: Now that I took a look at the paste you provided for a phone by the name of 100, it doesn't look bad. By not bad, I mean it should work fine. |
| 03:53.19 | godofmonkeys | hahah |
| 03:53.29 | godofmonkeys | thanks, i guess |
| 03:57.30 | godofmonkeys | damnit, i've broken something with device-device calling. 101 can call 100, 100 can |
| 03:57.35 | godofmonkeys | not call 101 |
| 03:59.03 | godofmonkeys | never mind. overwrote extensions.conf with a version before it had the dialplan for the new extensions |
| 03:59.08 | Maxus2 | Hi kaldemar, and account code we seperate individual calls |
| 03:59.22 | Maxus2 | Hi kaldemar, an account code wont seperate individual calls? |
| 04:02.02 | kaldemar | it is something that you set. it may or may not. |
| 04:03.32 | Maxus2 | yeah we need to use that to identify who all the calls belong to as a whole so i cant change it :( |
| 04:04.36 | Maxus2 | maybe i could tack somthing onto the end on a per call basis and strip it out i guess |
| 04:22.07 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:33.40 | *** join/#asterisk kannan (kann@123.238.238.126) |
| 04:36.24 | kannan | hello all. I am trying to integrate asterisk 1.4. latest with a panasonic PBX thru E1 PRI (euroisdn) . all is working except that outgoing calls thru the panasonic land on 's' exten only and not on other dialplan priority. I have tried to make chan_dahdi setting "immediate = no" . I am not able to to get the extension wise CDR reports , which was a key reason for the integration also |
| 04:37.39 | *** join/#asterisk frek818 (~herman@2001:470:f0f9:100:a6ba:dbff:feef:34b1) |
| 04:39.41 | p3nguin | It's not unusual for all calls to end up on extension s when using analog channel technology. |
| 04:39.47 | p3nguin | It's actually normal. |
| 04:40.25 | kannan | is there any way to know which extension from panasonic is dialling out the call? |
| 04:42.43 | kannan | right nowe i have asked the calling party to prefix the 3 digit internal phone number and then the actual number to be dialled out . they do this on a ivr setup in the s exten |
| 04:42.50 | godofmonkeys | how about auto provisioning? |
| 04:42.59 | kannan | every one hates this system now , lol |
| 04:51.12 | *** join/#asterisk JoeT (Cable@cpe-173-168-180-22.tampabay.res.rr.com) |
| 04:55.06 | *** join/#asterisk choppernator (~chopp@unaffiliated/chopp) |
| 04:58.02 | JoeT | Hi folks... Got a problem that I hope someone can help with. My setup was working fine until recently, now, incoming calls via Google Voice are only answered 30-40% of the time. Outgoing calls and everything else work fine. Google didn't turn up anything on point. Help? |
| 04:59.55 | *** join/#asterisk Russ (~russ@70.99.169.146) |
| 05:03.41 | godofmonkeys | ok, probably a borderline stupid question, but if i setup users.conf and turn on the httpd blah blah blah, do i need to manually add the same entries for the devices in sip.conf? |
| 05:04.19 | godofmonkeys | clarification: to make polycoms auto provision |
| 05:09.11 | p3nguin | Don't ever use users.conf. |
| 05:09.31 | p3nguin | ~users.conf |
| 05:09.31 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 05:11.05 | Maxus2 | LOL |
| 05:11.23 | JoeT | Hi folks... Got a problem that I hope someone can help with. My setup was working fine until recently, now, incoming calls via Google Voice are only answered 30-40% of the time. Outgoing calls and everything else work fine. Google didn't turn up anything on point. Help? |
| 05:11.57 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 05:13.30 | p3nguin | Polycom provisioning is not an Asterisk topic, per se. |
| 05:18.49 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:20.54 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-mhvlsdfkypcvszlm) |
| 05:20.55 | kaldemar | unless done with res_phoneprov. users.conf could be used for provisioning only, maybe. |
| 05:21.23 | p3nguin | Just leave everything commented out that doesn't pertain to provisioning? |
| 05:25.44 | p3nguin | Well, what I mean is that there's a bunch of crap in the sample file that ruins other perfectly good configs. |
| 05:27.54 | *** join/#asterisk KNERD (~KNERD@99.96.118.71) |
| 05:28.00 | Maxus2 | how do i acess a set iaxvar in the dial plan, i have tried simply doing NOOP(IAXVAR(varname)) |
| 05:28.22 | Maxus2 | after doing SET(IAXVAR(varname)=1234) |
| 05:29.10 | kaldemar | Maxus2: ${} |
| 05:29.36 | Maxus2 | so simply NOOP(${varname})? |
| 05:30.57 | kaldemar | Maxus2: you have to put ${} around all variables and functions to access the value. NoOp(${IAXVAR(varname)}) |
| 05:31.20 | Maxus2 | oh doh, sorry. |
| 05:31.31 | Maxus2 | i did know that im doing four lines down, sorry! |
| 05:31.37 | Maxus2 | long day |
| 05:31.42 | Maxus2 | thanks again! |
| 05:41.04 | *** join/#asterisk sahX (~sahX@65-119-47-100.dia.static.qwest.net) |
| 05:42.48 | *** join/#asterisk swk (~swk@freeswitch/developer/swk) |
| 05:53.19 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 05:55.45 | *** join/#asterisk jkroon (~jkroon@dsl-242-7-86.telkomadsl.co.za) |
| 05:55.49 | *** part/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 05:57.44 | godofmonkeys | p3nguin, i got it |
| 05:58.25 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 05:58.36 | godofmonkeys | amazingly the first tiem |
| 05:58.39 | godofmonkeys | time..... |
| 06:04.04 | p3nguin | Which thing? |
| 06:05.35 | godofmonkeys | provisioning |
| 06:08.27 | *** join/#asterisk voila (IceChat77@117.212.71.149) |
| 06:09.10 | voila | hello guys , i want to validate my agi php script , how can i do it ? |
| 06:10.05 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 06:10.28 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 06:11.55 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 06:14.35 | *** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net) |
| 06:16.23 | kaldemar | voila: define validate. |
| 06:17.11 | voila | kaldemar : point is i m very new to agi php , i just want to check ,, whether my dial plan execute my agi php or not .... |
| 06:17.50 | voila | http://pastebin.com/GXbcAbQX .. can you chceck my agi php plz |
| 06:18.58 | kaldemar | attach to asterisk CLI, enable verbosity and agi debug and make a call. you'll see what happens. |
| 06:19.10 | voila | http://pastebin.com/Qc2J4f8A ... this is all my dial plan also |
| 06:20.02 | voila | kaldemar : is my dialplan is correct ? basically calling og php agi |
| 06:21.22 | kaldemar | 1, 4, 5, 6? after 1 comes 2. after 2 comes 3 and so on. however, you should not use numbered priorities expect for 1. use n instead. |
| 06:21.33 | kaldemar | your dialplan is screwed up in many ways. |
| 06:25.48 | kaldemar | so drop the spaces around commas, replace numbered priorities with n and remove the Answer application. |
| 06:34.24 | godofmonkeys | well, i said i got it, the phone grabbs settings, but not what i expect. tring to register as sip:100@192.168.1.200 |
| 06:34.35 | godofmonkeys | wait, that may be my fault |
| 06:36.06 | godofmonkeys | correction, i know it is my fault, just where it lies is another story :/ |
| 06:40.41 | godofmonkeys | sip show peers |
| 06:40.49 | godofmonkeys | blah, wrong keyboard |
| 06:45.29 | *** join/#asterisk ]loy[ (~nobody@95.73.17.224) |
| 06:55.24 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 06:56.17 | godofmonkeys | so i know the opinion of users.conf, but can someone look at mine anyway? http://pastebin.com/eQXAQsM2 |
| 06:56.37 | godofmonkeys | phone gets the user info, but tries to register as 100 |
| 06:57.07 | *** join/#asterisk OrNix (~ornix@l49-131-248.cn.ru) |
| 07:01.37 | *** join/#asterisk oej (~olle@office.ipvision.dk) |
| 07:01.47 | godofmonkeys | output of show dialplan: http://pastebin.com/hpCzbYmq |
| 07:02.05 | *** join/#asterisk rlfx (~rlfx@post.prototechnika.lt) |
| 07:02.06 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 07:08.46 | kaldemar | godofmonkeys: phoneprov.conf and the config files also matter. |
| 07:08.56 | godofmonkeys | one moment |
| 07:09.11 | *** join/#asterisk jeffspeff (~jeff@ip72-202-201-77.fv.ks.cox.net) |
| 07:11.04 | jeffspeff | I've heard here and there that there are problems with stability and quality when running asterisk on debian using sip. can anybody confirm or deny that? I wouldn't think that it would vary that much between linux distros |
| 07:12.08 | godofmonkeys | ahh, stupid mistake.. assumed the reset to factory cleared out all settings on the device, let me look at that.... |
| 07:13.40 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:ecab:5fb5:ffa0:2583) |
| 07:14.22 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 07:15.48 | kaldemar | jeffspeff: won't confirm. |
| 07:16.37 | jeffspeff | ty |
| 07:25.37 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:34.26 | *** join/#asterisk schmidts (~schmidts@2001:858:4:0:66b9:e8ff:fec3:ae88) |
| 07:34.28 | schmidts | good morning |
| 07:35.48 | schmidts | TGIF!! |
| 07:56.37 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 07:57.21 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:57.24 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 07:59.57 | aberrios_ | woo Firmware update Friday. |
| 08:00.08 | aberrios_ | think I'll just call it Firmware Friday |
| 08:01.11 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 08:01.14 | jacc0 | hi all |
| 08:01.56 | aberrios_ | 'Sup |
| 08:02.34 | jacc0 | I have these ghost channels |
| 08:02.36 | jacc0 | http://pastebin.com/9ENjEZMy |
| 08:03.00 | godofmonkeys | jesus. this makes me hurt. back to manual configs. |
| 08:03.17 | jacc0 | what can I do about these channels? |
| 08:03.24 | jacc0 | where do they come from? |
| 08:03.37 | godofmonkeys | auto provisioning is not for me :/ |
| 08:03.46 | jacc0 | using : Asterisk 1.8.3.2 built by root @ ccentersip on a i686 running Linux on 2011-03-21 10:49:29 UTC |
| 08:04.33 | jacc0 | could it be from app_originate? |
| 08:05.34 | jacc0 | last message is Rx: BYE |
| 08:05.43 | jacc0 | why are these channels still active? |
| 08:07.18 | jacc0 | anyone? |
| 08:07.59 | aberrios_ | jacc0, have you check the full log for that Call-ID |
| 08:08.35 | jacc0 | do you mean: sip show channel 060bcfc55830d56 |
| 08:08.36 | jacc0 | ? |
| 08:08.52 | aberrios_ | jacc0, /var/log/asterisk/full |
| 08:09.16 | aberrios_ | jacc0, assuming you're logging at a decent verbosity you should be able to find those calls in the log |
| 08:09.55 | aberrios_ | jacoo, you may need to turn DEBUG on in logger.conf if there's no obvious reason there that you're getting these channels hanging around |
| 08:10.03 | jacc0 | from log: Autodestruct on dialog '060bcfc55830d56216ae852423dd4604@10.100.2.2:5060' with owner in place (Method: BYE) |
| 08:10.23 | aberrios_ | pastebin it in context |
| 08:12.16 | jacc0 | http://pastebin.com/BU40UyVF |
| 08:13.26 | jacc0 | the device closes the line after it recieves DTMF * |
| 08:13.27 | aberrios_ | line 11 is where its going tits up, turn debug on and verbosity up and replicate the issue again and take another look at the logs |
| 08:14.14 | jacc0 | okay, will do |
| 08:14.41 | jacc0 | full => notice,warning,error,debug,verbose,dtmf,fax |
| 08:14.49 | jacc0 | how can I get more login detail? |
| 08:14.56 | aberrios_ | hmm |
| 08:15.45 | aberrios_ | CLi> debug on |
| 08:16.21 | jacc0 | core set debug 9 |
| 08:17.08 | aberrios_ | yes, brb reboot |
| 08:17.52 | *** join/#asterisk atiti (~atiti@nbbd.imm.dtu.dk) |
| 08:18.18 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:18.42 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 08:19.01 | kaldemar | jacc0: does core show channels also list you channels? |
| 08:19.23 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:19.45 | kaldemar | jacc0: how long have those _same_ channels been listed by sip show channels? |
| 08:19.48 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 08:20.33 | *** join/#asterisk johanwilfer (~marvin@static-213-115-34-198.sme.bredbandsbolaget.se) |
| 08:21.06 | kaldemar | the broken pipe error is about the AGI. |
| 08:22.04 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 08:22.19 | kaldemar | jacc0: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg231722.html |
| 08:22.30 | jacc0 | core show channels doesn't show the channels |
| 08:22.37 | jacc0 | while sip show channels does |
| 08:23.06 | jacc0 | they have been arround for hours |
| 08:23.28 | jacc0 | broken pipe is about agi or ami : closed connection to fast |
| 08:23.35 | jacc0 | i guess |
| 08:28.25 | jacc0 | but does that result in these ghost channels? |
| 08:28.50 | jacc0 | or aren't these channels realy there? because core show channels shows nothing? |
| 08:30.12 | jacc0 | and I think the segfaults/core dups that asterisk is giving me are a result of to many ghost channels |
| 08:30.49 | kaldemar | don't think so. the broken pipe is a separate issue. look at sip debug for 10.100.1.4 when you make that call and see what happens on the signalling level. |
| 08:31.42 | jacc0 | should I use tshark to get a more objective view on that? |
| 08:31.55 | kaldemar | no, start with asterisk. |
| 08:31.59 | jacc0 | k |
| 08:32.17 | jacc0 | I would have to monitor for some time; it doesn't happen al the time |
| 08:32.21 | kaldemar | then you'll see what asterisk is seeing. |
| 08:33.53 | kaldemar | what version are you using? |
| 08:34.08 | kaldemar | oh, 1.8.3.2 it is. |
| 08:34.16 | jacc0 | <PROTECTED> |
| 08:36.07 | jacc0 | okay, I cleared the ghost channels by doing a core restart now, now I'll just keep open the cli till the apear again |
| 08:36.39 | jacc0 | (btw core restart now made me lose the astcanary; better use /etc/init.d/assterisk restart next time) |
| 08:37.06 | kaldemar | setting session timers in sip.conf could help you if the issue can't be fixed otherwise. |
| 08:37.43 | jacc0 | under general? (using asterisk realtime) |
| 08:39.00 | kaldemar | under general or per peer. see the sample config. |
| 08:40.15 | jacc0 | session-timers=originate |
| 08:40.17 | jacc0 | ? |
| 08:42.13 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 08:44.52 | kaldemar | jacc0: yes. |
| 08:46.02 | jacc0 | could these be unaswerd originated calls? |
| 08:46.20 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 08:47.50 | jacc0 | i've modded the hardcoded timeout of 30 seconds of app_originate.c to 36000 |
| 08:48.08 | jacc0 | (that's the only sourcecode modification I made) |
| 08:48.31 | jacc0 | a hardcoded timeout of 30seconds make app_originate a bit useless |
| 08:49.00 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:52.19 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 08:54.29 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:54.31 | jacc0 | don't you just love code hacking on the bleeding edge ? :P |
| 08:58.43 | kaldemar | 10 hours? isn't that a bit overkill? |
| 08:58.53 | *** join/#asterisk c4rg (crg@lagoon.freebsd.lublin.pl) |
| 08:59.02 | c4rg | how does one disable asterisk's realtime cache? |
| 08:59.04 | jacc0 | uhm, yeah could be |
| 09:01.00 | kaldemar | c4rg: what cache? |
| 09:01.42 | jacc0 | sip.conf : rtcachefriends=no |
| 09:01.54 | jacc0 | rtcache=no |
| 09:02.02 | jacc0 | rtcachepeers=no |
| 09:02.11 | jacc0 | rtsavesysname=no |
| 09:02.45 | jacc0 | rtautoclear=yes |
| 09:03.21 | jacc0 | ignoreregexpire=no |
| 09:04.00 | kaldemar | rtcache and rtcachepeers don't exist. |
| 09:05.35 | jacc0 | k |
| 09:07.51 | c4rg | queue table/queue members cache |
| 09:08.00 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227) |
| 09:08.52 | c4rg | I guess sip.conf cache is/should be different from queue rt cache? |
| 09:09.04 | c4rg | it worked fine in 1.4.x |
| 09:09.16 | c4rg | with 1.6 it's caching the information, which it shouldn't |
| 09:15.39 | kaldemar | c4rg: what version? |
| 09:16.37 | c4rg | 1.6.2.9-1~bpo50+3 (Debian) |
| 09:19.33 | c4rg | extconfig.conf is like queues => pgsql... queue_members => pgsql... |
| 09:19.47 | c4rg | I had simmiliar problems with odbc |
| 09:20.01 | c4rg | which means something is caching old information |
| 09:20.02 | kaldemar | the queue realtime data seems to be re-read when the queue app is used. doesn't look like you can disable caching. |
| 09:20.28 | c4rg | queue show xxx |
| 09:20.35 | c4rg | shows cached information |
| 09:20.40 | kaldemar | yes it does. |
| 09:20.46 | c4rg | also when Queue() app is started it uses old information |
| 09:21.00 | c4rg | it worked just fine in 1.4 |
| 09:21.55 | c4rg | realtime show pgsql cache |
| 09:22.01 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227) |
| 09:22.06 | c4rg | shows that queue members table is cached :\ |
| 09:29.55 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 09:30.57 | *** join/#asterisk sahX_ (~sahX@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 09:31.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 09:31.55 | *** join/#asterisk kchehab (~kchehab@212.98.141.202) |
| 09:32.08 | kaldemar | and it doesn't update the members when the queue app is executed? 1.4 and 1.6.2 seem to quite similar in that manner. |
| 09:32.34 | c4rg | unfortunately it's using the old information |
| 09:32.53 | kchehab | anyone have a idea how to capture the fax sent by clients on asterisk |
| 09:34.14 | kchehab | i used to recieve and send it |
| 09:34.38 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 09:34.40 | kchehab | but that is not a real time |
| 09:35.51 | Cadey | Hi guys, before I get stuck in checking my code, is tehre any reason why asterisk would boot off a manager connection if its listening for too long and doesnt send any new message to the server, so a listening only connection ? |
| 09:36.38 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:36.57 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 09:37.19 | kaldemar | Cadey: http or plain AMI? |
| 09:37.29 | Cadey | AMI |
| 09:38.30 | kaldemar | try writetimeout parameter in manager.conf. |
| 09:39.24 | Cadey | we have not set that param |
| 09:39.30 | Cadey | ill check what its default is |
| 09:39.49 | Cadey | as the user has both read and write so it could be that |
| 09:41.16 | Cadey | although this is talking about MS |
| 09:41.28 | Cadey | and this is doconnecting over night I thikn |
| 09:41.35 | Cadey | so when the system is proty much doing nothing |
| 09:43.42 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 10:01.49 | *** join/#asterisk no1peanut (~rudolf@h59ec0d32.stgertrud.dyn.perspektivbredband.net) |
| 10:01.53 | no1peanut | when using an applicationmap in features.conf is the dtmf string saved in some variable ? |
| 10:03.20 | *** join/#asterisk voila (IceChat77@117.212.71.108) |
| 10:03.41 | voila | hello guys .. how can i debug my agi php script ?? i m new to asterisk |
| 10:03.50 | kaldemar | no1peanut: doubt it. how so? |
| 10:04.21 | kaldemar | deja vu |
| 10:04.24 | kaldemar | voila: attach to asterisk CLI, enable verbosity and agi debug and make a call. |
| 10:04.49 | voila | kaldemar: how do i attach my php to asterisk client ?? |
| 10:04.56 | kaldemar | voila: if you're still struggling with the same dialplan, your AGI won't even execute. |
| 10:05.47 | kaldemar | voila: i never said "client", but CLI as in command line interface. you attach to it from shell with "asterisk -vvvr". you'll see output when stuff happens. |
| 10:06.50 | no1peanut | kaldemar, I would like to inject audio into a call, different depending on which dtmf key is pressed and have multiple audio file on each key |
| 10:08.10 | kaldemar | no1peanut: and why do you need the DTMF sequence in a variable for that? you have it in features.conf, right? |
| 10:12.02 | no1peanut | kaldemar, yes |
| 10:12.53 | kaldemar | ${CUT(AST_CONFIG(features.conf,applicationmap,featurename),",",1)} will get it from the file if you wish. |
| 10:13.09 | kaldemar | replace featurename with your own. |
| 10:14.05 | voila | kaldemar: i did , what you have said .. and in my php, i did this "agi->answer()" .... why i m getting decline on my softphone ? |
| 10:16.31 | kaldemar | voila: your dialplan executes the Hangup application |
| 10:16.45 | kaldemar | voila: what are you expecting it to do? |
| 10:16.52 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 10:18.02 | voila | kaldemar:http://pastebin.com/SzCkgdPP |
| 10:18.10 | voila | this is all .. i m trying to do |
| 10:18.27 | voila | very simple .. yet not able to do .. :( |
| 10:18.45 | *** join/#asterisk m_tadeu (~quassel@a89-152-97-133.cpe.netcabo.pt) |
| 10:18.50 | voila | i m dialing 1234 extension |
| 10:19.28 | kaldemar | for the third time. attach to CLI to see what happens when you make a call. |
| 10:20.00 | kaldemar | there are a few dozen things that could be wrong. |
| 10:20.24 | voila | ok .. i m using windows ... so .. cmd> c:\cyroot\bin\asterisk -c -vvvr |
| 10:20.32 | voila | then i make a call |
| 10:21.24 | voila | Apr 1 15:50:53 WARNING[8096]: channel.c:788 channel_find_locked: Avoided initia |
| 10:21.24 | voila | l deadlock for '0x8b43b8', 9 retries! |
| 10:21.25 | kaldemar | ~pb |
| 10:21.26 | 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. |
| 10:21.59 | voila | Apr 1 15:50:53 WARNING[8096]: channel.c:788 channel_find_locked: Avoided initia |
| 10:22.00 | voila | l deadlock for '0x8b43b8', 9 retries! |
| 10:22.02 | voila | srry |
| 10:22.12 | voila | http://pastebin.com/DSfiEGP3 |
| 10:23.43 | kaldemar | AsteriskWin32... that's a package i don't want to touch. |
| 10:24.18 | kaldemar | do yourself a favor and install a linux distribution and use real asterisk. |
| 10:24.35 | voila | okies ... |
| 10:24.43 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 10:24.54 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 10:27.17 | *** join/#asterisk Cadey (~Cadey@62.84.178.106) |
| 10:28.13 | Cadey | Hi guys, I need to call a group of external numbers (for a police response call) and i was wondering which is the best way to do this, for example is simply putting multiple numbers into the dial app the best way? |
| 10:39.23 | *** join/#asterisk aberrios_ (~aberrios@195.171.4.82) |
| 10:42.34 | *** join/#asterisk felipe_ (~felipe@unaffiliated/felipe) |
| 10:50.32 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 10:52.13 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-qvcakkxydgvoltoz) |
| 11:21.06 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 11:21.55 | *** join/#asterisk zkn (~zkn@213.115.26.228) |
| 11:28.30 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:31.22 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 11:31.53 | *** join/#asterisk wonderworld (~ww@port-92-201-26-77.dynamic.qsc.de) |
| 11:41.27 | *** part/#asterisk tasca (~tasca@189.34.27.64) |
| 11:46.38 | *** join/#asterisk davlefou (~david@41.227.53.55) |
| 11:52.50 | *** join/#asterisk vampi-the-frog (~zyx@mail.cmbtravel.ro) |
| 11:52.53 | vampi-the-frog | hi guys :D |
| 11:53.00 | vampi-the-frog | glad to have found this channel :D |
| 11:53.53 | tuxx- | we are also glad that you found this channel. |
| 11:53.59 | tuxx- | did you bring any beer / pie? |
| 11:53.59 | tuxx- | :X |
| 11:54.18 | vampi-the-frog | i have some pepsi |
| 11:54.31 | vampi-the-frog | but it's mine |
| 11:54.32 | vampi-the-frog | so no |
| 11:54.43 | vampi-the-frog | unless you want a lie pie |
| 11:55.02 | tuxx- | :( |
| 11:56.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:56.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:01.01 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:08.12 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 12:16.18 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 12:24.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:24.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:25.19 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:41.13 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 12:46.41 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:46.41 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:47.25 | Cadey | Anyone here got much experience on the best way to call multiple external numbers. I need to set up a group of people that will get a call when a extension is dialed, the best way I can see so far is to simply put all the numbers into the same Dial(Number1&Number2&Etc) Do any of you guys use a diffrent way? |
| 12:51.19 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 12:57.47 | *** join/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 12:58.04 | epart | hI Guys |
| 12:58.22 | epart | anyone familliar with vicidial? |
| 13:00.44 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 13:01.16 | orn | Cadey: You could use queues as well, but the method you're using is fine as well |
| 13:01.26 | orn | haha, two as wells in the same sentence |
| 13:02.01 | Cadey | you can put external numbers in a queue ? |
| 13:03.08 | Cadey | I was thinking only peers could go into queues |
| 13:03.23 | Cadey | but i guess it will call memebrs of the queue just like that string was in a dial |
| 13:03.51 | orn | yes, you can put external numbers there as well |
| 13:06.02 | Cadey | this may sound like a stupid question but can queues see global variables in the dial plan? |
| 13:08.24 | johanwilfer | Cadey: You can use ${GLOBAL(myvar)} |
| 13:10.01 | Cadey | inside queue.conf ? |
| 13:10.02 | johanwilfer | Cadey: And you can call with the local technology, so your call end up in the dialplan |
| 13:11.04 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 13:11.31 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:11.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:12.07 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:12.08 | Cadey | it seems putting external numbers into the queue does not work? |
| 13:12.53 | Cadey | arg |
| 13:13.01 | johanwilfer | Cadey: In the queue.conf you can have a queue with a persistant member like: |
| 13:13.08 | johanwilfer | Cadey: [demo] |
| 13:13.21 | *** join/#asterisk tonsofpcs (~tonsofpcs@april-fools/thesixthest/tonsofpcs) |
| 13:13.27 | johanwilfer | Cadey: member => Local/ext@context |
| 13:13.57 | johanwilfer | Cadey: And then you can have all the magic in the dialplan |
| 13:14.03 | Cadey | well to make external calls in a dial i use Dial(MyProvider/TheNumber) |
| 13:14.11 | Cadey | but when I put member => MyProvider/Number |
| 13:14.14 | Cadey | is does not work? |
| 13:14.18 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
| 13:14.26 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
| 13:14.42 | johanwilfer | You can try, I guess it should work |
| 13:14.57 | Cadey | thats jsut it, it dont work (: |
| 13:14.58 | johanwilfer | I've my magic in the dialplan :) |
| 13:15.18 | Cadey | dont worry ill just put multiple numbers in teh dial :) |
| 13:15.21 | Cadey | i know that works |
| 13:15.47 | johanwilfer | I use the method above, it work. |
| 13:17.05 | johanwilfer | works even.. |
| 13:17.07 | orn | on that note, has anyone ever wanted to invite a very large group of friends to the movies and have all of their cell phone numbers called at the same time? |
| 13:17.12 | orn | :) |
| 13:17.48 | Cadey | oh my... im so stupid :( |
| 13:17.57 | epart | can you help me how to install asterisk in ubuntu 10.10 maverick |
| 13:18.10 | johanwilfer | or wake them up in the saturday morning ;) |
| 13:18.50 | johanwilfer | Cadey: ? |
| 13:20.29 | *** join/#asterisk m_tadeu (~quassel@static-b5-252-50.telepac.pt) |
| 13:22.34 | *** join/#asterisk coppice (~chatzilla@9.160.232.220.dyn.pacific.net.hk) |
| 13:23.31 | Cadey | I forgot to put SIP/ infront of my provider in the quqe |
| 13:23.33 | Cadey | queue |
| 13:23.38 | Cadey | real face palm moment :( |
| 13:23.53 | johanwilfer | Aha |
| 13:24.34 | *** join/#asterisk mzb_ (~mzb@ppp108-88.static.internode.on.net) |
| 13:25.08 | jaytee | i have a permanent palm print on my face from too many of those moments :-) |
| 13:26.15 | m_tadeu | how can I set the sounds directory in asterisk 1.8? |
| 13:26.32 | serafie | m_tadeu: /etc/asterisk/asterisk.conf |
| 13:26.40 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:28.16 | m_tadeu | serafie: I only can see the prefix, which I'm guessing is /var/lib/asterisk/sounds/PREFIX...can you tell me where to set this prefix? |
| 13:30.22 | serafie | m_tadeu: change astdatadir |
| 13:30.38 | serafie | I believe the "sounds" subdirectory of astdatadir is not changeable. |
| 13:31.15 | m_tadeu | serafie: I see...and will that only affect the sounds directory, or will I have some other side effects? |
| 13:31.52 | *** join/#asterisk josta (~josta@unaffiliated/josta) |
| 13:31.58 | *** join/#asterisk dms (~Adium@nat/digium/x-wxundfhxouogunhq) |
| 13:32.36 | Cadey | does reloading the dialplan or queue reload all cause the manage AMI to dump its connections? |
| 13:32.57 | Cadey | as I think its just happened lol |
| 13:33.18 | serafie | m_tadeu: only sounds, and that same directory is used for Record() |
| 13:34.28 | m_tadeu | I see...thanx a lot |
| 13:39.43 | m_tadeu | asterisk doesn't seem to find the sound files in any format, dispite the fact I have .wav, .sln, .mp3, .ulaw and gsm for all sounds... |
| 13:39.56 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:40.01 | *** part/#asterisk dms (~Adium@nat/digium/x-wxundfhxouogunhq) |
| 13:40.15 | Cadey | is there a max amount an AMI connection can read before asterisk boots the connection? |
| 13:40.35 | Cadey | as we seem to have our AMI connection killed off when its left constantly connected |
| 13:42.55 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:44.19 | m_tadeu | is there a way to check the sounds directory in the asterisk console? |
| 13:44.33 | Cadey | check it how? |
| 13:44.56 | m_tadeu | for asterisk to show me which dir is using for sounds |
| 13:45.59 | Cadey | isnt that in asterisk.conf ? |
| 13:46.54 | jaytee | sounds are usually in /var/lib/asterisk/sounds/"language" |
| 13:47.14 | m_tadeu | it is...but I'm having trouble on playing sound files, which I have in the formats described above, and asterisk says it doesn't find the file in any format |
| 13:47.19 | jaytee | and it is set in asterisk.conf like Cadey said |
| 13:47.53 | jacc0 | hmm, I'm using asterisk 1.8 in a realtime configuration |
| 13:48.25 | jacc0 | the logging tels me asterisk was unable to open the asteriskAccount table (using a named pipe) |
| 13:48.48 | jaytee | m_tadeu, how are you referencing the sound files in extensions.conf? can you pastebin an example? |
| 13:48.56 | jacc0 | but the dialplan that uses MYSQL (localhost) worked fine |
| 13:49.13 | jacc0 | :S |
| 13:49.30 | jacc0 | so the database seemed to be available using the socket |
| 13:50.27 | *** join/#asterisk moy (~moy@bas5-toronto47-1088894188.dsl.bell.ca) |
| 13:50.45 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 13:51.03 | jacc0 | does anyone have a clue how this could have happend? |
| 13:51.17 | m_tadeu | jaytee: http://pastebin.com/W95tuu2v |
| 13:51.19 | jacc0 | asterisk was using up 100% CPU at that moment |
| 13:51.44 | jacc0 | after /etc/init.d/asterisk restart |
| 13:51.45 | Cadey | im sorry but i dont use realtime config : |
| 13:51.47 | jacc0 | everything was fine |
| 13:52.29 | *** join/#asterisk tasca (~tasca@189.34.27.64) |
| 14:04.31 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:05.05 | tuxx- | heya guys, i got the following error: [Mar 30 16:04:06] ERROR[22812]: res_musiconhold.c:645 monmp3thread: poll() failed: Interrupted system call. Anyone know how to fix this? Can't seem to find anything on the internets :-( |
| 14:06.30 | Faustov | can anyone recommend a datacenter in London? ;) |
| 14:08.05 | *** part/#asterisk tasca (~tasca@189.34.27.64) |
| 14:08.16 | jaytee | m_tadeu, can you also pastebin the Directories section of your asterisk.conf file and a paste from doing ls /var/lib/asterisk/sounds ? |
| 14:08.20 | *** part/#asterisk i___ (~killown@unaffiliated/killown) |
| 14:11.07 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:11.24 | jaytee | m_tadeu, and if you make a call to VoicemailMain do you hear the prompts or get the same error? |
| 14:13.33 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:13.50 | jacc0 | can anybody help me with the named pipe not being available? and what about asterisk running at 100%? |
| 14:14.01 | jacc0 | is there a astcanary log somewhere? |
| 14:14.31 | m_tadeu | jaytee: http://pastebin.com/HNzvx9Zg....how do I make that call(shouldn't be this ignorant by now)? |
| 14:15.48 | serafie | m_tadeu: we just talked about changing the location of the sounds directory. Did you do that? If so, did you move all the default sounds? |
| 14:16.33 | serafie | m_tadeu: ls -lah /usr/share/asterisk/sounds/en (or whatever language you're using) |
| 14:16.52 | epart | any one knows how to install vicdial? |
| 14:17.00 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:17.02 | m_tadeu | serafie: I didn't change the directory...only set the languageprefix = no and put the sounds in the /var/lib/asterisk/sounds dir |
| 14:17.23 | serafie | m_tadeu: in that pastebin of your asterisk.conf, astdatadir is set to /usr/share/asterisk |
| 14:18.34 | m_tadeu | serafie: crap...how did I miss that :( |
| 14:18.38 | *** join/#asterisk Sipster (~Sipster@modemcable143.199-202-24.mc.videotron.ca) |
| 14:18.48 | m_tadeu | serafie: thanx |
| 14:18.51 | serafie | welcome! |
| 14:22.07 | jaytee | m_tadeau, fixing that path should fix your sound issue |
| 14:23.10 | m_tadeu | serafie, jaytee: unfortunatelly, even restarting asterisk, it didn't solve the problem |
| 14:26.04 | m_tadeu | I changed asterisk.conf to astdatadir => /var/lib/asterisk...so the sounds dir should be /var/lib/asterisk/sounds, right? |
| 14:26.19 | serafie | yes |
| 14:26.36 | jaytee | correct |
| 14:26.55 | serafie | m_tadeu: you changed languageprefix from yes to no, or was it always no? |
| 14:27.10 | m_tadeu | sear |
| 14:27.13 | serafie | In any case, if it was working 30 mins ago, you should go back and make your desired changes again. |
| 14:27.27 | m_tadeu | serafie: it was commented, and I think the default is yes |
| 14:28.05 | jaytee | the default is yes and if you have english set as a default then it would look in /var/lib/asterisk/sounds/en |
| 14:28.36 | m_tadeu | serafie: it never worked...but I'll put it back as it was and try then....where do I set the language? |
| 14:28.50 | jaytee | it defaults to english |
| 14:28.54 | Cadey | does the dialplan have select/case logic or do we just have to use If's and GotoIf etc |
| 14:30.20 | serafie | Cadey: just GoToIf |
| 14:30.22 | jaytee | Cadey, no select/case. you'd have to use multiple GotoIfs |
| 14:30.31 | Cadey | no probs :) |
| 14:30.33 | Cadey | jsut wondered |
| 14:30.44 | jacc0 | I guess there is no astcanary log? |
| 14:30.48 | m_tadeu | jaytee, serafie, no luck....still not finding the files |
| 14:31.01 | serafie | what is the full path to the file you are trying to play? |
| 14:31.07 | serafie | jacc0: what information are you looking for? |
| 14:31.51 | jaytee | m_tadeau, what if you change a Playback statement in extensions.conf to point to the explicit path, i.e. Playback(/var/lib/asterisk/sounds/somesound) |
| 14:31.52 | jacc0 | I want to know if it was running |
| 14:32.08 | m_tadeu | serafie: /var/lib/asterisk/sounds/intro.gsm(I'm forcing gsm in the sip client) |
| 14:32.22 | jacc0 | bacause asterisk was taking up 100% cpu; astcanary should have restarted it or not? |
| 14:32.55 | m_tadeu | serafie: I also copyed all sound files to /var/lib/asterisk/sounds/en |
| 14:33.36 | serafie | jacc0: I'm only looking at the source code, here, but there doesn't appear to be any log statements in astcanary.c |
| 14:34.03 | serafie | m_tadeu: are you calling the file with the extension, i. e., Playback(intro.gsm) ? |
| 14:34.48 | jaytee | in his paste he doesn't use the file extension |
| 14:34.55 | m_tadeu | serafie: Playback(intro) |
| 14:34.56 | jacc0 | 2 bad |
| 14:34.58 | jaytee | and you shouldn't |
| 14:35.02 | serafie | is asterisk running as root? |
| 14:35.29 | m_tadeu | serafie: it is...as a daemon |
| 14:35.34 | serafie | watch the CLI after running "core set verbose 10" and see if you see anything funky. |
| 14:35.45 | serafie | while the call is happening. |
| 14:36.02 | jaytee | gotta run out to a client, bbl |
| 14:36.20 | serafie | you're certain your call is using the correct dialplan? Couldn't be that the statement is never being processed, could it? |
| 14:37.13 | m_tadeu | serafie: yes...I have the logs in the console...it goes through every step of the dialplan...I'll pastebin it for you |
| 14:37.59 | m_tadeu | serafie: http://pastebin.com/0szLYW2p |
| 14:38.18 | serafie | m_tadeu: [Apr  1 15:37:33] WARNING[8635]: file.c:644 ast_openstream_full: File intro does not exist in any format |
| 14:38.31 | serafie | so it is still using the wrong directory |
| 14:39.00 | m_tadeu | serafie: in deed....but I already restarted it several times after changing it |
| 14:39.35 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 14:39.47 | serafie | m_tadeu: look at Data Directory in "core show settings" |
| 14:40.25 | m_tadeu | serafie: it's still using the old dir....howcome? |
| 14:40.33 | serafie | It never reloaded |
| 14:41.35 | serafie | the CLI is your best friend. :) |
| 14:42.51 | m_tadeu | serafie: in deed...I was asking for this command before :) anyways, howcome it's getting the old config? I'm restarting the service over and over |
| 14:43.00 | serafie | how? |
| 14:43.40 | m_tadeu | 2 ways...service asterisk restart and /etc/init.d/asterisk restart(just in case) |
| 14:44.24 | serafie | perhaps try service asterisk stop and then start. |
| 14:44.31 | jaytee | what distro? |
| 14:44.55 | m_tadeu | jaytee: ubuntu server |
| 14:45.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:45.23 | serafie | ps ax | grep asterisk in between to make sure it is good and dead. |
| 14:45.36 | jaytee | IIRC, with Ubuntu you have to use /etc/init.d/asterisk restart |
| 14:45.43 | m_tadeu | jaytee: ubuntu serveri did that...it comes down |
| 14:46.10 | *** join/#asterisk ariel_ (~abatista@63.214.236.169) |
| 14:46.18 | jacc0 | now I have problems on some other server : channel.c: write() failed: Resource temporarily unavailable |
| 14:46.21 | *** join/#asterisk Tim_Toady (~moi@79.103.49.227.dsl.dyn.forthnet.gr) |
| 14:46.37 | jacc0 | the asterisk process stoped |
| 14:47.31 | *** join/#asterisk davlefou (~david@41.225.222.116) |
| 14:50.29 | jacc0 | I'm losing faith in the stability of asterisk 1.8 |
| 14:50.31 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:50.43 | jacc0 | :S |
| 14:51.01 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 14:51.17 | jacc0 | the fact that astcanary was ever invented tels a stroy |
| 14:51.22 | m_tadeu | serafie: when I do a core reload, shouldn't there be a line sayin something like "parsing /etc/asterisk/asterisk.conf"? |
| 14:52.53 | serafie | m_tadeu: no, you have to do a full restart when you change asterisk.conf |
| 14:53.08 | serafie | core restart now|gracefully|when convenient |
| 14:54.17 | m_tadeu | serafie: ahhh but the console gets disconnected and I can't see that log |
| 14:55.57 | serafie | you can see if it you do core stop now and then start asterisk with asterisk -gcvvvvvvvvvvv |
| 15:05.47 | m_tadeu | serafie: asterisk doesn't report any problem parsing asterisk.conf |
| 15:05.59 | *** join/#asterisk ccccp123 (~IceChat77@cpc1-acto12-0-0-cust537.4-2.cable.virginmedia.com) |
| 15:06.11 | serafie | core show settings still shows /usr/share/asterisk? |
| 15:06.21 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:07.42 | m_tadeu | it does |
| 15:08.14 | serafie | what does it show for Configuration directory: |
| 15:08.31 | m_tadeu | /etc/asterisk |
| 15:09.31 | jacc0 | bye all |
| 15:09.35 | jacc0 | have a nice weekend!!! |
| 15:09.41 | jacc0 | thanks for you help and support |
| 15:10.51 | ccccp123 | [custom-callback] |
| 15:10.51 | ccccp123 | exten => s,n,NoCDR() |
| 15:11.02 | ccccp123 | this is not working |
| 15:11.44 | ccccp123 | I am trying to disabled CDR for callback |
| 15:11.54 | serafie | m_tadeu: go to commandline and run cat /etc/asterisk | grep astdatadir |
| 15:12.04 | serafie | to make double sure the correct value is written in the file |
| 15:12.10 | serafie | not just displayed in a text editor somewhere |
| 15:12.15 | serafie | then stop asterisk completely |
| 15:12.49 | m_tadeu | astdatadir => /var/lib/asterisk |
| 15:13.03 | serafie | did you ever run ps ax | grep asterisk ? |
| 15:13.21 | serafie | when asterisk was supposed to be stopped? |
| 15:13.23 | m_tadeu | in deed...to make sure asterisk was stoping |
| 15:14.25 | serafie | OK, change astdatadir to some other directory that isn't /var/lib/asterisk or /usr/share/asterisk and try to get asterisk to pick up the change. |
| 15:14.33 | serafie | what version of asterisk? |
| 15:14.51 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 15:17.30 | m_tadeu | serafie: it's 1.8.3 |
| 15:17.50 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 15:18.14 | serafie | see if you can get it to pick up nonsense changes in the directories (but not astetcdir) |
| 15:18.26 | serafie | like astdatadir=/home/user/existing/directory |
| 15:18.42 | serafie | not a directory that doesn't exist, though. |
| 15:19.08 | m_tadeu | serafie: didn't change at all |
| 15:19.11 | *** topic/#asterisk by Qwell -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 2.0-beta1 (2011/04/01), 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:19.29 | serafie | Could it be something simple like you're in the wrong terminal or on the wrong server? |
| 15:19.57 | serafie | This is the point at which I would reboot. |
| 15:20.15 | m_tadeu | just to make sure, I'll go to the server and try it....I'm using ssh, but the server is here |
| 15:21.20 | m_tadeu | same result on the server.... |
| 15:22.33 | m_tadeu | is there some other way to force this directory? |
| 15:23.07 | serafie | not that I can see. |
| 15:23.28 | serafie | You can force it to use another main config file with -C |
| 15:26.12 | ccccp123 | how disabled CDR for callback? |
| 15:27.56 | *** join/#asterisk mateu (~mateu@suryahunter.com) |
| 15:28.47 | m_tadeu | serafie: none of the directories seem to change...I tryed the modules dir with success |
| 15:31.23 | *** join/#asterisk jamko (~chatzilla@173.160.6.201) |
| 15:31.34 | Qwell | m_tadeu: pastebin your asterisk.conf |
| 15:32.37 | jamko | Anyone here using voip.ms or vitality and have any information on the USA origination problem? I'm getting killed here and voip.ms doesn't even have a cause for the issue 2 hours in. |
| 15:33.16 | m_tadeu | serafie: http://pastebin.com/DY924TrZ |
| 15:33.43 | serafie | where are your directory definitions? |
| 15:33.50 | Qwell | what she said |
| 15:34.20 | m_tadeu | serafie: sorry...bad copy |
| 15:35.02 | m_tadeu | serafie: http://pastebin.com/YCypKS3K |
| 15:36.15 | m_tadeu | serafie: please ignore the modules dir, as I was testing |
| 15:36.38 | serafie | Maybe there's a funky character in all that whitespace... I'm pretty much grasping at straws here. |
| 15:37.04 | ccccp123 | how disabled CDR for callback? |
| 15:38.24 | m_tadeu | serafie: wouldn't asterisk complain about something like that? |
| 15:39.10 | m_tadeu | I'll write it by hand, just to make sure |
| 15:39.20 | serafie | I've never seen it. |
| 15:39.30 | serafie | I wouldn't assume it would, though |
| 15:45.16 | ccccp123 | how disabled CDR for callback? |
| 15:46.11 | Qwell | m_tadeu: (!) means something is a template. If nothing is using that template, they aren't applied. |
| 15:46.21 | Qwell | remove the (!) from the end of the [directories] line |
| 15:47.15 | m_tadeu | .....crap.... |
| 15:49.10 | m_tadeu | Qwell,serafie: thanx a lot...it's setting the dirs now |
| 15:49.36 | ccccp123 | how disabled CDR for callback? |
| 15:49.47 | Qwell | ccccp123: you don't need to ask every 5 minutes |
| 15:50.05 | ccccp123 | every 10 will help? |
| 15:50.15 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ctvfjodlpajiqtyb) |
| 15:50.45 | Qwell | Ask 1 time. If somebody wants to answer, they will. |
| 15:51.30 | ccccp123 | ok? this is just in case if some one joins and want to answer.ok? |
| 15:51.58 | Qwell | ...nobody joined between 3 of the times you asked. |
| 15:52.45 | ccccp123 | ok i will check b4 repeating |
| 15:54.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:55.22 | *** join/#asterisk voila (IceChat77@117.212.73.173) |
| 15:56.02 | ccccp123 | how disabled CDR for callback? |
| 15:56.12 | Qwell | rolls his eyes |
| 15:57.19 | *** join/#asterisk sahX (~sahX@4.53.128.211) |
| 15:58.33 | voila | hello guys .. im trying to run AGI php on asterisk Win32 sincd morning , but failed .. can anybody take a look at verbose output .. http://pastebin.com/LU7bHqmE |
| 15:58.44 | voila | any one here usinh asterisk win32 ?? |
| 15:58.52 | Qwell | ... |
| 15:58.54 | Qwell | I quit. |
| 15:59.04 | voila | Qwell: :( |
| 15:59.22 | Qwell | voila: astwin32 was an April Fools joke from 8 years ago. |
| 15:59.30 | voila | :( |
| 15:59.41 | chuckf | Qwell: and give up on the humor the channel brings? |
| 15:59.50 | voila | Qwell: hey is it possible to run UNIX asterisk on windows ?? |
| 15:59.51 | Qwell | chuckf: humor? |
| 16:00.03 | wonderworld | is there a way to compile and install asterisk into a single directory, for example /opt/asterisk ? |
| 16:00.08 | Qwell | voila: No. Install a real operating system. |
| 16:00.36 | voila | Qwell: ok suppose, then are there softphones avlailable for linuxss ?? |
| 16:00.42 | Qwell | wonderworld: ./configure --prefix=/opt/asterisk --sysconfdir=/opt/asterisk/etc --somethingelse=/opt/asterisk/something |
| 16:00.46 | Qwell | datadir, I think, maybe? |
| 16:00.53 | wonderworld | Qwell: thanks a lot |
| 16:01.25 | Qwell | wonderworld: localstatedir |
| 16:01.32 | Qwell | I think |
| 16:01.43 | wonderworld | Qwell: where would i look up a complete list of configure options? |
| 16:01.50 | Qwell | --localstatedir=/opt/asterisk/var |
| 16:01.53 | Qwell | wonderworld: ./configure --help |
| 16:01.59 | wonderworld | hehe ok thans |
| 16:02.28 | Qwell | voila: many, but...why would you need a softphone for Linux? Softphone doesn't have to run on the same system as Asterisk. |
| 16:03.16 | voila | Qwell: yaa i know . but i have only one system .. so for testing , i have install softphone on linux too ..:P |
| 16:05.09 | *** join/#asterisk micols (~mio@rlogin.dk) |
| 16:05.11 | wonderworld | twinkle is my favourite linux softphone |
| 16:05.31 | Qwell | voila: vmware.com - thank me later. |
| 16:05.51 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 16:06.55 | voila | virtualbox.com ... thanks me now :P :P |
| 16:08.36 | wonderworld | yawn. compiling asterisk is soooo slow on my crappy hardware |
| 16:10.33 | pabelanger | wonderworld: distcc FTW |
| 16:13.28 | wonderworld | yeah, i need to try it out next time |
| 16:13.31 | ccccp123 | how disabled CDR for callback? |
| 16:14.10 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 16:14.40 | Qwell | ccccp123: dude, seriously. stop it. |
| 16:14.55 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:15.47 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:17.19 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:23.28 | *** join/#asterisk mechbangirc (~cheetah@mbl-82-58-225.dsl.net.pk) |
| 16:24.34 | ccccp123 | <@Qwell> well I will any chance you could help? in extention_additional under [callback-custom] i am having exten => s,1,NoCDR() but it is not working |
| 16:24.42 | Kobaz | what's a good pci-e network card that has good linux support |
| 16:25.14 | Qwell | ccccp123: We don't support freepbx here. Try #freepbx |
| 16:25.39 | mechbangirc | can someone tell me is it possible that two parties talk to eachother using SIP without sending an ANSWER signal? i guess i'm talking about early media but not sure |
| 16:29.42 | mechbangirc | can early media be 2-way audio??? |
| 16:30.10 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 16:33.39 | Russ | mechbangirc, I thought it was |
| 16:34.52 | mechbangirc | Russ: i want asterisk to dial a number without giving answer signal and give answer signal after say 40 seconds. is it possible? |
| 16:36.25 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 16:36.33 | *** join/#asterisk zkn (~zkn@4.234.241.83.in-addr.dgcsystems.net) |
| 16:38.38 | JunK-Y | ;;bc,stats |
| 16:38.57 | Russ | answer signal? |
| 16:39.17 | mechbangirc | i mean SIP message 200 OK |
| 16:39.34 | Russ | I don't know much about SIP, I've only used IAX early media |
| 16:40.59 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:44.03 | mechbangirc | Russ: i've been trying to do this for months but couldn't find any answer. may be with IAX i can get some hint |
| 16:46.42 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 16:50.31 | mechbangirc | i want asterisk to dial a number without giving answer signal and give answer signal after say 40 seconds. is it possible? |
| 16:55.11 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:57.05 | JunK-Y | mechbangirc: Wait(40); then Answer() |
| 16:57.56 | mechbangirc | JunK-Y: but i want the early media also |
| 17:02.25 | *** join/#asterisk karl370 (~karl@12.139.131.2) |
| 17:06.28 | karl370 | Hello, I'm new to irc, so hopefully this is the appropriate forum to post my question. |
| 17:06.47 | karl370 | I have a question about the dialplan. |
| 17:10.44 | Qwell | karl370: asking the question would be a good start |
| 17:11.16 | *** join/#asterisk sourcode (~code@ppp-58-8-98-217.revip2.asianet.co.th) |
| 17:11.47 | karl370 | Hello, I'm new to irc, so hopefully this is the appropriate forum to post my question. |
| 17:11.48 | karl370 | I'm currently calling a macro for an extension and if they don't answer, sending them to voicemail. |
| 17:11.48 | karl370 | exten => s,1,Dial(${ARG1},20) |
| 17:11.48 | karl370 | exten => s,n,Goto(s-${DIALSTATUS},1) |
| 17:11.48 | karl370 | exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@default,u) |
| 17:11.48 | karl370 | Callers would like to be able to dial another extension rather than leave a voicemail for someone. Is this possible? |
| 17:13.04 | karl370 | they can get back to the operator by dialing 0, but would like to have it work with other extensions as well |
| 17:14.05 | *** join/#asterisk koffel (koffel@173-167-208-130-ip-static.hfc.comcastbusiness.net) |
| 17:14.37 | koffel | is there any asterisk guru's in here? |
| 17:16.05 | RickB17 | koffel: whats going on? |
| 17:16.56 | jaytee | karl370, if you don't use 0 for anything else you could create an extension 0 in your dialplan so when the caller zeros out it will go to that extension. In that extension you could setup a choice for the caller to pick from to either call another extension or be transferred to the "Operator" |
| 17:19.39 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 17:20.29 | *** join/#asterisk jeffgus (~jeffgus@green.zimage.com) |
| 17:22.05 | koffel | rickb17 i need some major help ... |
| 17:22.13 | karl370 | So does that mean that when they press any keys when they get the voicemail msg, they will be taken to the 0 extension, which then will allow them dial other extensions? |
| 17:22.29 | koffel | i tring to do asterisk 1.8 manual |
| 17:22.37 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:24.03 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 17:24.40 | karl370 | jaytee: So does that mean that when they press any keys when they get the voicemail msg, they will be taken to the 0 extension, which then will allow them dial other extensions? |
| 17:24.58 | mechbangirc | how can i start sip early media so that both parties can communicate each other and then convert to regular media after specific time |
| 17:29.19 | *** join/#asterisk boch (~fseratti@200.41.228.90) |
| 17:29.23 | boch | hi all |
| 17:31.26 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 17:31.37 | boch | one question, i need to listen AMI events using javascript for queues monitoring, how is the proper way to do this? is it ok if i connect and disconnect to AMI for every ajax refresh? |
| 17:33.52 | jaytee | karl370, no. only 0 and * are available to exit voicemail back to the dialplan |
| 17:35.18 | serafie | mechbangirc: I've never done it, so I can't help you implement it, but you should research the progressinband and prematuremedia sip settings and the Progress() and Playback() with "noanswer" option dialplan applications. |
| 17:38.46 | jaytee | karl370, sorry, but I made a typo. To use the zero out option the extension you use in the current context is o (small case letter o) not a zero. |
| 17:40.33 | jaytee | karl370, you can also choose to set the exit context to another context in your dialplan by enabling the exitcontext value and setting it to the context of your choosing. |
| 17:41.09 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:44.35 | *** join/#asterisk sahX (~sahX@4.53.128.211) |
| 17:47.32 | *** join/#asterisk sahX (~sahX@4.53.128.211) |
| 17:48.55 | cj | hey folks |
| 17:49.05 | cj | is asterisknow a newer version of switchvox? |
| 17:49.50 | cj | it looks like digium-switchvox-free_8634.iso is FC 6 |
| 17:50.30 | cj | and when I search for "switchvox root password", google recommends asterisknow forum entries |
| 17:51.48 | serafie | cj: no, they are separate entities. |
| 17:52.26 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 17:52.32 | *** join/#asterisk hfb (~hfb@pool-96-247-49-75.lsanca.dsl-w.verizon.net) |
| 17:52.35 | telnettech | hello guys |
| 17:52.50 | telnettech | needing some tftp questions answered |
| 17:54.21 | telnettech | we have 3 floors and each floor has a particular IP range that is handed out to phones on that floor in the DHCP scope |
| 17:54.55 | *** join/#asterisk fofware (~Fabian@wdctf.siup.gov.ar) |
| 17:55.38 | telnettech | the DHCP points the phones to the TFTP server and there is a directory that has the config file for the phone. Is there a way to point the phones to different directories based on the ip range for each floor???? |
| 17:57.40 | Qwell | telnettech: in the dhcp server, sure. it's already doing different things for each range, right? |
| 17:58.25 | telnettech | right now it is going to the same TFTP server for ALL phones |
| 17:58.39 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:58.41 | Qwell | change it so it has a directory for the range |
| 17:59.04 | telnettech | I want it to go to the same TFTP server but when it gets there, I want it to go to a different directory for the config file based on the ip |
| 17:59.46 | telnettech | the DHCP and TFTP are on different cervers |
| 17:59.48 | telnettech | servers |
| 18:00.02 | Qwell | you aren't following |
| 18:00.13 | Qwell | The DHCP server does different things for each IP range already, yes or no? |
| 18:00.29 | telnettech | no.....It just points them all to the same TFTP server |
| 18:00.45 | Qwell | It was a yes or no question. Is that currently the case, or is it not? |
| 18:00.50 | Qwell | forget about tftp |
| 18:00.58 | Qwell | We're talking about just the DHCP server. |
| 18:01.27 | telnettech | no |
| 18:01.48 | Qwell | Then why did you say that particular ranges are given based on floor? |
| 18:02.26 | telnettech | ok so yes it decides based on the request which floor it is from and assigns an ip based on that |
| 18:02.42 | Qwell | okay |
| 18:02.50 | Qwell | and the DHCP server is using what option to set tftp? |
| 18:02.56 | Qwell | 66? 150? |
| 18:03.16 | telnettech | 66 |
| 18:09.57 | Qwell | and option 66 currently contains what, exactly? |
| 18:10.20 | telnettech | the ip of the TFTP server |
| 18:10.38 | Qwell | tftp://1.2.3.4/path/ |
| 18:11.22 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 18:12.16 | swk | telnettech, what kind of phones are you using? |
| 18:12.16 | telnettech | yes |
| 18:12.26 | telnettech | Cisco 7960 |
| 18:12.43 | telnettech | i see where your going with this Qwell |
| 18:19.29 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 18:37.50 | *** join/#asterisk fofware (~Fabian@wdctf.siup.gov.ar) |
| 18:39.56 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 18:42.12 | cj | [Apr 1 18:40:17] NOTICE[3891]: chan_sip.c:21648 handle_request_register: Registration from '<sip:4242@172.27.66.110>' failed for '172.27.252.209' - No matching peer found |
| 18:42.23 | cj | what do I need to do in order to allow registration from that address? |
| 18:44.18 | jaytee | make sure you have a sip account for 4242 |
| 18:45.06 | jaytee | in your sip.conf file |
| 18:46.05 | p3nguin | telnettech: The phones' config allows you to set the path. |
| 18:48.20 | telnettech | p3nguin: Thanks...... |
| 19:00.15 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:08.51 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 19:09.01 | *** join/#asterisk pushpop (~pushpop@pool-173-77-230-33.nycmny.fios.verizon.net) |
| 19:09.09 | pushpop | hi all does anyone have a video phone here? |
| 19:12.25 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 19:14.23 | karl370 | jaytee, we have an o ("oh") already set up in the dialplan, that goes to a specific person. I simply wanted to enable it so that a caller when reaching a voicemail mailbox doesn't have to hang up & call back again to reach a different extension or reach the operator to have them transfer the call. The caller & the operator would like not to talk with each other for that 2nd extension dialing process. I just tested pressing o when getting the voicemail and |
| 19:14.23 | karl370 | <PROTECTED> |
| 19:15.10 | Aut0ExeC | does asterisk have a built in mail server? how do voicemail get mailed out? |
| 19:15.33 | Aut0ExeC | email that is |
| 19:15.41 | benngard | therawr: is no internal mail server |
| 19:16.20 | benngard | u have to configure sendmail, exim postfix or whatever u prefer to do that |
| 19:16.21 | therawr | :) |
| 19:16.40 | Aut0ExeC | benngard: once I configure one of them is it automatic? |
| 19:16.51 | Aut0ExeC | benngard: once i include my email address in voicemail.conf |
| 19:16.53 | Aut0ExeC | ? |
| 19:17.10 | benngard | let me check in my configs |
| 19:17.27 | Aut0ExeC | k |
| 19:17.50 | benngard | nupp u have to add at least: |
| 19:17.51 | benngard | attach=yes |
| 19:17.51 | benngard | mailcmd=/usr/sbin/sendmail -t |
| 19:18.01 | Aut0ExeC | ahhh ok |
| 19:18.05 | benngard | in voicemail.conf |
| 19:18.12 | Aut0ExeC | good stuff man |
| 19:18.14 | Aut0ExeC | thanks alot |
| 19:18.15 | Aut0ExeC | :) |
| 19:18.28 | Aut0ExeC | i'm gonna prolly use a mini smtp mailer |
| 19:18.38 | Aut0ExeC | sendmail is overkill |
| 19:18.42 | Aut0ExeC | :| |
| 19:18.51 | Aut0ExeC | as well as postfix |
| 19:18.56 | Aut0ExeC | and exim |
| 19:18.56 | benngard | i use postfix, |
| 19:19.02 | Aut0ExeC | wow |
| 19:19.05 | Aut0ExeC | ok |
| 19:19.11 | benngard | sendmail is just a symlink that postfix creates |
| 19:19.32 | Aut0ExeC | yeah.. i know but there is another app.. that super small... all it does is send mail |
| 19:19.41 | Aut0ExeC | cant remember the name |
| 19:20.06 | benngard | dont know, but i know postfix does the job very well |
| 19:20.25 | Aut0ExeC | ssmtp |
| 19:20.26 | benngard | combine it with mutt and u can do almost everything |
| 19:20.32 | Aut0ExeC | yeah i know |
| 19:20.35 | Aut0ExeC | too much for me tho |
| 19:20.40 | Aut0ExeC | i use ssmtp before its nice |
| 19:20.42 | Aut0ExeC | i'll try that |
| 19:20.43 | Aut0ExeC | :) |
| 19:21.08 | benngard | i guess it will work just addopt the config line i sent u |
| 19:21.20 | Aut0ExeC | yes thank you once again |
| 19:21.35 | benngard | np, we are here for helping each other |
| 19:22.53 | benngard | is happy, playing around with q.sig info and sip |
| 19:27.06 | benngard | just saw q.sig CT inf got moved into BLINDTRANSFER, guess if i liked that? ;) |
| 19:28.17 | Aut0ExeC | benngard: last thing... do you know how to get a softphone to get voicemail notification? or any phone for that matter |
| 19:29.30 | pushpop | no video phone users = ( |
| 19:29.50 | Aut0ExeC | pushpop: nah bro |
| 19:30.03 | pushpop | ok =) |
| 19:30.41 | Aut0ExeC | at least not me sorry |
| 19:30.43 | Aut0ExeC | i'm a nub here |
| 19:30.46 | Aut0ExeC | :| |
| 19:32.08 | *** join/#asterisk Mw3 (mw3@mw3.hu) |
| 19:32.43 | benngard | Aut0ExeC: just set up a dialplan entry for voicmail, put that number in softphone |
| 19:33.03 | benngard | guess i can post my entry, hang n |
| 19:33.10 | Aut0ExeC | k |
| 19:33.48 | Aut0ExeC | benngard: I dont mean to dial voicemail... i mean after u get a voicemail to notify u |
| 19:33.52 | Aut0ExeC | red light or something |
| 19:34.25 | Aut0ExeC | some kind of indicator that you have new voicemail |
| 19:34.44 | Aut0ExeC | without dialing in i guess |
| 19:35.03 | benngard | but that what i meant |
| 19:35.16 | Aut0ExeC | ohh ok |
| 19:35.17 | benngard | u need an entry in dialplan for that |
| 19:35.21 | Aut0ExeC | oh i see |
| 19:35.33 | benngard | so your softphone can check |
| 19:35.38 | Aut0ExeC | ok |
| 19:36.06 | benngard | atleast it works for me |
| 19:36.42 | Aut0ExeC | can you post yours? |
| 19:36.46 | Aut0ExeC | pastbin or something |
| 19:36.49 | benngard | exten => 0317998990/0317998975,n,VoiceMailMain(0317998975@inputinterior.se,s) |
| 19:37.39 | benngard | number to voicemail / my extension ,s skip passwaord |
| 19:38.46 | Aut0ExeC | i see |
| 19:39.07 | Aut0ExeC | then my softphone should light up? |
| 19:39.11 | Aut0ExeC | or indicate? |
| 19:39.21 | benngard | so if u like to wake me up in some hours, just add +46 to my extension and my cell phone will ring :) |
| 19:39.21 | Aut0ExeC | xlite |
| 19:39.33 | Aut0ExeC | hahah |
| 19:39.39 | Aut0ExeC | ok |
| 19:40.06 | benngard | if i remember correctly yes |
| 19:40.18 | Aut0ExeC | ok thanks man... again... good info |
| 19:41.27 | *** join/#asterisk fullstop (~fullstop@64-121-54-104.c3-0.tlg-ubr1.atw-tlg.pa.cable.rcn.com) |
| 19:41.58 | fullstop | Hi. I'm going through some testing with AT&T so that we're allowed to use Asterisk with one of their services... |
| 19:42.50 | benngard | http://www.google.com/tisp/install.html <- just for fun, its friday! ;) |
| 19:42.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:43.09 | fullstop | and.. they want to know what asterisk will return if it does not have the resources to accept the call. Now, we've always sized our hardware such that this issue has never been encountered.. |
| 19:43.35 | fullstop | Any idea what asterisk will do? What SIP message it will return when it can't accept the call? |
| 19:47.12 | *** join/#asterisk kaarlo (~weissbier@unaffiliated/kaarlo/x-8749308) |
| 19:47.18 | kaarlo | hi |
| 19:48.16 | kaldemar | fullstop: depends on the reason, forbidden, declined and so on with a cause code that you can set yourself too. |
| 19:49.01 | kaarlo | when i try to load dahdi i get |
| 19:49.03 | kaarlo | [Apr 1 21:48:03] ERROR[8162]: chan_dahdi.c:10163 mkintf: Unable to open channel 1: No such device or address |
| 19:49.03 | kaarlo | here = 0, tmp->channel = 1, channel = 1 |
| 19:49.12 | kaarlo | whats wrong with it? |
| 19:50.54 | kaldemar | no channel 1 configured in dahdi itself. |
| 19:50.59 | *** join/#asterisk syadnom (~syadnom@216.187.176.106) |
| 19:51.51 | syadnom | hi all. looking for some help with a potential dropped call issue. asterisk 1.4.26.1. I need some help identifying a dropped call from my logs. |
| 19:52.24 | Aut0ExeC | syadnom: u like 1.8? just curious |
| 19:52.40 | syadnom | Aut0ExeC, like 1.8?? |
| 19:52.44 | Aut0ExeC | version |
| 19:52.52 | syadnom | Aut0ExeC, never run it |
| 19:52.53 | kaarlo | hmm, okay, now i get another error message |
| 19:52.56 | kaarlo | [Apr 1 21:52:35] ERROR[8346]: chan_dahdi.c:10177 mkintf: Signalling requested on channel 1 is FXS Kewlstart but line is in FXO Kewlstart signalling |
| 19:53.07 | kaarlo | but it _is_ a fxs card |
| 19:53.45 | kaldemar | kaarlo: fxs uses fxo signalling and vice versa |
| 19:53.57 | kaarlo | oh, okay :S |
| 19:54.06 | kaarlo | nvm |
| 19:54.27 | syadnom | does this mean anything to you guys? "Polarity Reversal event occured - DEBUG 2: channel 1, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -20918912" |
| 19:56.39 | kaldemar | syadnom: polarity reversal can be used to indicate hangup. if you dont want it, there is hanguponpolarityswitch or something like that in chan_dahdi.conf. |
| 19:57.19 | ectospasm | in some locations it can also be used to indicate a new call |
| 19:57.28 | kaarlo | kaldemar, if i pick up the phone now, i should here a dialtone, right? |
| 19:58.31 | syadnom | I don't have hanguponpolarityswitch enabled. My logs say Ignore switch to REVERSED Polarity. Im try to identify dropped calls |
| 19:59.23 | ectospasm | syadnom: so set hanguponpolarityswitch=no |
| 19:59.30 | ectospasm | to make it explicit. |
| 20:00.07 | kaldemar | kaarlo: if everything is correct |
| 20:00.26 | syadnom | ectospasm, is that just to clear the log entry? |
| 20:01.13 | ectospasm | you're getting a polarity reversal (polarity switch), do they coincide with dropped calls? |
| 20:01.24 | syadnom | ectospasm, I cant tell |
| 20:01.49 | syadnom | ectospasm, I am 1000 miles away and the staff on site is impossible to talk to |
| 20:02.08 | syadnom | ectospasm, meaning (retarded), not technical issue |
| 20:02.11 | ectospasm | syadnom: you don't have ssh access to the site? |
| 20:02.17 | syadnom | ectospasm, I do |
| 20:02.36 | syadnom | ectospasm, but I cant reproduce the dropped call issue that is being complained about |
| 20:03.05 | ectospasm | what kind of card is this? |
| 20:03.07 | syadnom | ectospasm, and the on-site people seem to be incapable of giving and even remoutely accurate time of when the hypothetical dropped calls happened. |
| 20:03.14 | syadnom | ectospasm, Sangoma B600DE |
| 20:03.24 | syadnom | ectospasm, wait, this site is A200 |
| 20:03.31 | ectospasm | I don't know sangoma cards |
| 20:05.32 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:09.24 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 20:12.11 | *** join/#asterisk ccccp123 (~IceChat77@cpc9-glfd5-2-0-cust157.6-2.cable.virginmedia.com) |
| 20:14.39 | *** part/#asterisk ccccp123 (~IceChat77@cpc9-glfd5-2-0-cust157.6-2.cable.virginmedia.com) |
| 20:14.44 | *** join/#asterisk ccccp123 (~IceChat77@cpc9-glfd5-2-0-cust157.6-2.cable.virginmedia.com) |
| 20:16.19 | syadnom | "Dunno what to do with event 0" ?? |
| 20:20.44 | kaarlo | hmm, kaldemar i get strange tones, like a numbersstation from a secret service |
| 20:23.39 | fullstop | we're on to you, kaarlo |
| 20:25.50 | *** join/#asterisk zkn (~zkn@82.131.93.186.cable.starman.ee) |
| 20:27.45 | *** join/#asterisk Greek-Boy (~Greek-Boy@41.191.92.13) |
| 20:27.59 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 20:32.19 | *** join/#asterisk Cadey (~x@host217-43-34-101.range217-43.btcentralplus.com) |
| 20:41.02 | *** join/#asterisk kaarlo (~weissbier@unaffiliated/kaarlo/x-8749308) |
| 20:41.05 | kaarlo | oops |
| 20:41.15 | kaarlo | <PROTECTED> |
| 20:41.15 | kaarlo | Write failed: Broken pipe |
| 20:41.16 | kaarlo | :S |
| 20:51.02 | *** join/#asterisk mawhiii (~trav@tcmsdev.com) |
| 21:02.12 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 21:08.12 | *** part/#asterisk killown (~killown@unaffiliated/killown) |
| 21:10.25 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 21:17.03 | *** join/#asterisk sahX (~sahX@99-105-56-250.lightspeed.sntcca.sbcglobal.net) |
| 21:19.02 | p3nguin_ | benngard: You mentioned that to send voicemail notification, sendmail, exim, or postfix would need to be used. Just a note: I've switched to an embedded appliance for Asterisk, and I currently use gmail as my MTA and use msmtp to send the mails off the box. It's much lighter/better/easier than running your own smtp daemon all the time. |
| 21:19.08 | kaarlo | is pulse dialing with dahdi possible? |
| 21:19.49 | kaarlo | if its possible i'll get a rotary dial phone and attach it to my asterisk :3 |
| 21:21.27 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 21:25.20 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 21:31.10 | *** join/#asterisk ccesario (~ccesario@201-95-147-176.dsl.telesp.net.br) |
| 21:38.09 | *** join/#asterisk peplin (~peplin@c-71-199-115-206.hsd1.pa.comcast.net) |
| 21:40.07 | peplin | Hello - I'm stuck setting up Asterisk for outgoing Google Voice calls. The outgoing calls start up, but end because of "congestion" with this in the logs: "Channel allocation failed: Can't create alert pipe!". Any ideas? |
| 21:44.50 | peplin | I understand increasing the file descriptor limit if I had a very busy server, but this is just a single call |
| 21:46.02 | peplin | ls /proc/<fd of asterisk>/fd/ seems to fill up rapidly when I place a call |
| 21:49.36 | *** part/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 21:49.39 | *** join/#asterisk fofware (~Fabian@wdctf.siup.gov.ar) |
| 21:54.04 | *** join/#asterisk csnook_laptop (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 22:12.27 | peplin | nevermind - I think I had some circular extensions that were causing way too many pipes to be opened |
| 22:12.34 | peplin | trimmed down the config and it's working now |
| 22:16.01 | *** join/#asterisk wonderworld (~ww@port-92-201-26-77.dynamic.qsc.de) |
| 22:30.01 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 22:32.35 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:47.52 | *** join/#asterisk wonderworld (~ww@port-92-201-26-77.dynamic.qsc.de) |
| 22:47.52 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:52.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:53.31 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 22:58.02 | *** join/#asterisk ariel_ (~abatista@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 23:09.01 | *** join/#asterisk guilhermebr (~Guilherme@189.63.54.47) |
| 23:22.18 | *** join/#asterisk jetlag (jetlag@pool-173-61-216-136.cmdnnj.east.verizon.net) |
| 23:26.03 | *** part/#asterisk guilhermebr (~Guilherme@189.63.54.47) |
| 23:32.08 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 23:34.00 | *** join/#asterisk ANurmi (~Adam@63.230.70.254) |
| 23:34.50 | ANurmi | I am having a voicemail issue and I googled the symptoms but can not find a solution. |
| 23:40.07 | ANurmi | it is happening intermittenly on a couple of extensions, the voice mail indicator will go off, and the user will call the voice mail extension and be told there is a new message, but attempting to play it will instanting go to "press 3 for advanced options. |
| 23:40.35 | ANurmi | and in the console it will display "WARNING[27740]: app_voicemail.c:7200 play_message: No message attribute file?!! (/var/spool/asterisk/voicemail/default/6001/INBOX/msg0001.txt)" |
| 23:41.36 | ANurmi | is there a database that the system polls that says there is still this message that was most likely deleted by the user long ago? |
| 23:53.49 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 23:53.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 23:56.34 | p3nguin | anurmi: Check the directory to see what files exist in it. |
| 23:57.37 | p3nguin | anurmi: E.g., if msg0000.WAV exists, but msg0000.txt does not, that's a problem. |
| 23:58.00 | p3nguin | or in your case, it seems it's msg0001 that's the issue. |
| 23:58.05 | ANurmi | well looking at the folder the msg0001.txt doesn't exist |
| 23:58.16 | ANurmi | and there is no msg0001.wav or .gsm |
| 23:58.30 | p3nguin | Is there any msg0001.* ? |
| 23:58.35 | ANurmi | no |
| 23:58.36 | p3nguin | any msg0001 file at all? |
| 23:59.46 | p3nguin | My first thought was that msg0001.gsm (or whatever format you are writing voicemails in) exists but had almost no length. |
| 23:59.57 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |