| 00:00.03 | l1nuxman | when I look at commmand line it shows the callerid but in the body of email it only says from "FXOPort" . Why? |
| 00:00.32 | p3nguin | Show me your email command. |
| 00:05.52 | l1nuxman | p3nguin, mailcmd=/usr/sbin/sendmail -f sda@rogas.com -t |
| 00:06.11 | p3nguin | Where did you find that? |
| 00:06.30 | l1nuxman | I wrote it in voicemail.conf |
| 00:06.43 | p3nguin | You didn't need to. |
| 00:07.12 | p3nguin | Unless you're not using a sendmail compatible MTA, you can leave the mailcmd line commented out. |
| 00:07.19 | p3nguin | Put your email address on your voicemail entry down below. |
| 00:08.07 | l1nuxman | I have to have that p3nguin because I'm using SmartHost |
| 00:08.09 | p3nguin | The content of the email is configured in emailbody. |
| 00:08.17 | p3nguin | I doubt it. |
| 00:08.22 | p3nguin | But suit yourself. |
| 00:08.45 | l1nuxman | yea p3nguin in emailbody it has VM_CALLERID |
| 00:09.08 | l1nuxman | but it writes FXOPOrt in email instead of callerid like in command line |
| 00:09.31 | p3nguin | FXOPort must be the name of the device that left you voicemail. |
| 00:09.45 | WIMPy | callerid name? |
| 00:10.09 | p3nguin | I don't remember what VM_CALLERID is supposed to contain. |
| 00:10.20 | p3nguin | I'd have to go look at one of my emails from getting voicemail. |
| 00:10.40 | p3nguin | Yes, I meant callerid name. |
| 00:10.56 | l1nuxman | - Executing [grandstream@in-pstn:2] Set("SIP/phoneline-fxo-00000001", "CALLERID(number)=ASZ DRAG" <sip:4165566740@192.168.1.115>") in new stack |
| 00:11.17 | p3nguin | ASZ DRAG is not a valid CALLERID(num). |
| 00:11.20 | l1nuxman | but not in email |
| 00:11.29 | WIMPy | Why not? |
| 00:11.45 | p3nguin | Mostly because it's not a number. |
| 00:12.04 | WIMPy | Who said that the number may only contain digits? |
| 00:14.01 | p3nguin | I'm sure someone with authority said it. I don't have all the rules in writing, so I can't give you a name. |
| 00:14.27 | WIMPy | That hasn't even been true in the PSTN. |
| 00:14.59 | WIMPy | Just that every known telco happens to filter any non-digit, but it's a ASCII string. |
| 00:15.01 | p3nguin | Since some carriers will reject or drop caller id numbers containing letters, it's never a good idea to even try it. |
| 00:15.09 | WIMPy | Or probably rather IA5 or something. |
| 00:16.32 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 00:16.48 | l1nuxman | yea I get callerid(all) like this sip:4165566740@1921681115 |
| 00:17.21 | WIMPy | sipgate use customer numbers as caller id if not configured. And if you have multiple accounts the callerid consists of customer+"e"+subaccount. I don't see anything wrong with that. |
| 00:22.21 | p3nguin | So ${CALLERID(all)} shows a SIP URI? |
| 00:42.18 | l1nuxman | wow yea |
| 00:42.40 | l1nuxman | well I want to take off the 'sip:' and '@xxxxxxxxx' part |
| 00:43.21 | p3nguin | That's not a typical caller id. I'd try to find out why that's what is showing up instead of the normal Name <Number>. |
| 00:43.39 | l1nuxman | its my IP address |
| 00:53.10 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 00:55.38 | *** join/#asterisk moy_ (~moy@69.158.118.39) |
| 00:57.38 | *** join/#asterisk talntid (~talntid@c-98-247-3-28.hsd1.wa.comcast.net) |
| 01:08.10 | *** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com) |
| 01:10.22 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:19.13 | *** join/#asterisk greenwolf (~chatzilla@cpe-74-77-221-5.buffalo.res.rr.com) |
| 01:19.54 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 01:21.00 | leftist | when setting up the phone for the user does each user have to have their own designated phone #? or can they share the same one? |
| 01:21.18 | leftist | or am i thinking backwards? |
| 01:21.24 | leftist | i knnow what i am thinking anyway |
| 01:21.29 | *** join/#asterisk coppice (~chatzilla@m121-202-64-200.smartone-vodafone.com) |
| 01:21.39 | p3nguin | One extension can dial multiple phones. |
| 01:22.08 | leftist | ok |
| 01:22.55 | leftist | well p3nquin i am using goautodial for a call center and i was trying to avoid creating all the phones for each individual agent however i think i may have to. i am not sure to be honest. |
| 01:23.15 | leftist | it's all outbound at this facility |
| 01:23.23 | p3nguin | Every single phone will have its own entry in sip.conf (if every phone is a SIP phone). |
| 01:23.35 | p3nguin | But no phones are required to have extensions to be able to call them. |
| 01:23.37 | leftist | ok |
| 01:23.43 | leftist | ahh i see |
| 01:24.01 | p3nguin | You could have one single extension to call 100 phones if you wanted. |
| 01:24.18 | leftist | ahh i see |
| 01:24.21 | p3nguin | But each of the 100 phones will have a peer entry in sip.conf. |
| 01:24.29 | leftist | ok |
| 01:24.35 | leftist | let me look at this config |
| 01:24.48 | p3nguin | Of course for 100 phones, using a database might make more sense. |
| 01:25.08 | leftist | yes |
| 01:28.09 | leftist | asterisk is truely remarkeble. i remember when i had to work with rolm cbx's without a manual :D. boy was that a trip |
| 01:40.30 | leftist | when creating a new phone and lets say i am creating for a test 10. can i use the same park extension and conf extension for the phones? just to do some testing? or is that not practical? |
| 01:41.03 | p3nguin | Like using extension 700 for parking and for a conference? |
| 01:41.28 | leftist | no say 700 for parking and 799 for conf for example |
| 01:41.49 | p3nguin | How is that considered the same when it's different? |
| 01:41.50 | leftist | for each phone i create using those 2 values as default |
| 01:42.27 | p3nguin | You probably won't be creating extensions for every single phone to be able to call those things. You'll have one instance of those extensions and all the phones will be able to call the numbers. |
| 01:42.30 | leftist | so each phone can use 700 and 799? i mean i have not done this type stuff in over 25 years i am way ignorant. |
| 01:42.41 | leftist | oh i see |
| 01:43.08 | p3nguin | If every phone has a context of 'phones', then phones might have extension 799 in it. Then all phones can call 799. |
| 01:43.43 | leftist | ok |
| 01:44.26 | leftist | i am feeling my way thru this thru trial and error however i am taking my time and going slow. it works now but i am just beinng causious at this point. |
| 01:44.34 | leftist | spelling is bad at this hour |
| 01:44.37 | p3nguin | ~book |
| 01:44.37 | 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 |
| 01:44.48 | p3nguin | Take a break; read a book. |
| 01:46.02 | leftist | ok thanks lol |
| 01:46.56 | l1nuxman | this is incorrect syntax? |
| 01:46.59 | l1nuxman | same => n,Set(CALLERID(num)=${CUT(CUT((SIP_HEADER(P-Asserted-Identity):4),@,1),:,2)}) |
| 01:47.24 | p3nguin | Yes. |
| 01:47.49 | p3nguin | ${CUT(${SIP_HEADER(stuff)})} |
| 02:00.48 | l1nuxman | are you sure |
| 02:00.50 | l1nuxman | same => n,Set(CALLERID(num)=${CUT(${CUT(${SIP_HEADER(P-Asserted-Identity):4},@,1)},:,2)}) |
| 02:01.11 | leftist | can multiple agents have the same phone number? every time i try to add a new phone i am basically cloning the 1st one i created yet it doesnt accept the values i use. it could be that the values have to be different for conf/vmx i cant get the book till tomorrow so i am just asking it for some thought. |
| 02:04.44 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 02:11.18 | p3nguin | Syntax looks good, but I can't guarantee you'll get the result you want. |
| 02:11.54 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 02:12.32 | l1nuxman | says I have a missing argument |
| 02:15.18 | dijib | linuxman do you know linux? |
| 02:16.30 | l1nuxman | yea |
| 02:17.00 | dijib | familiar with linking .so's? |
| 02:17.00 | l1nuxman | I can't find a missing argument |
| 02:17.10 | dijib | let me see your dialplan |
| 02:17.34 | dijib | if its macro or c i probably cant help |
| 02:18.04 | l1nuxman | http://pastebin.com/BPrj5J5a |
| 02:18.31 | p3nguin | I'd still be concentrating on why callerid is jacked up instead of worrying about hacking the broken callerid. |
| 02:18.54 | dijib | im having issues setting my cid aswell |
| 02:19.00 | p3nguin | Did you consider setting the callerid in the peer entry? |
| 02:19.07 | dijib | did that already |
| 02:19.25 | p3nguin | What's the problem with yours? I know what's wrong with the other guy's. |
| 02:19.29 | WIMPy | I should save that one for the ITSPs. |
| 02:19.30 | dijib | but when you make a call does it rewrite that variable? |
| 02:19.49 | p3nguin | What variable? |
| 02:19.59 | WIMPy | Wouldn't be bad if the P-asserted-identity stuff was used by default. |
| 02:20.13 | dijib | i have a missing dependency i cant find, libcurl.so.3 linked it to the 4.1.1 i have install. didnt work |
| 02:20.43 | p3nguin | What is asking for libcurl? |
| 02:21.26 | dijib | for lumenvox |
| 02:21.42 | dijib | and does that <tab> same thing work? |
| 02:21.52 | p3nguin | tab same? |
| 02:21.59 | dijib | in his dialplan |
| 02:22.15 | p3nguin | I don't know if tabs work, but same does. I didn't look at his paste. |
| 02:22.17 | dijib | and he doesnt have the extention defined |
| 02:22.24 | dijib | same does? |
| 02:22.36 | dijib | hows it work? same => same |
| 02:22.37 | p3nguin | He has an extension defined. |
| 02:22.45 | dijib | grandstream? |
| 02:22.49 | p3nguin | yes. |
| 02:22.51 | dijib | for all actions? |
| 02:23.03 | p3nguin | It works exactly as he wrote it, sans the tabs. |
| 02:23.14 | p3nguin | If the tabs don't break it, then tabs work, too. |
| 02:23.18 | WIMPy | tabs are ok |
| 02:23.31 | p3nguin | Good to know. |
| 02:23.57 | dijib | tabs to wort it out, but it works anyways |
| 02:23.57 | WIMPy | (at least in practice) |
| 02:23.58 | p3nguin | same => was implemented in 1.6.2 I believe. |
| 02:24.12 | dijib | im running 1.8.5 |
| 02:24.17 | dijib | so i should be good. |
| 02:24.26 | p3nguin | same => works for you, too. |
| 02:24.35 | p3nguin | You just probably don't use it. |
| 02:25.29 | dijib | so would yum check version of the libcurl library or would it have missing functions or something in 4.1.1 that were in 3? |
| 02:26.46 | p3nguin | Update it and see what happens. |
| 02:27.52 | dijib | its up to date |
| 02:28.16 | p3nguin | rebuild that other app against your new version. |
| 02:28.31 | dijib | i had to make a softlink from libcurl.so.3 libcurl.so.4.1.1 |
| 02:33.36 | dijib | i think i need a svn truck of the lumenvox software |
| 02:33.40 | dijib | trunk |
| 02:33.43 | dijib | truck lol |
| 02:33.47 | dijib | chevy on my mind |
| 02:41.20 | dijib | can i run multiple commands in a system command? |
| 02:41.37 | p3nguin | I don't see why not. |
| 02:42.52 | dijib | so like killall asterisk && /usr/sbin/asterisk |
| 02:43.29 | p3nguin | I'm not sure if you kill asterisk if System() will still execute the rest of the command. |
| 02:43.45 | dijib | should. |
| 02:43.55 | p3nguin | Try it. |
| 02:44.07 | dijib | and it would be a killall asterisk ; /usb/sbin/asterisk |
| 02:44.19 | p3nguin | Why wouldn't && work? |
| 02:44.28 | dijib | if killall has an error |
| 02:44.33 | dijib | asterisk wont be run |
| 02:44.38 | p3nguin | If it had an error, you wouldn't need to run asterisk. |
| 02:44.41 | dijib | if it does with ; it still will |
| 02:44.45 | p3nguin | (it would be running already) |
| 02:44.59 | dijib | yeah but it would be a kill and reload |
| 02:45.20 | p3nguin | Running asterisk when it is already running is not going to "reload" as you put it. |
| 02:45.52 | dijib | ok how do i reinitiate moh? my streams die on me sometimes. |
| 02:46.20 | p3nguin | Maybe System(pkill asterisk && /usr/sbin/asterisk || asterisk -rx 'core restart now'); would be better. |
| 02:46.42 | p3nguin | module unload res_musiconhold.so ; module load res_musiconhold.so |
| 02:47.25 | dijib | how do i run commands in asterisk? |
| 02:47.39 | p3nguin | type them, press enter. |
| 02:48.01 | dijib | how do i do it offsite over asterisk? |
| 02:48.13 | dijib | i mean online in asterisk |
| 02:48.33 | p3nguin | Did you already connect to the CLI? |
| 02:48.43 | dijib | no assume i cannot connect to cli |
| 02:48.46 | p3nguin | asterisk -r |
| 02:48.53 | dijib | no ssh or telnet |
| 02:48.54 | Nugget | telnet is eeeeeeevil! |
| 02:49.03 | p3nguin | If you can't connect to the CLI, you can't run commands. |
| 02:49.06 | dijib | Nugant? |
| 02:49.10 | p3nguin | Ted? |
| 02:49.16 | dijib | i can run system commands |
| 02:49.30 | p3nguin | If you can't connect to the CLI, you can't run commands. |
| 02:49.45 | p3nguin | If you can't connect to it with asterisk -r, then asterisk -rx will not run commands either. |
| 02:49.48 | dijib | oh know what your right. |
| 02:49.56 | p3nguin | Imagine that. |
| 02:50.08 | p3nguin | My right. |
| 02:50.11 | p3nguin | And my left, too. |
| 02:50.21 | dijib | and up and down |
| 02:57.27 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 03:03.34 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 03:08.40 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 03:17.56 | *** join/#asterisk atan2 (~atan@unaffiliated/atan) |
| 03:54.22 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 03:56.51 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 04:22.08 | *** join/#asterisk moy_ (~moy@74.198.87.35) |
| 04:29.08 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 04:44.16 | p3nguin | Well this is new in 1.8.7.0, and I am not a fan: WARNING[24334]: res_jabber.c:1473 aji_send_raw: JABBER: Unable to send message to asterisk, we are not connected[Sep 25 23:43:46] WARNING[24334]: res_jabber.c:1737 aji_act_hook: Jabber didn't seem to handshake, failed to authenticate. |
| 04:44.37 | p3nguin | So back to 1.8.6.0 I go. |
| 04:50.55 | *** join/#asterisk paec (~user@AAmiens-551-1-68-36.w92-142.abo.wanadoo.fr) |
| 04:51.15 | dym | mhh |
| 04:51.23 | dym | ive continuously had a strange jabber error too |
| 04:51.48 | dym | but in 1.8.5.0 |
| 04:51.56 | dym | oughtta upgrade i gutess |
| 04:55.48 | p3nguin | Every 55 seconds, that warning would show up. |
| 04:56.29 | p3nguin | System uptime: 122 |
| 04:56.50 | p3nguin | Hasn't shown up once now that I'm back on 1.8.6.0. |
| 04:57.17 | p3nguin | Just to be certain it's the version, I'll go back to 1.8.7.0 again. |
| 04:59.19 | drmessano | Jabber is fine here on 1.8.7... I am still getting some XML errors, but I have had those since 1.8.0 |
| 04:59.33 | p3nguin | The one saying invalid XML? |
| 04:59.40 | p3nguin | failure to parse |
| 04:59.43 | drmessano | yeah |
| 04:59.44 | p3nguin | some crap like that? |
| 04:59.57 | p3nguin | It seems to parse it just fine. |
| 05:00.58 | drmessano | From what I can gather, doing the "Read 10 google responses on the issue, average them out, separate the stupid appliance operator errors out" sort of reckoning, it appears to be a unicode issue with iksemel |
| 05:01.04 | *** join/#asterisk mickecarlsson (~Micke@h10n3c1o1101.bredband.skanova.com) |
| 05:01.30 | p3nguin | 54-55 seconds on 1.8.7.0, that warning pops up. |
| 05:01.38 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:01.46 | p3nguin | I rolled back to 1.8.6.0, it's gone. Back up to 1.8.7.0, it's back. |
| 05:01.56 | drmessano | Hmmm |
| 05:02.24 | p3nguin | [Sep 26 00:02:04] WARNING[25183]: res_jabber.c:1473 aji_send_raw: JABBER: Unable to send message to asterisk, we are not connected[Sep 26 00:02:05] WARNING[25183]: res_jabber.c:1737 aji_act_hook: Jabber didn't seem to handshake, failed to authenticate. |
| 05:02.36 | p3nguin | Not connected to what?! |
| 05:02.58 | drmessano | I am not seeing that here.. 2 Google accounts and 2 accounts connected to an Ejabberd server here. Anything unique on your end? |
| 05:03.01 | p3nguin | jabber show connections says that all of them *ARE* connected. |
| 05:03.06 | drmessano | Hmm |
| 05:03.44 | p3nguin | I have four on GTalk and one jabber component. |
| 05:04.14 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 05:04.30 | p3nguin | During the load up of 1.8.6.0 I see that warning from res_jabber. |
| 05:04.33 | p3nguin | Once. |
| 05:04.34 | drmessano | jabber component? Like a devstate connection or something? |
| 05:04.52 | p3nguin | Rather than asterisk being a client, it is a component. |
| 05:04.57 | drmessano | ok |
| 05:05.39 | p3nguin | Oops, I thought it was doing it on 1.8.6.0... but I forgot to change the package back. :/ |
| 05:06.41 | drmessano | What is a case usage of connecting as a component? |
| 05:06.45 | p3nguin | Okay with 1.8.6.0, I don't see the res_jabber message during loading. |
| 05:08.11 | p3nguin | I just use my jabber to give users IMs with call information. It could still do that as a client, but I had issues with TLS stuff. |
| 05:08.20 | drmessano | Ah ok |
| 05:08.47 | p3nguin | It makes asterisk more like a peered jabber server. |
| 05:08.57 | drmessano | Well, I am doing the same with one of my connections. Let me switch that over to a component connection and see if I get the errors |
| 05:09.12 | atan2 | Who has the best rate in inbound DIDs? |
| 05:09.22 | p3nguin | I guess I could drop that for a minute and just leave the gtalk stuff active. |
| 05:12.32 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 05:15.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 05:16.40 | p3nguin | I disabled the component config for my jabberd, and the warning is not appearing after 137 seconds. |
| 05:24.43 | drmessano | Interesting |
| 05:25.12 | p3nguin | So something changed to where now asterisk does not like being a component with my jabberd. |
| 05:26.20 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 05:26.54 | p3nguin | It doesn't work with tls nor sasl. Jabber didn't seem to handshake... so what could it be? |
| 05:27.20 | *** join/#asterisk talntid (~talntid@c-98-247-3-28.hsd1.wa.comcast.net) |
| 05:33.27 | *** join/#asterisk freeman_u (~freeman@193.110.114.54) |
| 05:36.10 | p3nguin | Hmm, that really sucks. |
| 05:36.31 | p3nguin | I don't want to be locked in at 1.8.6.0 for the rest of eternity. |
| 05:48.02 | drmessano | I can't get it to connect at all |
| 05:48.10 | drmessano | As a component |
| 05:48.12 | drmessano | 1.8.7 |
| 05:48.30 | p3nguin | Does it give you that same warning that I get? |
| 05:48.44 | drmessano | Lemme check.. only got as far as it never connecting |
| 05:48.52 | p3nguin | It tells me that crap and the status is always "Connecting." |
| 05:49.26 | drmessano | Jabber didn't seem to handshake, failed to authenticate. |
| 05:49.30 | drmessano | Yepper |
| 05:49.36 | drmessano | "Shits broke" |
| 05:49.42 | *** join/#asterisk Takapa (vegard@svanberg.no) |
| 05:49.58 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 05:49.58 | drmessano | I guess we can fram 1.8.7 right in the dooker |
| 05:50.12 | p3nguin | I tried without tls and sasl, with one but not the other, with the other but not one, and with both. Same message no matter what combination. |
| 05:50.48 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 05:50.55 | p3nguin | Back on 1.8.6.0, any combination of tls and sasl gives me no issues. |
| 05:51.51 | p3nguin | But, on an unrelated issue, I do have a problem with musiconhold never playing mp3s unless I manually unload and load res_musiconhold after asterisk is up. |
| 05:52.07 | p3nguin | I even preload format_mp3.so in modules.conf. |
| 05:52.08 | drmessano | nice |
| 05:52.44 | drmessano | MoH has been a problem for some time, in one way or another.. Streaming MoH hasn't worked at all in 1.8 with the dahdi timer loaded |
| 05:52.48 | p3nguin | If I restart asterisk and forget to fix the module, I might play silence. |
| 05:53.16 | p3nguin | As long as I unload and load the module so mp3 files will play, I can play a stream. |
| 05:54.01 | drmessano | hmm |
| 05:54.43 | p3nguin | preload loads the module early... is there any way to wait until later to load a module? |
| 05:54.48 | pabelanger | p3nguin: http://svnview.digium.com/svn/asterisk?view=revision&revision=333265 |
| 05:55.09 | p3nguin | I think if I can get res_musiconhold to wait until last it may work. |
| 05:56.08 | p3nguin | That revision seems to be unrelated. I'm not doing any devstate stuff with jabber. |
| 05:56.19 | p3nguin | And Asterisk isn't segfaulting. |
| 05:56.27 | *** join/#asterisk bluregard (~mattbrei@c-98-228-3-34.hsd1.il.comcast.net) |
| 05:56.35 | bluregard | greetings |
| 05:57.11 | drmessano | Same here |
| 05:57.53 | pabelanger | p3nguin: that is the patch that introduces the WARNING message. Something to start with |
| 05:58.22 | p3nguin | If that patch puts it in, you're thinking I could unpatch it? |
| 05:58.54 | p3nguin | I guess I really need to file a bug on this moh problem. |
| 05:59.10 | pabelanger | If you want, though I have no idea what you are doing or what the patch does. I just found the source of the WARNING |
| 05:59.17 | pabelanger | & |
| 05:59.22 | bluregard | anyone out there that might be able to point me in the right direction? I need to create an auto-dialer that can grab numbers out of a mysql database, place a call and play a pre-recorded message. I'm stuck on how to actually initiate the call. Callfile or originate... |
| 05:59.54 | p3nguin | I originate with a shell script. |
| 06:01.14 | bluregard | that's what I've been leaning towards. My other problem is I'm not sure how to handle making multiple simultanious calls via a SIP channel. |
| 06:01.44 | p3nguin | I use a basic for loop in a shell script. |
| 06:01.53 | drmessano | I don't think that patch is related.. I think the warning is being logged because we are indeed not connected.. which is a result of the real issue |
| 06:02.28 | drmessano | I'm going back a bit |
| 06:02.45 | kaldemar | bluregard: by SIP channel you probably mean a device or a peer. you don't have to handle it in any way, just dial. |
| 06:03.26 | p3nguin | When I used asterisk 1.4, I could originate as many calls as I wanted right in a row and the calls would be concurrent. Using the same script on 1.8, they would block and would run the calls consecutively. The solution for that was to background the originates. Now I can call as many concurrent as I want again. |
| 06:04.19 | bluregard | I mean peer. By handle I mean limit the number of calls that are going out. I need to keep the volume at a rough hourly rate. |
| 06:05.12 | kaldemar | bluregard: do it in the script that originates the calls. |
| 06:05.35 | p3nguin | By looping through a list of numbers, I had to implement a marker in the list where to wait so the existing calls could die off. |
| 06:05.55 | p3nguin | Otherwise it would call every number in the list at once. |
| 06:06.04 | bluregard | yeah. if I let it I'd be pushing out >1000 calls at a time which would then lead to an angry call from my sip provider. |
| 06:06.12 | p3nguin | I wanted to keep it limited to 15-ish. |
| 06:06.56 | p3nguin | Do you even have enough bandwidth for 1000 calls at once? |
| 06:07.20 | bluregard | probably not |
| 06:07.45 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:07.46 | schmidts | good morning |
| 06:08.22 | drmessano | HA |
| 06:08.24 | p3nguin | Maybe I should stick with 1.8.6.0 and find the patches to fix the voicemail times. |
| 06:08.30 | drmessano | well ok then |
| 06:08.30 | bluregard | although I might end up putting the asterisk server in my provider's datacenter, we'll see. |
| 06:09.31 | drmessano | p3nguin: I reverted that patch and it went away.. |
| 06:09.41 | p3nguin | Interesting. |
| 06:09.44 | drmessano | Yeah |
| 06:09.53 | bluregard | I like the marker in the list of numbers idea though. I guess I could pepper those throughout the database. |
| 06:10.12 | p3nguin | You could even implement a counter. |
| 06:10.41 | bluregard | that's what I was orignially thinking |
| 06:10.42 | p3nguin | I just did the marker because it was an easy fix without changing the script too much. |
| 06:11.01 | p3nguin | But now that I think about it, I'll probably add a counter. |
| 06:11.19 | p3nguin | I'll let it count N amount of numbers from the list, then wait, then restart. |
| 06:11.46 | bluregard | yeah |
| 06:13.38 | *** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105) |
| 06:20.32 | bluregard | is your list of numbers a text file or are you using a db? |
| 06:20.52 | p3nguin | I'm just doing a text file right now, but I'm hoping to move to a db pretty soon. |
| 06:22.11 | bluregard | I have to be able to add information to the list of numbers, like call start/stop time, number of retries, reason for retry, etc. So I might as well build it with a db from the start. |
| 06:25.04 | p3nguin | I'm having trouble getting that patch to revert. |
| 06:25.29 | drmessano | Go to line 1468 |
| 06:25.44 | p3nguin | patch -Rp2 < patchfile |
| 06:25.48 | p3nguin | no worky |
| 06:25.55 | drmessano | I did it by hand |
| 06:25.58 | drmessano | Worked fine |
| 06:26.05 | p3nguin | I really need to do it automatic. |
| 06:26.11 | drmessano | ok |
| 06:28.59 | p3nguin | patching file res/res_jabber.c |
| 06:28.59 | p3nguin | Hunk #1 FAILED at 1465. |
| 06:29.13 | p3nguin | I had this same trouble with chan_sccp a while ago. |
| 06:29.21 | drmessano | Why don't you just fix the file and make your own patch |
| 06:29.24 | p3nguin | I had to recreate the patch locally. |
| 06:29.27 | p3nguin | Exactly. |
| 06:29.38 | p3nguin | I think it's a problem with copy/paste or something. |
| 06:35.45 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 06:38.40 | p3nguin | I guess my unpatch works -- no error. |
| 06:41.26 | atan | p3nguin, are you messing with 10.x? |
| 06:42.18 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 06:42.26 | p3nguin | no |
| 06:46.26 | *** join/#asterisk reber (~reber@ns1.sylogix.net) |
| 06:48.38 | *** join/#asterisk joelsolanki (~joelsolan@123.237.174.136) |
| 06:50.18 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:53.40 | joelsolanki | is it possible that calls which are at 15 mins gets auto disconnect ? i just want any calls to go beyond 15 mins. |
| 06:53.55 | p3nguin | Take a look at session-timers. |
| 06:54.03 | joelsolanki | i just dont want calls to go beyond 15 mins i mean |
| 06:54.10 | joelsolanki | session timers. ok let me see |
| 06:54.55 | p3nguin | There is also an associated setting, but I'd have to go look to see the name. Might be session-expire or something similar. |
| 06:55.45 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 06:57.43 | joelsolanki | oh ok |
| 07:01.22 | kaldemar | or option L() in application Dial. session timers are just for SIP. |
| 07:01.38 | p3nguin | Alrighty, then. Unpatched 1.8.7.0 res_jabber.c and the problem seems to be gone. |
| 07:01.55 | p3nguin | Asterisk is connected to jabber and no warnings. |
| 07:02.47 | kaldemar | actually, the session timers are just for refreshing the session. they cannot necessarily be used to limit call duration. |
| 07:03.21 | kaldemar | only if the other end doesn't support session timers. |
| 07:04.41 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e2.bcn.adamo.es) |
| 07:06.05 | p3nguin | This moh problem is really annoying. I've preloaded all 19 format_*.so modules, but still mp3 class will not play until I have run module unload and then module load on res_musiconhold.so. |
| 07:06.46 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 07:06.49 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 07:06.54 | p3nguin | Verbose output says starting musiconhold class mp3, and lsof shows that mpg123 is playing files, but there's no sound from them. |
| 07:07.05 | p3nguin | unload, load, works fine. |
| 07:10.02 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 09:20.32 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 09:20.32 | *** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.6.0 (2011/08/31), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 09:26.44 | *** join/#asterisk ollii (~risker@port-87-193-161-154.static.qsc.de) |
| 09:29.20 | joelsolanki | ok i checked session timer stuff i will test it. |
| 09:29.29 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 09:29.32 | joelsolanki | but is it possible to disconnect calls at exact 15 mins ? |
| 09:30.01 | kaldemar | joelsolanki: yes, but not with session timers as stated earlier. |
| 09:31.02 | joelsolanki | ok then what can i use to disconnect calls at 15 mins ? |
| 09:31.30 | kaldemar | i already told you... |
| 09:33.23 | joelsolanki | let me check it again |
| 09:33.51 | joelsolanki | option L() in application Dial |
| 09:33.55 | joelsolanki | you mean right ? |
| 09:34.19 | kaldemar | yes |
| 09:34.19 | irroot | joelsolanki kaldemar yip it cuts the user off at this Limit |
| 09:34.33 | joelsolanki | great. :) |
| 09:42.59 | *** join/#asterisk mutex7c (~mutex7c@212.184.118.18) |
| 09:45.12 | WIMPy | Or TIMEOUT(absolute) |
| 09:52.16 | devil_evoxxx | irroot: this, has significant value for dtmf not working trough quescom? Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) |
| 09:52.23 | devil_evoxxx | i'm debungging |
| 09:54.47 | irroot | that seems normal too |
| 09:55.18 | irroot | its combined is 0x1 not 0x0 |
| 09:55.28 | irroot | when you sip debug |
| 09:55.39 | irroot | and press a button on the phone while in a call |
| 09:55.56 | irroot | you should see sip messages with the dtmf pressed |
| 09:56.21 | irroot | make sure that the right ammount of messages and that the right messages are coming through |
| 09:56.21 | devil_evoxxx | ..i not see sip message with dtmf |
| 09:56.22 | devil_evoxxx | only |
| 09:56.28 | devil_evoxxx | on |
| 09:56.32 | devil_evoxxx | debug |
| 09:56.34 | devil_evoxxx | core debug |
| 09:56.36 | irroot | you on sip phones ?? |
| 09:56.37 | devil_evoxxx | i see |
| 09:56.46 | devil_evoxxx | dtmf start and dtmf end messages |
| 09:56.52 | irroot | sip set debug ip ..... |
| 09:57.00 | irroot | the ip of the quescom |
| 09:57.26 | kaldemar | dtmf won't show up in sip debug unless dtmfmode is info. |
| 09:58.19 | devil_evoxxx | is setted in rf2833 :( |
| 09:58.58 | devil_evoxxx | the only message i can see is << [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [SIP/voce-inc-000001dd] |
| 09:59.01 | devil_evoxxx | >> [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [SIP/fromff-000001de] |
| 09:59.25 | devil_evoxxx | and before, the corresponding message , but with start |
| 10:03.25 | *** join/#asterisk Diffen (~diffen@host-90-238-142-129.mobileonline.telia.com) |
| 10:05.02 | irroot | so it goes |
| 10:05.13 | irroot | that is right then |
| 10:05.31 | irroot | there is a start and end frame |
| 10:05.49 | irroot | devil_evoxxx try change to sip info type |
| 10:10.11 | *** join/#asterisk enoch (~enoch@unaffiliated/enoch) |
| 10:10.12 | enoch | hi all |
| 10:10.46 | enoch | guys i need a suggestion... a sip client with easy call-forward function... |
| 10:11.17 | enoch | i mean that an user should be able to forward a call to another sip user by clicking it on the address book |
| 10:11.53 | enoch | jitsi permits it but have a poor quality |
| 10:22.55 | devil_evoxxx | irroot: ok |
| 10:34.02 | *** join/#asterisk jkroon (~jkroon@dsl-242-10-149.telkomadsl.co.za) |
| 10:34.06 | devil_evoxxx | irroot: if i set info in dtmf |
| 10:34.12 | devil_evoxxx | i can not seee anything in |
| 10:34.24 | devil_evoxxx | the asterisk box where i'm debugging |
| 10:38.48 | irroot | you need to reload the peer/sip to get it to updat4e |
| 10:40.20 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 10:45.45 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 10:50.07 | devil_evoxxx | yes ..i've reloaded it.. |
| 10:50.15 | devil_evoxxx | but |
| 10:50.34 | devil_evoxxx | i can see only Null Frame |
| 10:55.53 | devil_evoxxx | and no DTMF |
| 11:00.08 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:13.55 | *** join/#asterisk lister_ (~lister@scarda1.lnk.telstra.net) |
| 11:16.43 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 11:19.09 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 11:24.58 | *** join/#asterisk atheos (~atheos@208.119.68.16) |
| 11:36.19 | *** join/#asterisk wengole (~bcole@178.78.119.76) |
| 11:38.06 | *** join/#asterisk SteveWilliams (~SteveWill@220.224.235.78) |
| 11:38.08 | *** part/#asterisk wengole (~bcole@178.78.119.76) |
| 11:38.17 | *** join/#asterisk wengole (~bcole@178.78.119.76) |
| 11:39.30 | *** join/#asterisk nix8n82-phone (~AndChat@63.sub-174-253-176.myvzw.com) |
| 11:47.02 | *** join/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76) |
| 11:47.16 | wengole | o/ |
| 11:47.25 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 11:47.27 | Vilius_Invade | \o |
| 11:52.00 | SeRi | guys a "all circuits are busy" message its a carrier issue or a user issue? |
| 11:55.29 | *** join/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76) |
| 11:55.39 | SeRi | this is happening to one number only but not the other number in the same carrier. I get a few calls in and than I start getting a "all circuits are busy please try your call later" |
| 11:58.16 | *** part/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76) |
| 11:58.30 | *** join/#asterisk Vilius_Invade (~Vilius_In@178.78.119.76) |
| 11:59.05 | *** join/#asterisk cjk (~cjk@85.93.217.128) |
| 12:00.13 | cjk | hi, my asterisk misses dtmf's for sip calls but dtmf work fine on dahdi. I played with dtmf mode without success. any idea on how to debug this? |
| 12:00.15 | *** part/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 12:05.15 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 12:05.22 | *** join/#asterisk PHANTOM`` (~phantom@203.212.18.190) |
| 12:05.22 | wengole | SeRi: "all circuits are busy" is (unfortunately) a generic failure message played for any failure reason. Look in the logs for HANGUPCAUSE= to get the ISDN result code |
| 12:05.41 | *** join/#asterisk neurosys (~neurosys@107.49.135.68) |
| 12:05.53 | Vilius_Invade | cjk: have you tried playing with dtmfmode setting in sip.conf? |
| 12:06.54 | cjk | Vilius_Invade, yes tried them all |
| 12:07.47 | *** join/#asterisk Guest60547 (~naomi@79.135.102.10) |
| 12:08.51 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 12:10.10 | Vilius_Invade | cjk: i would do a packet traces and see if dtmf is coming in from your softphone or whatever you are using |
| 12:10.25 | Guest60547 | hi, when i press #1 to transfer a call, and then start to type the digits, its only waits for 1 digit and says its invalid |
| 12:10.50 | kaldemar | cjk: enable core debug, then you'll see detected DTMF. |
| 12:11.39 | kaldemar | cjk: and what do you mean by asterisk missing DTMF? |
| 12:14.36 | SeRi | wengole, will do thanks for the help. |
| 12:15.31 | wengole | SeRi: no problem :) |
| 12:17.08 | *** join/#asterisk _naomi (~naomi@79.135.102.10) |
| 12:17.33 | cjk | kaldemar, exactly, on the cli I see that asterisk constantly misses one dtmf |
| 12:17.45 | _naomi | hi, sorry didnt log in properly before. Having problem with #1 transfer - it only waits for 1 digit then says invalid ext |
| 12:17.51 | cjk | kaldemar, about the 5th or 6th digit I press does not appear on the cli |
| 12:18.01 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 12:19.36 | *** join/#asterisk bchia (~Adium@nat/digium/x-awcolojnubrnqrht) |
| 12:22.26 | _naomi | transferdigittimeout is set to 3 seconds |
| 12:23.13 | SeRi | wengole, I dont see anything related to HANGUPCAUSE in my logs. |
| 12:23.13 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 12:24.43 | azv4 | If I want to find a company that can offer support for our company's phone system, what would I search? I search "Business Telephone Support" and I get listings for internet services and child support websites! |
| 12:25.02 | SeRi | this odd I have two numbers routed to the same system one works fine and the other one works sporadically... |
| 12:30.20 | kaldemar | cjk: are you experiencing packet loss? |
| 12:36.02 | cjk | kaldemar, no, not all all the network is perfect |
| 12:36.41 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 12:42.11 | *** join/#asterisk psilikon (~joel@rrcs-24-73-84-158.se.biz.rr.com) |
| 12:44.04 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:46.54 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 12:52.34 | cusco | hi folks |
| 12:53.12 | *** join/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 12:53.16 | cusco | when using queue() can I specify that a peer has higer priority other than using removequeuemember() and addqueuemember() ? |
| 12:53.23 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 12:53.46 | Faustov | how often is the meetme.conf [rooms] context read? I thought every module app_meetme gets reloaded, but it seems it is being read on the fly - why? |
| 12:54.36 | irroot | cusco there is a option to set rules that after x time the the priority jumps |
| 12:55.12 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:57.24 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:02.33 | kaldemar | [TK]D-Fender: welcome back |
| 13:03.36 | kaldemar | cjk: if you're using rfc2833, try rtp debug or dump the network interface to see if you actually get the packet. |
| 13:05.13 | *** join/#asterisk serafie (~erin@nat/digium/x-qsorbmvazmpnfyue) |
| 13:09.02 | [TK]D-Fender | kaldemar, mornin' |
| 13:12.03 | *** join/#asterisk nix8n82-phone (~AndChat@24.143.28.16) |
| 13:14.24 | jaytee | mornin [TK]D-Fender |
| 13:15.02 | [TK]D-Fender | jaytee, y0 |
| 13:16.31 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:17.27 | *** part/#asterisk paec (~user@AAmiens-551-1-68-36.w92-142.abo.wanadoo.fr) |
| 13:18.56 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 13:22.03 | *** join/#asterisk wesphillips (~wphill04@137.237.194.133) |
| 13:23.56 | *** join/#asterisk SteveWilliams (~SteveWill@220.224.235.78) |
| 13:24.18 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 13:25.03 | SteveWilliams | I would like to know how to configure Sangoma UT51 hardware timer for my asterisk server? |
| 13:26.34 | *** part/#asterisk wengole (~bcole@178.78.119.76) |
| 13:28.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:35.15 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 13:35.40 | *** join/#asterisk clarkmili (~clarkmili@bl14-193-132.dsl.telepac.pt) |
| 13:35.50 | ollii | SteveWilliams: ask #sangoma |
| 13:36.00 | ollii | they have their own channel :) |
| 13:36.06 | ollii | with sangoma employees |
| 13:41.17 | Katty | drags in |
| 13:41.30 | irroot | Katty yo there |
| 13:41.56 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:42.11 | [TK]D-Fender | Katty, Mew. |
| 13:44.32 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:45.55 | Katty | hugs irroot |
| 13:46.18 | Katty | [TK]D-Fender: hello |
| 13:46.52 | Dovid | wow. TK |
| 13:46.55 | Dovid | it's been a while |
| 13:49.05 | clarkmili | hi everybody |
| 13:50.07 | clarkmili | someone has compiled *1.6.2 with amr support? |
| 13:50.33 | *** join/#asterisk oldhack (~oldhack@cpe-24-28-23-78.austin.res.rr.com) |
| 13:50.44 | Katty | what's the word |
| 13:52.46 | clarkmili | or know how-to use a patch to 1.6.0 version on 1.6.2 |
| 13:53.33 | leifmadsen | anything in 1.6.0 would be in 1.6.2 |
| 13:53.51 | leifmadsen | if it's a third party module or change, then you'll need to port it yourself |
| 13:54.31 | [TK]D-Fender | Katty, Haven't you heard? http://www.youtube.com/watch?v=2WNrx2jq184 |
| 13:56.04 | clarkmili | yes, ok... I'll check the link |
| 13:56.24 | clarkmili | lol... yes, I hear you |
| 14:01.45 | Naikrovek | ohmygosh d-fender is back |
| 14:01.53 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:02.56 | leifmadsen | freak out! |
| 14:03.45 | Dovid | lol |
| 14:06.05 | *** join/#asterisk master_of_master (~master_of@p57B53759.dip.t-dialin.net) |
| 14:08.59 | p3nguin | So how do we get patches that have been implemented taken back out? The one for res_jabber.c broke shit. |
| 14:09.21 | p3nguin | I don't want to have to unpatch every time I need to upgrade. |
| 14:10.09 | pabelanger | p3nguin: reopen the JIRA issue |
| 14:10.20 | p3nguin | Let me see if I can figure out how. |
| 14:11.19 | *** join/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu) |
| 14:11.59 | leifmadsen | clikc the Reopen button :) |
| 14:13.00 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 14:13.47 | MarKsaitis | hi |
| 14:14.21 | MarKsaitis | if I am calling a number 212.45.345.23###452334, what kind of voip number is this? |
| 14:14.28 | p3nguin | I realize I am a littler jira-retarded, but will you tell me how to fix the reopen button? |
| 14:14.32 | MarKsaitis | what are all these # after the IP and numbers after that? |
| 14:14.37 | MarKsaitis | could somebody please clarify |
| 14:14.59 | p3nguin | s/fix/find/ |
| 14:15.48 | p3nguin | It says issues which are closed can be reopened, but I don't see any button for that. |
| 14:16.23 | MarKsaitis | pls help? |
| 14:16.45 | Naikrovek | that's not a voip number. |
| 14:16.55 | Naikrovek | a "voip number" is the same as a "regular number" most of the time |
| 14:17.11 | leifmadsen | p3nguin: you're logged in? |
| 14:17.14 | p3nguin | yes |
| 14:17.26 | leifmadsen | what is the issue number? who knows with jira, permissions are wonky sometimes |
| 14:17.35 | leifmadsen | it'll just be easier for me to reopen it |
| 14:17.40 | p3nguin | ASTERISK-18078 |
| 14:17.42 | Naikrovek | MarKsaitis: OR, it'll be something like "201@54.234.11.5" |
| 14:18.06 | MarKsaitis | Naikrovek, do you know what my sample would be? |
| 14:18.45 | Naikrovek | no |
| 14:19.00 | MarKsaitis | it actually has two ## |
| 14:19.16 | p3nguin | Maybe it's some goofy way of dialing a SIP URI. |
| 14:19.21 | MarKsaitis | is that sip in ur example? |
| 14:19.23 | leifmadsen | p3nguin: actually, while I reopened it, I think a new issue that I link as "Caused By" ASTERISK-18078 would be better to be honest |
| 14:19.29 | leifmadsen | reopening issues causes things to .... get stick |
| 14:19.32 | leifmadsen | sticky* |
| 14:19.47 | p3nguin | leifmadsen: http://svnview.digium.com/svn/asterisk?view=revision&revision=333265 |
| 14:19.53 | p3nguin | leifmadsen: This patch is the problem. |
| 14:19.54 | leifmadsen | p3nguin: yes.... |
| 14:19.57 | leifmadsen | ok |
| 14:20.07 | leifmadsen | let me know what the new issue is and I'll link them |
| 14:20.09 | leifmadsen | and get it triaged |
| 14:21.12 | leifmadsen | p3nguin: I reclosed ASTERISK-18078 and will link the new issue as a regression against ASTERISK-18078 |
| 14:21.46 | p3nguin | Running 1.8.6.0, no problems... upgrade to 1.8.7.0, and res_jabber spews this every 55 seconds: [Sep 26 01:06:54] WARNING[7052]: res_jabber.c:1473 aji_send_raw: JABBER: Unable to send message to asterisk, we are not connected[Sep 26 01:06:55] WARNING[7052]: res_jabber.c:1737 aji_act_hook: Jabber didn't seem to handshake, failed to authenticate. |
| 14:22.11 | leifmadsen | grumbles something about testing RCs before releases are made |
| 14:22.25 | p3nguin | It comes from having asterisk as a component for a peered jabber server. |
| 14:22.45 | leifmadsen | p3nguin: make sure you mark all that down in the issue you're going to open |
| 14:22.59 | p3nguin | drmessano and I both unpatched and the problem went away. |
| 14:23.03 | leifmadsen | I get that |
| 14:23.10 | leifmadsen | open a new issue |
| 14:23.11 | leifmadsen | I'll triage it |
| 14:23.29 | Katty | pamples things |
| 14:24.17 | pabelanger | p3nguin: is that all it is doing? Just writing a WARNING or is there an actual problem with the patch? |
| 14:24.39 | pabelanger | eg: It was working in 1.8.6.0 but stopped in 1.8.7.0 |
| 14:25.10 | p3nguin | What the warning says is actually happening. It says cannot handshake/authenticate, and it never connects to the jabber server. |
| 14:25.36 | p3nguin | Stays in "Connecting" forever while giving the warning repeatedly. |
| 14:25.47 | p3nguin | Unpatch it, and problem is solved. |
| 14:26.38 | p3nguin | Weird. 1.8.7.0 is listed as an unreleased version. |
| 14:27.04 | leifmadsen | p3nguin: because I haven't clicked the little button yet -- forgot to do that on Friday |
| 14:28.01 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:28.19 | p3nguin | Issue found by: ... am I customer (I usually don't think of myself as a customer unless I am buying something)? |
| 14:28.30 | *** join/#asterisk Maxxed (~Maxxed@216.215.95.118) |
| 14:28.43 | pabelanger | p3nguin: ignore it |
| 14:28.59 | leifmadsen | Just worry about Affects Version and Component |
| 14:29.08 | leifmadsen | and click Regression: Yes |
| 14:29.11 | pabelanger | It should not be exposed for Asterisk users |
| 14:30.22 | *** join/#asterisk clarkmili (~clarkmili@bl14-193-132.dsl.telepac.pt) |
| 14:30.37 | Faustov | how often is the meetme.conf [rooms] context read? I thought every module app_meetme gets reloaded, but it seems it is being read on the fly - why? |
| 14:31.02 | leifmadsen | why not? |
| 14:31.23 | leifmadsen | if it's not per a manual reload, then that means it is read each time MeetMe() is called |
| 14:32.09 | leifmadsen | I don't use that meetme.conf file, so I don't know for certain if it's per reload or not |
| 14:33.56 | clarkmili | I want to use * with AMR support, how-to upgrade a existing 1.6.0 AMR patch to 1.6.2 version |
| 14:34.12 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:34.58 | leifmadsen | clarkmili: asked and answered |
| 14:35.17 | leifmadsen | clarkmili: if it's standard in 1.6.0, then it's already in 1.6.2. If that's a third party patch, then you will have to port it to 1.6.2 |
| 14:36.05 | Faustov | leifmadsen: it is confusing for management. Configuration files should be read upon reload unless otherwise stated. When there are changes performed in a larger project involving config changes, everything is usng the old settings until a reload command is issued |
| 14:36.17 | leifmadsen | ok |
| 14:36.22 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 14:36.23 | Faustov | to read one of all the files on the fly is just against common sense |
| 14:36.43 | Faustov | sorry if I sound a bit sour after a "fukup", trying to think objectively |
| 14:36.51 | cjk | kaldemar, sorry for the late reply, but on the network level (wireshark sniff) the dtmf is received correctly, it just does not appear on the asterisk cli. |
| 14:38.02 | clarkmili | hum... too complicated... |
| 14:38.52 | *** join/#asterisk jkroon (~jkroon@197.170.168.41) |
| 14:38.56 | clarkmili | anyway, thanks |
| 14:39.42 | p3nguin | Done. ASTERISK-18626 |
| 14:42.01 | beek | mornin' gang. |
| 14:42.40 | leifmadsen | p3nguin: thanks will triage shortly |
| 14:43.23 | cjk | I receive DTMF's over SIP (rfc 2833) , I see them all in the wireshark sniff, but asterisk slips one dtmf. It does not appear on the cli. any idea? |
| 14:45.28 | clarkmili | which dtmf mode is used |
| 14:45.30 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:47.55 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 14:50.15 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 14:51.12 | irroot | p3nguin ASTERISK-18626 can you please put a ast_verb line in there to show the timeout ?? and status so we can see what is missing |
| 14:51.44 | p3nguin | I have no idea what that means. |
| 14:52.02 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:52.06 | _naomi | hi everyone, im quite new here and new to irc in general |
| 14:52.09 | irroot | ok if put a patch on the issue you can run it ?? |
| 14:52.26 | p3nguin | Maybe. |
| 14:52.31 | irroot | lets see if there is feedback |
| 14:52.36 | p3nguin | Okay. |
| 14:53.08 | irroot | then we can add a debug line to see what that commit is missing |
| 14:53.10 | p3nguin | I filed the issue about my moh, too. ASTERISK-18627 |
| 14:53.15 | _naomi | not sure of the ettiquette |
| 14:53.17 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 14:54.08 | p3nguin | I have only tested two asterisk versions with the problem, so I hope readers of the issue don't think those are the only two versions where it happens... I'd imagine it happens on earlier 1.8 versions, but I didn't test them to know. |
| 14:54.16 | leifmadsen | _naomi: basically, don't be a douche, and just go ahead and ask your question :) |
| 14:54.25 | *** join/#asterisk navaismo (~IceChat77@201.123.42.167) |
| 14:54.32 | beek | _naomi: Welcome! |
| 14:54.48 | p3nguin | _naomi: And don't flood us if you have something to share. If you have something to paste, put it in the pastebin. |
| 14:54.57 | irroot | ASTERISK-18078 p3nguin fyi thats the original issue |
| 14:55.43 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:04.50 | *** join/#asterisk d_preston215 (~chatzilla@173-12-4-137-panjde.hfc.comcastbusiness.net) |
| 15:06.08 | d_preston215 | Is work being done on dynamic spans (especially regarding redFONE devices) to fix the fact that DAHDI 2.5 crashes the server when I use it? |
| 15:06.59 | leifmadsen | I doubt it |
| 15:07.08 | leifmadsen | I don't even think a bug was opened for that |
| 15:07.14 | leifmadsen | at least from what I've seen |
| 15:07.27 | d_preston215 | Any chance of that being worked on if I send in a bug report? |
| 15:10.05 | leifmadsen | d_preston215: not sure, depends on what the priority ends up being for a developer |
| 15:10.15 | leifmadsen | there is always a chance |
| 15:10.21 | *** join/#asterisk navaismo (~navaismo@201.123.42.167) |
| 15:11.02 | _naomi | hi thanks i'll try not to be a douche! |
| 15:11.13 | d_preston215 | Thanks. |
| 15:11.27 | d_preston215 | I'll put in a bug report for this if the people at redFONE haven't. |
| 15:11.31 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 15:12.10 | _naomi | having prob with attended transfer feature in 1.6.2.20 - it only waits for 1 digit then says invalid extension. transferdigittimeout is 3 secs |
| 15:12.32 | _naomi | wondering if should report it as bug or am i missing something |
| 15:12.42 | d_preston215 | Which although they (people at redFONE) know that DAHDI 2.5 is broken for their products, probably haven't sent one in yet. |
| 15:13.27 | leifmadsen | ~asteriskversions |
| 15:13.33 | leifmadsen | ~asteriskversioning |
| 15:13.34 | infobot | Asterisk versions and their support levels are documented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 15:13.38 | leifmadsen | _naomi: ^^^ |
| 15:13.56 | leifmadsen | you'll have to reproduce on 1.8 or later as 1.6.2 is no longer receiving bug maintenance |
| 15:15.06 | *** join/#asterisk scubes13 (~scubes13@rrcs-70-60-211-241.midsouth.biz.rr.com) |
| 15:18.56 | *** join/#asterisk dms (~dms@nat/digium/x-lpsuishlsyomqfps) |
| 15:20.37 | _naomi | any other config settings i should check first? AFAICS its just the transferdigittimeout |
| 15:23.23 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 15:23.46 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:24.54 | *** part/#asterisk asterisk-Tester (~RAMYT@210.5.215.40) |
| 15:25.41 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 15:25.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:29.46 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:30.11 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:30.35 | Micc | naomi, have you tried previous versions? |
| 15:38.50 | _naomi | 1.6.2.19 is the same. im thinking my config must be wrong since no mention of this apparent bug online |
| 15:38.50 | Qwell | leifmadsen: ping! |
| 15:38.56 | leifmadsen | Qwell: pong! |
| 15:39.03 | Qwell | leifmadsen: I was poked about the topic being out of date |
| 15:39.31 | *** topic/#asterisk by leifmadsen -> #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.7.0 (2011/09/26), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 15:39.38 | leifmadsen | Qwell: not sure what you're talking about -- looks right to me |
| 15:39.40 | *** join/#asterisk luckman212 (~irc@pool-108-41-8-176.nycmny.fios.verizon.net) |
| 15:39.59 | coppice | we're out of dates too, and figs |
| 15:40.57 | Qwell | >.> |
| 15:41.16 | leifmadsen | Qwell: :) |
| 15:43.04 | Micc | _naomi, try 1.6.2.17.3 |
| 15:43.19 | Micc | _naomi, then if its still the same its probably your config. |
| 15:43.51 | _naomi | great thanks Micc, will do |
| 15:43.55 | *** join/#asterisk devcoder (~leemelnyk@216.18.243.44) |
| 15:44.33 | Micc | 1.6.2.18 and 1.6.2.19 have a lot of known issues. I haven't tried .20 but I think some of the issues were not fixed because it was already end of life by the time 1.6.2.20 came out just to fix a major crash condition. |
| 15:47.31 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 15:49.04 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176138825.dsl.bell.ca) |
| 15:50.14 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 15:50.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 15:53.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:53.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:54.00 | p3nguin | drmessano: With what version of ejabberd did you test that authentication problem? |
| 15:58.52 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:03.27 | devcoder | hey can anyone help me out with a cisco 7975 phone. I want to put sip on it but can't download it from cisco. |
| 16:03.57 | p3nguin | The problem is that you don't have SIP firmware for the phone? |
| 16:04.07 | devcoder | that is correct |
| 16:04.19 | p3nguin | Do you have SCCP firmware on it now? |
| 16:04.51 | devcoder | yeah. I know you can make a that work but it would just be nicer to have sip i think |
| 16:04.58 | p3nguin | I doubt it. |
| 16:05.07 | [TK]D-Fender | ~pri |
| 16:05.07 | infobot | pri is, like, [~pri] Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, R1T1,R2T1,R4T1, etc. |
| 16:06.00 | p3nguin | I don't have a 7975, but I do have a 7960 and 7940... and SIP on them sucks, so I use SCCP for my Cisco phones. |
| 16:06.16 | _naomi | yep its the same with 1.6.2.17.3 |
| 16:06.35 | _naomi | any idea what could be the prob with the config? |
| 16:07.00 | devcoder | maybe i will try that then. the phone is nice. i have the 7940 and 7960 stuff also. |
| 16:07.11 | [TK]D-Fender | _naomi, look at the contexts things are pointing to |
| 16:08.04 | _naomi | [TK]D-Fender, do you mean in /etc/asterisk/features.conf? |
| 16:08.10 | devcoder | its been a while since I ran asterisk, I am assuming the skinny support has been greatly improved? |
| 16:08.27 | [TK]D-Fender | _naomi, no, generally those of the device doing the transfer |
| 16:08.48 | p3nguin | Hmm, when'd you slip in? |
| 16:08.56 | [TK]D-Fender | _naomi, if it cuts you off it's quite likely that it is because they can't dial the number you are starting to dial. |
| 16:10.20 | _naomi | you can make an internal call to 100 without any issues. But you can't transfer a call to 100 or to any number at all |
| 16:11.12 | p3nguin | Several hours ago, it seems. |
| 16:11.26 | [TK]D-Fender | p3nguin, referring to me? Yes, 8:55am EST |
| 16:11.35 | p3nguin | Yep. Didn't see you here. |
| 16:11.42 | [TK]D-Fender | p3nguin, and yes, first time since back then. |
| 16:11.47 | Katty | it's good to see fender bender |
| 16:11.56 | Katty | now i have someone to pester again |
| 16:12.10 | p3nguin | Now I won't always be the bad guy. |
| 16:12.13 | [TK]D-Fender | Katty, You'd found me elsewhere, I guess I just saved you a small trip ;) |
| 16:12.30 | [TK]D-Fender | p3nguin, Actually.. the odds have increased... |
| 16:12.35 | p3nguin | oh |
| 16:12.42 | Katty | yeah a whole alt+number |
| 16:12.53 | p3nguin | Slowing down in your old age, huh? |
| 16:13.25 | Katty | is going to have a birthday soonish |
| 16:13.38 | p3nguin | checks the calendar |
| 16:14.08 | [TK]D-Fender | Katty, Already? But you had one last year.... |
| 16:14.30 | Katty | *hee* |
| 16:14.39 | Faustov | Katty: I figured out that birthdays can be disruptive or "not really", based on how many bits you need to change to write your new age... so which one is yours? ;) |
| 16:15.03 | p3nguin | less than two weeks to go! |
| 16:16.23 | Katty | Faustov: i'm livin the dream baby |
| 16:16.33 | Katty | Faustov: in my prime and enjoyin every minute of it |
| 16:16.52 | _naomi | i'm on the right track now, i see its somehow configured to use a context that it should not be trying to use |
| 16:16.58 | _naomi | thanks for all the help everyone |
| 16:18.11 | Faustov | Katty: which means you should be quite attractive, but if you were, you wouldn't be here. Does not compute |
| 16:18.29 | Faustov | is fooling around, don't pay attention pls |
| 16:19.31 | Katty | i am a firm believer that someone's appearance has little to do with anything |
| 16:19.45 | p3nguin | except how nice they look. :) |
| 16:19.45 | Katty | maybe choice of eyeshadow color, but that's about it |
| 16:20.06 | Katty | i can wear sweatpants with the best of them *hee* |
| 16:20.07 | [TK]D-Fender | </tammyfaye> |
| 16:20.16 | p3nguin | Oh man. |
| 16:20.21 | [TK]D-Fender | The best of them.. don't wear sweatpants... |
| 16:20.23 | p3nguin | I haven't seen her in years. |
| 16:20.36 | Faustov | Katty: I'm a firm observer that people who find themselves not attractive are very annoying because they are always unhappy |
| 16:21.07 | coppice | realism often makes you unhappy |
| 16:21.19 | Faustov | I'm not saying it is wrong |
| 16:21.20 | Katty | not enough hugs often leads to unhappiness and low self esteem |
| 16:21.36 | p3nguin | And those who insist they are very attractive are usually rather ugly in other ways. |
| 16:21.47 | Faustov | that's often true |
| 16:22.03 | p3nguin | coppice: I reject your reality and substitute my own. |
| 16:22.12 | Katty | that's just because no one tells them they're crap. |
| 16:22.22 | Katty | they don't know they need to grow as an individual |
| 16:22.50 | Katty | maybe i should make this a life goal..hmm |
| 16:22.51 | Faustov | glad there are people that can tell others who they are and where they belong, world wouldn't be the same without them ;) |
| 16:23.03 | Faustov | perfect, Katty ;) |
| 16:24.49 | Katty | Faustov: are you joining the xmas card exchange this year? |
| 16:25.10 | Faustov | Katty: first time I hear about it |
| 16:25.41 | Faustov | I get to send you guys anthrax^Hxmas cards? |
| 16:27.01 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 16:29.15 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 16:31.02 | Katty | Faustov: it's just a list of people who have signed up. you don't have to send cards to everyone, just the people you know and talk to on a regular basis... i'll probably send cards to everyone on the list tho |
| 16:31.10 | *** join/#asterisk InsektO (~InsektO@janus.all-kom.com.ar) |
| 16:31.29 | *** join/#asterisk irroot (~irroot@41.51.116.169) |
| 16:31.42 | Katty | quick! everyone hide! |
| 16:32.29 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 16:32.51 | as001 | Hello I want to set channel group but I get error ERROR[5581]: pbx.c:3401 ast_func_write: Function Group not registered I use 1.6.2.20 |
| 16:33.51 | irroot | any idea why when dialing into a siemens i dont get ringing on the ip phone while proceeding even with a 180/183 |
| 16:34.12 | as001 | in dialplan I get this Set(Group()=9999) Is it ok ? |
| 16:34.12 | Katty | irroot: did you try hugging it? |
| 16:34.15 | irroot | the siemens is on PRI im net |
| 16:34.20 | Katty | irroot: turning it off, and back on again? |
| 16:34.42 | irroot | its few 100 km away and not in a nice part of the country wild animals and tropical conditions :P |
| 16:34.47 | p3nguin | as001: Try GROUP instead. |
| 16:34.58 | irroot | maybe the monkeys did katty ?? |
| 16:34.59 | *** join/#asterisk jerware (~jerryg@c-71-58-179-44.hsd1.pa.comcast.net) |
| 16:35.00 | as001 | oh thanks |
| 16:35.01 | jerware | folks. |
| 16:36.27 | jerware | Somone (a college kid) advised not to go with open source when it comes to VoIP but couldn't explain why. |
| 16:36.31 | jerware | and recomended cisco. |
| 16:36.45 | p3nguin | He must be a reseller. |
| 16:38.20 | irroot | jerware look how its going for cisco |
| 16:38.59 | irroot | college kids are there cause they know for sh1t if they were any good they would not go |
| 16:40.03 | *** join/#asterisk imox (~imox@p4FC5C75B.dip0.t-ipconnect.de) |
| 16:40.35 | InsektO | hi, i'm trying to figure out how 'rrmemory' strategy works if i have 3 or more agents logged in. what is the order agents are rung? |
| 16:41.25 | [TK]D-Fender | Cirular from the last one called based on the order they were added to the queue |
| 16:41.27 | irroot | InsektO its random but once a pattern is established it remembers who was last |
| 16:41.52 | [TK]D-Fender | No "last caller" = first in list goes first. Circular thereafter |
| 16:42.08 | coppice | [TK]D-Fender: you've served your sentance? |
| 16:42.11 | irroot | [TK]D-Fender indeed |
| 16:42.43 | InsektO | ahh, thanks irroot, thats precisely what i did not know (the random part) |
| 16:43.13 | irroot | round robin memory |
| 16:43.24 | irroot | so it remembers |
| 16:43.39 | [TK]D-Fender | coppice, Something like that I guess |
| 16:43.52 | [TK]D-Fender | InsektO, isn't "random". |
| 16:44.09 | [TK]D-Fender | InsektO, Starts with the first and circular in order of add from the last one to take a call. |
| 16:45.55 | InsektO | mm, so its kinda similar to the 'linear' strategy? i dont get the difference (agents will be static) |
| 16:49.16 | [TK]D-Fender | InsektO, it's that it remembers who last took a call to ring for the next caller |
| 16:49.24 | [TK]D-Fender | InsektO, Ensures more even distribution |
| 16:51.34 | Katty | wants to live in tropical conditions. |
| 16:51.42 | InsektO | ok, it remembers the last, but what criteria does it use to choose who's next? (if they are static agents) order in the list? |
| 16:52.32 | [TK]D-Fender | InsektO, As I told you, it continues in the order in which they were added to the queue |
| 16:52.38 | InsektO | (sorry for all the question, but im trying to understand which option is more convenient) |
| 16:52.46 | InsektO | ok, thanks [TK]D-Fender |
| 16:53.08 | [TK]D-Fender | InsektO, You're welcome. |
| 16:59.20 | Micc | anyone have experience with aastra phones using tcp? |
| 17:00.31 | Micc | I'm having a bit of a problem even with a very recent firmware on the phone. Its saying it failed registration because its not a valid domain. |
| 17:00.56 | Micc | the domain is the ip:port of the phone, not the server for some reason. seems like a phone problem to me, but I'm not sure. |
| 17:01.29 | Katty | irroot: what does your country do with illegal aliens? |
| 17:01.50 | irroot | gives them ID's and passports then there is no problem right :P |
| 17:02.10 | Katty | i can't believe your country does that |
| 17:03.07 | Micc | jerware, we steal customers from cisco all the time because they can't get their voip stuff to work right. |
| 17:03.09 | irroot | Katty seriously its not as easy as that but with millions of them its easier to tax em and get them legal |
| 17:03.41 | irroot | the cost of deporting them is quite hi the ones that dont qualify get put on a train home |
| 17:04.00 | Katty | they'd have a hard time getting me on a train home |
| 17:04.06 | Katty | unless the train goes across the pond |
| 17:04.46 | irroot | Katty lol most are "refugees" from zimbabwe/mozambique |
| 17:05.08 | Katty | i couldn't pass as a refugee |
| 17:05.18 | jaytee | so give 'em a free DVD of The Lion King and kick them the hell out! |
| 17:05.26 | Faustov | Katty: are the same people meeting at astricon? |
| 17:05.49 | Katty | Faustov: i'm not sure who all on the list is going to astricon, but you could probably ask |
| 17:06.16 | Faustov | Katty: I've been trying to get the company to send me there for a while now ;( |
| 17:06.22 | Faustov | sigh, one day |
| 17:06.27 | Katty | hugs Faustov |
| 17:06.27 | irroot | jaytee Katty they regularly find eaten bodies in the bush there is no need for lion king DVD when you get met and eaten |
| 17:06.33 | Faustov | anyway, off work, time to go home |
| 17:06.36 | Faustov | o |
| 17:06.38 | Faustov | o/ |
| 17:08.06 | irroot | Katty but labor is cheap 200R per day thats 25$ for 8hrs thats 3$/hr |
| 17:08.17 | Katty | goodness. |
| 17:08.17 | *** part/#asterisk clintc (~clintc@n128-227-109-39.xlate.ufl.edu) |
| 17:08.24 | Katty | i guess the cost of living is a bit less than here tho |
| 17:08.34 | Katty | i can't even buy lunch for 3 bucks |
| 17:08.42 | Katty | i probably can't drive to work for 3 bucks |
| 17:11.02 | *** join/#asterisk italorossi (~italoross@201.76.154.127.intranet.digi.com.br) |
| 17:11.26 | citywok | Katty: how far do you live? that's a little less than a gallon of gas, well, for me gas is still $4.10 so not just a little less, but yea. |
| 17:11.42 | italorossi | hello, is it possible to use custom columns with cel odbc? asterisk 1.8.7 |
| 17:13.34 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:14.21 | leifmadsen | italorossi: check the CEL section here: http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html |
| 17:15.24 | leifmadsen | italorossi: I don't actually use CEL, but it looks like it isn't adaptive like CDR yet |
| 17:15.29 | leifmadsen | could be way off base |
| 17:15.48 | irroot | Katty the cost is similar the quality is lower big distinction they live 4/5 per room in a 3 bedroom |
| 17:16.28 | Katty | citywok: i have a 15min drive to work |
| 17:16.43 | Katty | on a side note, what's that anime with ang in it? |
| 17:16.53 | Katty | has that big blue arrow thingy on his head |
| 17:17.12 | Katty | and that girl with the braids, and her older brother.. |
| 17:17.32 | p3nguin | What does this mean? Leif Madsen changed the Link to 'This issue is the original version of this clone: ASTERISK-18630' on ASTERISK-18627. There is no ASTERISK-18630 that I can find. |
| 17:17.37 | Katty | irroot: yeah i don't think i could do that. i have 1 person in a 2 bedroom |
| 17:17.48 | italorossi | leifmadsen: Indeed, I'll use CELGenUserEvent as a workaround... |
| 17:18.13 | leifmadsen | p3nguin: that's because it was moved to another project |
| 17:18.36 | italorossi | leifmadsen: is it possible to track transfers with cdr in 1.8.7? |
| 17:18.46 | p3nguin | What does the clone part mean? I duplicated a report that I didn't know about? |
| 17:19.09 | leifmadsen | specifically, I cloned and moved an issue so it could be used by the Digium swdev team for time tracking and assignment |
| 17:19.13 | leifmadsen | p3nguin: no you didn't do anything -- I ddi it |
| 17:19.16 | irroot | italorossi leifmadsen linkedid in the cdr record is a win |
| 17:19.33 | leifmadsen | p3nguin: clone means duplicate |
| 17:19.34 | p3nguin | As I mentioned earlier, I'm JIRA-retarded. |
| 17:19.45 | leifmadsen | ya don't worry, jira is a bit retarded too sometimes |
| 17:19.56 | *** join/#asterisk godmachine-x6 (~g0d@unaffiliated/godmachine-x6) |
| 17:22.29 | italorossi | irroot: How can you store linkedid in cdr? Set(CDR(linkeid)... ? |
| 17:22.54 | irroot | italorossi no need its done internally |
| 17:23.05 | *** join/#asterisk hdiogenes (~humberto@201.76.154.132.intranet.digi.com.br) |
| 17:23.06 | irroot | the root call uniqueid = linkedid |
| 17:23.19 | irroot | then all calls after that the linkedid stays the same |
| 17:23.53 | *** join/#asterisk jwiggins (~James@gateway/tor-sasl/jwiggins) |
| 17:23.55 | *** join/#asterisk jnl- (~jnl@michael.catholic.org) |
| 17:25.15 | *** join/#asterisk singler (~singler@c.wapgw.bi.lt) |
| 17:25.27 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 17:37.11 | Kobaz | i like jirar |
| 17:38.03 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:39.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:39.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:57.53 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 17:58.22 | *** join/#asterisk netmax (~netmax@is.linux-administrator.com) |
| 17:58.43 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 18:03.32 | *** join/#asterisk didnot (~didnot@unaffiliated/didnot) |
| 18:05.28 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:07.20 | dijib | p3nguin, im about to do an asterisk rebuild. its going to be done on CentOS5 as 6 doesnt have all dependencies i need atm. would you suggest anything? |
| 18:07.37 | *** join/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:07.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 18:08.07 | p3nguin | dijib: Get the AsteriskNOW ISO. Install it. Choose the NO GUI OPTION. |
| 18:08.38 | dijib | i can manage it just the same as i do now? |
| 18:08.47 | p3nguin | How do you manage it? |
| 18:08.53 | dijib | ssh |
| 18:09.05 | p3nguin | Yes. Using the NO GUI option give you that ability. |
| 18:09.13 | dijib | not sure it works on this old laptop im using but ill try. |
| 18:09.15 | dijib | k thanks. |
| 18:10.05 | Qwell | If CentOS works, AsteriskNOW will work. |
| 18:11.11 | p3nguin | ...Considering AsteriskNOW is CentOS, and all. |
| 18:17.55 | Micc | 1.8 seems to treat domains differently. I don't see anything in the register packet that has the ip address, yet asterisk says registration failed for 'ip:port' - not a local domain. The ip is the external ip of the phone registering. Why would I have to put every IP address of each customer in my sip.conf. |
| 18:18.33 | Micc | is there some setting thats causing it to take the ip that the register is coming from and use it instead of whats in the register header? |
| 18:19.04 | Micc | wait, nevermind. I think I'm reading this wrong. |
| 18:19.34 | Micc | yeah, sorry disregard. |
| 18:21.10 | *** join/#asterisk jkroon (~jkroon@41.55.230.24) |
| 18:23.03 | irroot | jkroon yo yo |
| 18:25.54 | irroot | reality tv idea 1 biscuit 3 kids ... weakest link come amazing race come survivor .... |
| 18:26.04 | irroot | its quite fun in a sadistic way |
| 18:26.41 | *** join/#asterisk jkroon (~jkroon@197.171.171.167) |
| 18:32.50 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v008-150.mobile.uci.edu) |
| 18:32.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 18:43.54 | *** join/#asterisk jkroon (~jkroon@41.52.230.203) |
| 18:52.08 | Katty | dances with Qwell |
| 18:52.25 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053206.ics.uci.edu) |
| 18:52.25 | Qwell | trips over his feet repeatedly |
| 18:55.58 | jaytee | "One of my legs is longer than the other and both of my feets too long, got no natural rhythm.....I'm a dancing fool....." |
| 18:57.00 | *** join/#asterisk miztic (~miztic@75-149-203-105-Illinois.hfc.comcastbusiness.net) |
| 19:01.14 | *** join/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2) |
| 19:02.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:11.27 | hovel | fears deprecation |
| 19:11.42 | p3nguin | totally read that wrong. |
| 19:12.11 | jaytee | at first I thought it said "bowel fears defecation" |
| 19:12.12 | p3nguin | I thought you were talking about coprophobia. |
| 19:12.50 | hovel | this is a constipation proclamation |
| 19:13.12 | p3nguin | Or maybe rhypophobia. |
| 19:14.56 | p3nguin | Nope, Google tells me I was right the first time. |
| 19:15.16 | hovel | anyway thanks |
| 19:16.06 | *** join/#asterisk NathanWheeler (4016ebcd@gateway/web/freenode/ip.64.22.235.205) |
| 19:16.10 | *** join/#asterisk linuxplatform (~centoslin@88.87.48.115) |
| 19:16.22 | *** join/#asterisk timahvo1 (~rogue@41.223.57.74) |
| 19:17.47 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 19:21.09 | *** join/#asterisk gogasca (~Adium@nat/cisco/x-sndxmmfaujwwzcsr) |
| 19:25.01 | NathanWheeler | I just set up a trixbox server to handle a handful of polycom phones, and I have a trunk provided by nexVortex. Outbound calls work fine, but inbound calls I receive a "The number you have called is not in service" message. Obviously, this has something to do with my inbound routes, but I'm not sure where to begin troubleshooting this |
| 19:25.15 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 19:25.57 | NathanWheeler | where can I find more detailed logs in trixbox other than just the reports? |
| 19:26.10 | Naikrovek | NathanWheeler: trixbox is a mistake. /var/log/asterisk/full |
| 19:26.16 | [TK]D-Fender | Asterisk CLI will confirm where the call lands |
| 19:26.30 | [TK]D-Fender | Always use live CLI, not logs. |
| 19:26.34 | Naikrovek | look to see that the caller ID sent to you matches what you ahve set up for inbound routes |
| 19:26.44 | Naikrovek | and what [TK]D-Fender said |
| 19:27.05 | Naikrovek | i had to add a +1 to all my "inbound routes" recently when things broke. |
| 19:27.14 | Naikrovek | my provider just decided to change things without notice... |
| 19:28.00 | jaytee | I just stripped the +1 and passed the call on using Goto |
| 19:28.40 | p3nguin | If they send to the extension that is your phone number, then suddenly add +1 to the extension they are calling, that breaks things. |
| 19:29.05 | p3nguin | First you have to know they have done it. |
| 19:29.43 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 19:29.45 | [TK]D-Fender | Step 1 : really look at the actually call ans see what is coming in. |
| 19:29.50 | NathanWheeler | just out of curiousity why is trixbox a mistake? We were using a digium box and it exploded, my boss told me to set up 3CX (on Winblows... never ever ever again) and now we're going to trixbox... |
| 19:30.08 | [TK]D-Fender | So go log into the Asterisk CLI, place a call and pastebin the complete output from beginning to end |
| 19:30.11 | [TK]D-Fender | ~pb |
| 19:30.11 | 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. |
| 19:30.25 | jaytee | love that puke green color of Trixbox. |
| 19:30.26 | p3nguin | You know what they say about closed-source patches to our open-source project... |
| 19:30.55 | [TK]D-Fender | NathanWheeler, Trixbox runs a forked version of FreePBX, and I believe a modded version of Asterisk as well. Picture (no warranty support because of mods) |
| 19:31.13 | [TK]D-Fender | NathanWheeler, And they are dumping their free product as well last I heard |
| 19:31.51 | NathanWheeler | ah, that makes sense... |
| 19:32.04 | *** join/#asterisk pdtpatrick (~pdtpdt@mainstwan.farheap.com) |
| 19:32.08 | jaytee | What do you call a gay man with no sense of fashion? Kerry Garrison :-) |
| 19:32.19 | pdtpatrick | Question .. 1.8 does not understand insecure mode = very ? |
| 19:32.20 | pdtpatrick | [Sep 26 12:31:30] WARNING[4309]: chan_sip.c:25897 set_insecure_flags: Unknown insecure mode 'very' on line 1363 |
| 19:32.26 | p3nguin | nope |
| 19:32.49 | jaytee | insecure=port,invite |
| 19:32.56 | p3nguin | If you need to set insecure modes, you'll use insecure=port; insecure=invite; or insecure=port,invite. |
| 19:32.57 | jaytee | very was deprecated in 1.4 |
| 19:33.33 | Katty | weeeeeeeeeee |
| 19:33.53 | jaytee | ? caffiene buzz? |
| 19:33.56 | pdtpatrick | thanks |
| 19:34.04 | Katty | no that's the sound of happy |
| 19:34.05 | *** join/#asterisk moos3 (~rgenthner@cpe-76-178-240-227.maine.res.rr.com) |
| 19:34.18 | Katty | jaytee: are you going to join the xmas card exchange this year? |
| 19:34.29 | jaytee | I thought there was a yip at the beginning of the sound of happy/ |
| 19:34.38 | jaytee | s///? |
| 19:34.56 | jaytee | xmas? is that one of those pagan holidays? |
| 19:35.00 | Katty | even if you're not joining the xmas card exchange, i want to send you a card. |
| 19:35.05 | jaytee | ok |
| 19:35.26 | jaytee | I haven't even ordered my Cthulu cards yet for 2011 |
| 19:35.54 | Katty | that's ok |
| 19:36.14 | Katty | a cthulu card will be exciting! |
| 19:36.41 | [TK]D-Fender | jaytee, I'm waiting for 2012's : http://img.chan4chan.com/img/2010-01-05/cthulhu4prez.jpg |
| 19:36.59 | _Corey_ | nice |
| 19:37.17 | Katty | lol how cute |
| 19:37.28 | Katty | i want to knit a cthulhu |
| 19:38.14 | p3nguin | Is that word able to be pronounced? |
| 19:38.28 | jaytee | Katty, one of my friends knitted a cthulu |
| 19:38.29 | pdtpatrick | Question .. has anyone used JabberReceive on asterisk 1.6 ? |
| 19:38.35 | jaytee | she knits all the time |
| 19:38.40 | _Corey_ | I'd go with the South park pronunciation.. |
| 19:38.42 | Katty | awesome. |
| 19:38.45 | Katty | i'm still working on my tardis |
| 19:38.45 | pdtpatrick | for some reason it does not know about that function .. JabberSend works fine |
| 19:38.49 | jaytee | she knitted me a Jayne Cobb hat from Firefly |
| 19:38.52 | Katty | maybe cthulu next |
| 19:39.26 | Katty | lol nice, is it orange? |
| 19:39.57 | jaytee | Katty, it looks just like the hat in the episode "The Message", orange and yellow. |
| 19:40.07 | jaytee | "Pretty cunning, doncha think?" |
| 19:40.41 | jaytee | "What'd y'all order a dead guy for, anyways?" |
| 19:41.02 | d_preston215 | dahdi_dummy isn't needed anymore in any version of DAHDI, right? |
| 19:41.29 | p3nguin | There is no dahdi_dummy in recent versions. |
| 19:41.36 | p3nguin | So it's not a matter of need. |
| 19:41.53 | d_preston215 | I'm running DAHDI 2.3.0.1 for compatibility issues. |
| 19:42.34 | p3nguin | I don't feel like looking for which version the change happened. If you have dahdi_dummy, use it. If you don't have it, just use the regular dahdi module. |
| 19:42.40 | NathanWheeler | ok, here's my pastebin of an incoming call http://pastebin.com/d6JC1qNj |
| 19:42.49 | d_preston215 | Ok. |
| 19:42.53 | jaytee | so if you want to run MeetMe you need an earlier version? or does it still supply timing? |
| 19:44.10 | [TK]D-Fender | NathanWheeler, Your call is langing in [from-sip-external] which looks like you did not set the context to from-trunk as is the norm |
| 19:44.14 | p3nguin | nathanwheeler: What does the GotoIf at from-sip-external s,1 do? |
| 19:44.38 | p3nguin | jaytee: The recent versions of dahdi have timing in the dahdi module, and dahdi_dummy is gone. |
| 19:45.10 | [TK]D-Fender | landing* |
| 19:45.43 | NathanWheeler | [TK]D, it's set to from-trunk |
| 19:46.04 | jaytee | p3nguin, thanks. I'd seen some other posts earlier and was wondering what was up. |
| 19:46.05 | p3nguin | Good old FreePBX troubleshooting. Gotta love it. |
| 19:46.20 | [TK]D-Fender | NathanWheeler, "sip set debug on". Enable this, and then pastebin another call. |
| 19:46.35 | NathanWheeler | p3nguin, I have no idea...? |
| 19:46.56 | p3nguin | I don't understand your question. |
| 19:49.11 | p3nguin | If you're going to ask for support in #asterisk, you'd better learn how to answer questions about your asterisk. |
| 19:50.08 | p3nguin | If you can't answer questions about it because you use FreePBX/Trixbox/Elastix/other, then you're obviously in the wrong place. |
| 19:52.54 | pdtpatrick | Question .. is jabberreceive named something else in 1.6 ? [Sep 26 12:51:51] WARNING[10779]: pbx.c:3680 pbx_extension_helper: No application 'jabberreceive' for extension (ptjabber, s, 5) |
| 19:52.58 | NathanWheeler | new pastebin: http://pastebin.com/AcdvyB0M (tried somewhat to sanitize it, but I got bored :P( |
| 19:53.00 | pdtpatrick | i keep getting that warning |
| 19:53.10 | pdtpatrick | i've tried JABBER_RECEIVE .. JabberReceive |
| 19:53.13 | pdtpatrick | none of which works |
| 19:53.16 | p3nguin | jonathanrose: Putting in that ast_verb stuff isn't changing the messages that res_jabber gives on the cli. |
| 19:53.20 | *** join/#asterisk azv4 (~azv4@host-66-202-7-218.pit.choiceone.net) |
| 19:54.05 | JonathanRose | Hmmm, maybe the segfault is happening before the verb statement is made. |
| 19:54.25 | p3nguin | There is no visible segfault. |
| 19:54.37 | JonathanRose | What are we talking about then? |
| 19:54.55 | JonathanRose | I might be confusing what you are working on with another bug. |
| 19:54.57 | p3nguin | Just that "Jabber didn't seem to handshake, failed to authenticate." message every 55 seconds and never connecting to the jabber server. |
| 19:55.02 | JonathanRose | Oh |
| 19:55.10 | [TK]D-Fender | NathanWheeler, No user '14172807622' in SIP users list <- * cannot match the incoming call to your trunk |
| 19:55.24 | JonathanRose | I didn't expect that to change anything, I just wanted to know what it would show. |
| 19:55.27 | [TK]D-Fender | NathanWheeler, And it is being treated as an anonymous SIP call. |
| 19:55.34 | [TK]D-Fender | NathanWheeler, You need to fix your trunk. |
| 19:55.39 | NathanWheeler | that's my cell number... the number I'm calling from |
| 19:55.55 | p3nguin | Do I need to turn up debug level or verbose level to see any messages from ast_verb? |
| 19:56.09 | JonathanRose | verbosity needs to be whatever level was the first argument there. |
| 19:56.09 | [TK]D-Fender | NathanWheeler, Sorry... addendum followed : Found peer 'Nexvortex' for '14172807622' from 66.23.129.253:5060 |
| 19:56.21 | JonathanRose | I think it was 3? |
| 19:56.23 | [TK]D-Fender | NathanWheeler, Looking for 8883683201 in from-sip-external (domain 64.22.235.236 |
| 19:56.26 | p3nguin | Okay, I increased to 3. |
| 19:56.44 | [TK]D-Fender | NathanWheeler, Indeed your peer is either specifying the wrong context, or none at all |
| 19:56.50 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:58.15 | NathanWheeler | so I need to find out what context the provider is using? |
| 19:58.53 | [TK]D-Fender | NathanWheeler, these are your trunk settings. |
| 19:59.00 | p3nguin | jonathanrose: Comment is posted with the results. |
| 19:59.00 | [TK]D-Fender | Nothing on the provider side |
| 19:59.10 | JonathanRose | p3nguin: thanks |
| 19:59.50 | pdtpatrick | Question .. is jabberreceive named something else in 1.6 ? [Sep 26 12:51:51] WARNING[10779]: pbx.c:3680 pbx_extension_helper: No application 'jabberreceive' for extension (ptjabber, s, 5) |
| 20:00.06 | p3nguin | core show applications like jabber |
| 20:00.29 | *** join/#asterisk smash- (~smash@173-11-0-109-oregon.hfc.comcastbusiness.net) |
| 20:02.00 | p3nguin | I don't even see anything related to jabber receive. |
| 20:04.16 | blizzow | The business that set up our asterisk system just asked me if someone rebooted our asterisk server and we had a weird instance in the past where asterisk seemed to get restarted in the middle of the day. I looked at ps -aef and saw this output: |
| 20:04.17 | blizzow | asterisk 4028 18769 99 Sep20 ? 7-22:05:59 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c |
| 20:04.17 | blizzow | Nobody was on the server on September 20. Am I reading it right that Asterisk has only been running since Sep 20? Is it typical for the asterisk process to give itself a HUP or somehow get refresh it's run time? |
| 20:05.25 | p3nguin | core show uptime |
| 20:05.28 | *** join/#asterisk tamiel (~tamiel@ip-183.net-81-220-86.toulouse.rev.numericable.fr) |
| 20:08.11 | JonathanRose | p3nguin: Here's an idea that... might be a bad one. Try removing the client->timeout != 0 part of the condition. |
| 20:10.45 | JonathanRose | Actually, we should probably be talking about this in #asterisk-dev, so once you try that, tell me your results in there. |
| 20:10.53 | p3nguin | tm1000_away: I wish you'd hurry up and fix that. It is VERY annoying. |
| 20:11.25 | tm1000_away | p3nguin: yes I know :-( |
| 20:11.27 | tm1000_away | ughhhh |
| 20:11.31 | p3nguin | tm1000_away: Please. |
| 20:11.33 | tm1000_away | sorry everyone |
| 20:12.26 | *** join/#asterisk alan17532 (~d@41-134-22-10.dsl.mweb.co.za) |
| 20:13.28 | tm1000 | p3nguin: fixed.sorry |
| 20:13.34 | p3nguin | tm1000: Thank you! |
| 20:14.14 | alan17532 | I replaced my old legasy pbx for asterisk, love it, what would you guys recommend for a call accounting system, to trace employees calls? |
| 20:14.22 | Katty | ohai |
| 20:15.11 | alan17532 | Katty is that a application? |
| 20:15.16 | p3nguin | haha |
| 20:15.21 | alan17532 | lol |
| 20:15.32 | alan17532 | i think not |
| 20:15.34 | p3nguin | I thought it, but I didn't say it. ;) |
| 20:16.32 | alan17532 | i googled it and came up with a few, but just thought i would ask the experts? |
| 20:16.40 | smash- | hey |
| 20:16.46 | smash- | does anyone need a sangoma wanpipe? |
| 20:16.51 | smash- | like 150 bucks =P |
| 20:16.57 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1176002162.dsl.bell.ca) |
| 20:16.59 | smash- | its just storage box pimping atm |
| 20:16.59 | smash- | haha |
| 20:17.10 | dijib | centos5 & asterisknow dont boot. |
| 20:17.11 | [TK]D-Fender | smash-, a wanpipe what? |
| 20:17.18 | dijib | centos6 boots. |
| 20:17.19 | smash- | TDM |
| 20:17.20 | dijib | wtf |
| 20:17.30 | [TK]D-Fender | smash-, got model #'s? |
| 20:17.33 | smash- | yah |
| 20:17.49 | smash- | AFT Series MODEL AFT BASE [2005] |
| 20:17.54 | smash- | rev 2.1 |
| 20:17.55 | [TK]D-Fender | smash-, helps when people know what you're actually offering :) |
| 20:17.56 | alan17532 | looks like everybody is chatting in code |
| 20:18.01 | smash- | A102 |
| 20:18.04 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 20:18.18 | miztic | smash-, does it have hw echo canceling ? |
| 20:18.19 | alan17532 | can't beat them join them --> mISDN |
| 20:18.21 | smash- | yes |
| 20:18.36 | *** part/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 20:18.41 | [TK]D-Fender | smash-, then that would be an A102d |
| 20:18.45 | smash- | all sangoma tdm cards have them i believe |
| 20:18.54 | [TK]D-Fender | Absolutely not |
| 20:18.59 | miztic | i thought it was an option with mine |
| 20:19.06 | smash- | its 2 seperate cards. |
| 20:19.14 | smash- | the echo cancellation is on the second part i thought |
| 20:19.21 | [TK]D-Fender | ... |
| 20:19.22 | miztic | its an addon board |
| 20:19.22 | smash- | kuz its like 2 cards put together 1xPCI slot |
| 20:19.24 | [TK]D-Fender | no |
| 20:19.40 | smash- | i know it has hardware echo cancellation because i would not have purchased it if not. |
| 20:19.47 | [TK]D-Fender | smash-, sub-board, no HWEC. this is their ealier gen build |
| 20:19.58 | smash- | yes it is earlier build |
| 20:21.22 | smash- | i think ur right [TK]D-Fender, i think i did the EC in the Cisco WIK |
| 20:21.55 | smash- | idk its old, i dont use any tdm voice though so I have 0 use for it. |
| 20:22.15 | smash- | we just moved offices and i found it. |
| 20:22.26 | smash- | figured someone in here might want it for a toy. |
| 20:24.02 | [TK]D-Fender | that would be a handy backup for sure |
| 20:25.03 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:25.24 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 20:25.37 | alan17532 | I replaced my old legasy pbx for asterisk, love it, what would you guys recommend for a call accounting system, to trace employees calls? |
| 20:27.26 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:27.27 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:28.24 | [TK]D-Fender | alan17532, http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 |
| 20:28.45 | alan17532 | thank you [TK]D-Fender |
| 20:29.16 | [TK]D-Fender | checkout time, BBIAB |
| 20:37.14 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:38.48 | smash- | Fender shoot me a text if you want it |
| 20:41.53 | *** join/#asterisk DrDigi (~mmurphy@50-73-49-110-static.hfc.comcastbusiness.net) |
| 20:43.48 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 20:45.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:46.17 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:47.25 | *** join/#asterisk JasonL (~jason@216.223.114.3) |
| 20:49.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:52.22 | jaytee | wb |
| 20:55.13 | JasonL | Is there a different between DTMF behaviour from 1.6.2.9 and 1.8.7.0... Since upgrading DTMF is failing when dialed quickly. I went back to 1.6.2.9 and DTMF works fine. |
| 20:56.08 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 20:59.36 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 21:01.11 | Katty | guess who's goin home and chillaxin! |
| 21:01.21 | Katty | DIS GIRL. later gaters! |
| 21:02.08 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 21:02.58 | oldhack | after while crocodile! |
| 21:04.08 | wdoekes2 | JasonL: https://issues.asterisk.org/jira/browse/ASTERISK-18339 ? |
| 21:04.44 | *** join/#asterisk jkroon (~jkroon@dsl-241-237-12.telkomadsl.co.za) |
| 21:04.53 | *** join/#asterisk sflemming (~stefan@85.183.53.64) |
| 21:06.38 | *** join/#asterisk leftist (~dizzy@173.160.65.209) |
| 21:06.53 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:08.40 | sflemming | Hi all, I found a severe bug in the asterisk calendar integration and would like to ask if someone is out there that can reproduce the problem before submitting a bug |
| 21:10.18 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:11.52 | *** join/#asterisk inluck2 (ae747e04@gateway/web/freenode/ip.174.116.126.4) |
| 21:13.15 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 21:13.39 | inluck2 | Does Asterisk, when configured with ODBC mysql voicemail storage, support two asterisk servers accessing the same mysql database for voicemail? |
| 21:14.42 | d_preston215 | How to check to see if asterisk loaded dahdi? |
| 21:14.44 | d_preston215 | In asterisk 1.8, would I have dahdi options at the cli? |
| 21:14.46 | d_preston215 | If dahdi was loaded? |
| 21:16.16 | pabelanger | d_preston215: *CLI> module show like dahdi |
| 21:17.02 | d_preston215 | 4 modules. |
| 21:17.16 | d_preston215 | Timing Interface |
| 21:18.03 | pabelanger | Use count? |
| 21:18.08 | d_preston215 | 1 |
| 21:18.13 | wdoekes2 | inluck2: yes |
| 21:18.25 | pabelanger | so you should see chan_dahdi.so |
| 21:18.32 | inluck2 | wdoekes2: thank you |
| 21:18.36 | pabelanger | then you have chan_dahdi loaded |
| 21:18.38 | d_preston215 | I don't see chan_dahdi.so |
| 21:18.42 | d_preston215 | Crud. |
| 21:18.51 | pabelanger | so it is missing |
| 21:19.05 | inluck2 | wdoekes2: hard to find much information, wanted some sort of confirmation before I head down the long road of configuration |
| 21:19.32 | wdoekes2 | odbc voicemail is actually not that long a road.. but I see your point |
| 21:20.09 | d_preston215 | chan_dahdi.so is there, but not loaded. |
| 21:20.14 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 21:20.57 | inluck2 | wdoekes2: im just seeing everything doubled, two pbx |
| 21:21.36 | rdegges | Sup guys. |
| 21:21.48 | rdegges | I was wondering if any of you are using ``extenpatternmatchnew`` in your dialplan? |
| 21:21.53 | d_preston215 | Figured it out. |
| 21:22.08 | rdegges | And if it makes a performance difference during call execution, or only during reload? |
| 21:25.45 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1176002162.dsl.bell.ca) |
| 21:27.09 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:30.56 | *** join/#asterisk LittleFool (~LittleFoo@95.129.212.120) |
| 21:31.42 | d_preston215 | Can I use any version of libopenr2 with any version of DAHDI? |
| 21:34.15 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 21:37.20 | cusco | hi folks |
| 21:37.27 | cusco | any body using amr codec? |
| 21:56.16 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 21:56.24 | *** join/#asterisk devcoder (~leemelnyk@216.18.243.44) |
| 21:56.29 | devcoder | hey everyone, wiped out the firmware on my phone totally now on my cisco 7975 grrrr... looking for either cmterm-7975-sccp.8-5-2.zip or cmterm-7975-sip.8-5-2.zip if anyone could help |
| 22:20.10 | p3nguin | If you aren't going to download it from Cisco, you'll have to google it. |
| 22:20.31 | p3nguin | You obviously know the format of the filenames, so you're half way there. |
| 22:20.45 | p3nguin | I typically have to tell people the file names so they can search for them. |
| 22:23.30 | *** join/#asterisk corretico (~luis@200.12.40.18) |
| 22:23.39 | corretico | hi |
| 22:23.58 | corretico | i need some assitance |
| 22:24.22 | corretico | is possible to make a trunk sip between asterisk and cisco 2800 |
| 22:24.46 | [TK]D-Fender | sure |
| 22:25.14 | corretico | well, the real question is: how i can do, to tranfers a call from asterisk to this cisco (or any other) using the sip trunk |
| 22:26.33 | corretico | I see the sip trunk OK with i use sip show peers |
| 22:27.35 | corretico | sorry 4 my english!!! |
| 22:28.54 | devcoder | p2ngin, yeah been googleing for hours now, best i have found is a cop.sgn file. |
| 22:28.55 | [TK]D-Fender | dial the peer like you would any other SIP provider |
| 22:29.10 | devcoder | trying to download it waiting for cisco to get back with me |
| 22:29.18 | devcoder | just the fastest people around |
| 22:29.55 | p3nguin | The firmware files are available in multiple places online. You could have already downloaded all of them by now. |
| 22:30.29 | corretico | i create a outbound rule to transfer internal call from asterisk to this other equipment... |
| 22:31.19 | corretico | but when i press the cisco extension, the bussy/congest appers |
| 22:33.43 | *** join/#asterisk Kyosh (~whoa@pool-74-108-19-39.nycmny.fios.verizon.net) |
| 22:33.44 | [TK]D-Fender | "sip set debug on" |
| 22:33.55 | [TK]D-Fender | and then look at the actual call to see what the Cisco is saying. |
| 22:34.03 | [TK]D-Fender | ~pb |
| 22:34.03 | 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. |
| 22:34.13 | [TK]D-Fender | and pastebin it so we can also see and help you understand it |
| 22:37.25 | nobodyshome | hey p3nguin around? |
| 22:37.30 | p3nguin | yes |
| 22:37.41 | nobodyshome | fresh install of c-os6 |
| 22:37.48 | nobodyshome | cant lspci? |
| 22:37.55 | p3nguin | Why not? |
| 22:38.06 | nobodyshome | nothing in /etc/sysconfig/networking/devices ? |
| 22:38.13 | p3nguin | /usr/sbin/lspci |
| 22:38.51 | p3nguin | If you don't have it, you must need to install the pciutils. |
| 22:39.02 | corretico | http://pastebin.com/N3sn3Eur |
| 22:39.23 | corretico | the cisco 2800 have one extension... 2899 |
| 22:39.26 | nobodyshome | i think thats the case. |
| 22:39.38 | nobodyshome | then i need to resolve my netowkring issues |
| 22:40.00 | corretico | in my outbound rule include this extension and use the sip trunk to the cisco |
| 22:40.00 | p3nguin | What does "whereis lspci" tell you? |
| 22:40.13 | nobodyshome | where do i statically set eth0 /etc/sysconfig/networking/ ...? |
| 22:40.34 | nobodyshome | lspci: |
| 22:40.39 | nobodyshome | nowhere |
| 22:40.46 | nobodyshome | didnt know whereis |
| 22:40.55 | p3nguin | And /usr/sbin/lspci said not found? |
| 22:41.07 | nobodyshome | yep. |
| 22:41.10 | nobodyshome | 6-minimal |
| 22:41.14 | p3nguin | I guess you need to install some things. |
| 22:41.36 | nobodyshome | i think i need to fix this networking issue |
| 22:41.44 | [TK]D-Fender | corretico: You have not enabled SIP debug like I told you was necessary for this. Please do so and pastebin a new call |
| 22:41.46 | nobodyshome | cant get away from the computer.... |
| 22:41.49 | p3nguin | Maybe you need to install networking stuff. |
| 22:42.00 | nobodyshome | how with no network? |
| 22:42.19 | nobodyshome | and centos5 wont run on this thing. which is what asterisknow uses......whiich is less |
| 22:42.22 | nobodyshome | or less |
| 22:42.22 | p3nguin | from the CD, of course |
| 22:42.37 | nobodyshome | its a 300mb iso |
| 22:42.43 | nobodyshome | i dont even think it has it |
| 22:43.16 | p3nguin | What was your reason for not using AsteriskNOW? Wanted to have to force it to work rather than just installing it and using it right away? |
| 22:43.30 | nobodyshome | it wouldnt boot |
| 22:43.39 | nobodyshome | nor cEntos6 |
| 22:43.41 | nobodyshome | i mean 5 |
| 22:43.42 | Kyosh | using trixbox 2.8 (asterisk 1.6), i have a sangoma A200D 8 port POTS card. the first 4 ports i want to use solely for incoming calls while the last 4, for outbound. i installed the sangoma drivers, ran the configs and everything is almost fine. the system sees the entire card, all 8 ports as 'Zap/g0'. I was hoping to separate them as 'Zap/g0-1(through 4)' and 'Zap/g1-1(through 4)'. Is this possible and if so, how? |
| 22:43.57 | p3nguin | Did you verify your checksum of the iso image before you burned the CD? |
| 22:44.18 | nobodyshome | nope. you think its a corrupted extraction? |
| 22:44.25 | p3nguin | There's no extraction. |
| 22:44.26 | citywok | Kyosh: yes, in the zaptel/dahdi config you can create groups of channels |
| 22:44.31 | nobodyshome | did say it had any failiers |
| 22:44.35 | *** join/#asterisk leftist (~dizzy@50-10-91-159.gar.clearwire-wmx.net) |
| 22:44.39 | p3nguin | You just download hte image, verify it, and burn it. |
| 22:44.42 | nobodyshome | yikes. you need a perfect disk then? |
| 22:44.48 | p3nguin | Well yeah. |
| 22:44.53 | Kyosh | citywok, anything more specific? that doesn't give me much to go on. |
| 22:45.03 | p3nguin | Verify your image. |
| 22:45.03 | nobodyshome | tcp isnt reliable enough? |
| 22:45.08 | nobodyshome | i rarely md5 |
| 22:45.09 | citywok | i haven't messed with the configs in a very long time, but the docs should explain it. |
| 22:45.20 | citywok | i run pure sip now, no more dealing with zaptel |
| 22:45.25 | p3nguin | Check your md5sum. |
| 22:45.39 | nobodyshome | ive had the image work before, if anything its the shady dvdrw i grabbed |
| 22:45.47 | Kyosh | citywok, sorry to say, i've been through the docs and 3 reinstalls after botching everything up. i would need something a bit more specific to achieve my goals :( |
| 22:46.03 | navaismo | that was very specific |
| 22:46.35 | citywok | Kyosh: i'm not going to write your config for you, i told you where to look to do it, and the keyword group should be a pretty obvious hint. http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 22:46.43 | p3nguin | You used it before and still you're trying to install another OS? |
| 22:47.01 | Kyosh | citywok, there is zapata.conf and chan_dahdi.conf and neither give me much insight |
| 22:47.12 | citywok | use a flash drive, it's much easier :P -- although i haven't tried it with asterisknow i do it for debian all the time |
| 22:47.31 | p3nguin | I'm not so sure the AsteriskNOW image is built for USB. |
| 22:47.34 | [TK]D-Fender | <PROTECTED> |
| 22:47.41 | [TK]D-Fender | <PROTECTED> |
| 22:47.42 | citywok | chan_dahdi is for dahdi, not zaptel. |
| 22:47.43 | [TK]D-Fender | <PROTECTED> |
| 22:47.47 | [TK]D-Fender | <PROTECTED> |
| 22:47.58 | citywok | [TK]D-Fender: mean, you should have made him open the wiki doc which showed that in it. lol. |
| 22:48.06 | Kyosh | i never asked for you to write anything, only if anyone had any ideas to help pout, maybe point to a resource online, its not like voip-info.org is up to date and maybe someone recently worked through this matter themselves |
| 22:48.31 | citywok | Kyosh: the doc i gave you is up to date as far as zaptel goes, considering zaptel has been renamed to dahdi... |
| 22:48.38 | [TK]D-Fender | Kyosh: core config for this hasn't changed in a decade |
| 22:48.52 | [TK]D-Fender | Kyosh: ... and I just handed it to you |
| 22:48.53 | citywok | and it had the answers you needed, if you combined looking at the doc with the directions i gave you shuold be able to connect the dots |
| 22:49.18 | [TK]D-Fender | Kyosh: make settings. asign channels. change some settings. Assign more channels |
| 22:49.19 | citywok | or you could use [TK]D-Fender's answer which he so gratefully wrote for you |
| 22:49.27 | Kyosh | i read that ZAP was renamed to DAHDI for trademark reasons |
| 22:49.34 | Kyosh | hence the reason i was lead to DAHDI |
| 22:49.52 | p3nguin | s/hence the reason/hence/ |
| 22:50.00 | [TK]D-Fender | Correct. the contents of chan_dahdi.conf are 99% zapata.conf <- |
| 22:50.04 | p3nguin | Can't stand that. |
| 22:50.22 | [TK]D-Fender | if it was renamed, then that alone does not imply that the actual contents are differen, just the brand on ht e surface |
| 22:51.20 | citywok | yea, in all actuality it makes it easier :P |
| 22:52.22 | Kyosh | sadly i feel that its always a struggle to get help in here. yes you are all knowledgable but only after having to put up a defensive shield against the onslaught of initial pokes and stabs about being a newb or lazy or something like that, then maybe i get put in the right direction. its almost like efnet in a sad way. no insult intended, just the way it seems much of the time. |
| 22:52.27 | corretico | http://pastebin.com/grHTYQet |
| 22:52.40 | corretico | this is the pastebin with debug mode on |
| 22:52.59 | citywok | Kyosh: when you ask a question and you get an answer, especially if it includes a link you should read the document |
| 22:53.03 | p3nguin | I guess I missed where anyone said anyone else was lazy. |
| 22:53.37 | citywok | p3nguin: i said i wasn't going to write the config for him (which tk then did). i guess he took that as me calling him lazy. |
| 22:54.11 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 22:54.24 | p3nguin | I tell people all the time that I'm not going to do something for them. |
| 22:54.41 | citywok | Yep |
| 22:54.51 | p3nguin | Doesn't mean I'm saying they are lazy, just saying do it yourself with the tools I give you. |
| 22:55.09 | citywok | and i thought i gave pretty good direction on where he could do what he wanted |
| 22:55.33 | citywok | but apparently that means he needs a defensive shield and we're calling him lazy :-\ |
| 22:55.36 | citywok | oh well. so how are you sir p3nguin? |
| 22:57.03 | p3nguin | blah |
| 22:57.08 | moy | d_preston215: pretty much, yes |
| 22:57.10 | citywok | that good eh? |
| 22:57.12 | Kyosh | <citywok> Kyosh: yes, in the zaptel/dahdi config you can create groups of channels, kinda vague as i already explained that i tried to make changes there but ended up screwing the install therefore i had 3 installs later to try again and this time, ask someone who may be able to tell me a bit more than what you said, which is what i guessed, but exactly "where" and "what" are the important parts |
| 22:57.49 | [TK]D-Fender | corretico: SIP debug is still not enabled in there |
| 22:57.54 | citywok | Kyosh: there are only a couple config files, only one of which defines groups. i'd suggest following the wiki article that explains how to do it. |
| 22:57.56 | Kyosh | im no idiot, but i dont live this stuff and every year or so i come here to the people who know best, but its becoming increasingly hostile in here |
| 22:58.31 | citywok | Kyosh: we never called you an idiot. we gave you docs, tk wrote the actual config for you. what more do you want? |
| 22:58.31 | p3nguin | Try getting help with apache httpd over in #httpd... you'll LOVE it here! |
| 22:58.36 | *** join/#asterisk f2knight (~ben@c-76-115-44-207.hsd1.or.comcast.net) |
| 22:58.38 | Kyosh | as [TK]D was pretty specific and led me in a direction i will certainly try. thank you for the assist |
| 22:59.14 | Kyosh | p3nguin, i have an apache guy to mess with for that. sadly im the go to person in my company for voip and i wish i weren't since im a cisco person |
| 22:59.28 | [TK]D-Fender | citywok: defacto answer "Would you like fires with that, sir?" :) |
| 22:59.45 | [TK]D-Fender | fries even :p |
| 22:59.56 | citywok | i was like wtf fires? lol. |
| 23:00.11 | citywok | that makes a lot more sense -- my brain didn't fix that typo for me. |
| 23:02.39 | Kyosh | ok sadly this is trixbox and apparently they screwed with which file handles ZAP cause inside /etc/asterisk and /etc there is NO zapata.conf, just a zapata_additional.conf and it's empty |
| 23:02.44 | beek | Good evening [TK]D-Fender |
| 23:04.53 | Kyosh | and citywok, im sure i could use that config he wrote, if i knew where to put it. up my ass wont fix the pbx tho :p |
| 23:05.30 | corretico | <[TK]D-Fender> http://pastebin.com/8it88fPG |
| 23:05.58 | citywok | Kyosh: if you'd like the awesome support you aren't getting here the #trixbox channel should be able to help (i've never been able to get help in there) |
| 23:06.09 | p3nguin | Third time's a charm, I guess. |
| 23:06.14 | citywok | Kyosh: if a config doesn't exist... and you need it... make it... |
| 23:06.49 | corretico | i think that the call is not on the cisco sip trunk |
| 23:07.00 | citywok | Kyosh: seriously, have you not read this document? it explains every question you've asked... http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 23:07.13 | Kyosh | i went over that page 5 times |
| 23:07.17 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 23:07.36 | Kyosh | it doesnt explain breaking up the trunk ports in a way thats clicking in my dense head |
| 23:07.52 | Kyosh | its 1, 8 port sangoma POTS |
| 23:08.01 | citywok | really? because tk gave you the code |
| 23:08.03 | Kyosh | i know regardless sangoma or not |
| 23:08.11 | citywok | and in the example config files in that link, it shows code that looks VERY similar... |
| 23:08.19 | citywok | 1 + 1 = 3 |
| 23:08.21 | Kyosh | to put in zapata.conf which after rebooting the system its still not working |
| 23:08.54 | Kyosh | and as i said, zapata.conf did not exist before i created it. |
| 23:09.15 | Kyosh | so its possible that since its shitbox, its looking for a different file |
| 23:09.53 | nobodyshome | k all good now |
| 23:09.56 | nobodyshome | brain fart |
| 23:10.06 | nobodyshome | installing sshd |
| 23:10.10 | p3nguin | I doubt it, but I don't know about that since it's not asterisk. |
| 23:10.22 | Kyosh | oh its asterisk, on some nasty roids |
| 23:10.36 | p3nguin | By asterisk, I mean just asterisk. |
| 23:11.00 | Kyosh | very true |
| 23:11.36 | [TK]D-Fender | corretico: that only has SIP debug, and not even the entire call. we need the full CLI (dialplan). We need both |
| 23:12.31 | citywok | Kyosh: i assumed you already had it working and just wanted to create the 4/4 grouping |
| 23:12.38 | citywok | now it sounds like it doesn't work at all |
| 23:12.45 | [TK]D-Fender | Kyosh: if you're running DAHDI, then kill the zaptel & zapata configs. You risk clashing the two |
| 23:13.07 | p3nguin | To find out what file my dahdi is looking for, I use something like ''strings /usr/lib/asterisk/modules/chan_dahdi.so |grep "\.conf"'' |
| 23:13.13 | p3nguin | I'd assume the same would work for zap. |
| 23:13.46 | nobodyshome | k what asterisk packages do i need? asterisk asterisk-voicemail asterisk-core-sounds asterisk-extra?-sounds asterisk-meetme asterisk-dahdi ? |
| 23:13.47 | Kyosh | citywok, i have the sangoma working fine, i only wanted to divide it into 2 groups of 4 ports each. |
| 23:14.12 | p3nguin | nobodyshome: Using AsteriskNOW? |
| 23:14.32 | dijib | no, centos6 |
| 23:14.36 | p3nguin | dijib: Did you configure the Digium repositories? |
| 23:14.39 | dijib | only thing that runs on this EVO |
| 23:14.41 | citywok | Kyosh: then find the part of your config that makes it work in the first place, and defines group0 |
| 23:14.47 | citywok | Kyosh: that's where you need to define both groups |
| 23:14.51 | dijib | not yet ... waiting for a yum update to finish |
| 23:14.57 | p3nguin | dijib: If CentOS 6 will run on it, AsteriskNOW will run on it. |
| 23:15.04 | dijib | it wont. |
| 23:15.09 | p3nguin | Yes it will. |
| 23:15.13 | dijib | the cd doesnt boot, nor does centos5 |
| 23:15.17 | dijib | no it wont. |
| 23:15.29 | p3nguin | I'm not going to argue with you over it. |
| 23:15.31 | dijib | nothing runs on this thing. |
| 23:15.44 | citywok | dijib: then maybe you shouldn't use it to run your pbx |
| 23:15.49 | dijib | i tries asterisk now with 1.7.1 |
| 23:16.09 | dijib | its got a battery backup built into it. |
| 23:16.30 | dijib | i ghost the hdd |
| 23:17.00 | citywok | and your point? |
| 23:17.09 | citywok | if the hardware is unstable it's not going to make a good pbx |
| 23:18.09 | Kyosh | http://pastebin.com/aWWrX1pM |
| 23:18.26 | Kyosh | i only changedd the group from 0 to 1 in channels 5 through 8 and the pbx took a dump on me |
| 23:18.43 | citywok | Kyosh: where in the example do you see group = 0 being defined 8 times? |
| 23:18.51 | Kyosh | so i changed it back and all was well in Oz |
| 23:18.56 | citywok | my guess is your group = 0 is comprised of one chanenl. being 8. |
| 23:19.16 | Kyosh | the paste i just showed you is my chan_dahdi.conf |
| 23:19.42 | citywok | no kidding |
| 23:20.09 | Kyosh | no really :-p |
| 23:20.20 | citywok | about 3/4 of the way down this page lies the answer you are looking for http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 23:20.28 | citywok | which is the same answer that tk gave you earlier |
| 23:20.28 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:20.58 | citywok | it's time for me to go drink, so gl. you have the tools needed to succeed. |
| 23:21.09 | Kyosh | oh yes i know and i tried that yesterday and it killed the pbx |
| 23:21.25 | navaismo | just change the number group for desired channels, now all is set to 0 |
| 23:21.58 | Kyosh | imma gonna install a previous version of shitbox and see if it still has the same problem. i hate shitbox for this very reason |
| 23:22.16 | p3nguin | If you're going to go to that trouble, why wouldn't you install the current asterisk? |
| 23:22.35 | citywok | Kyosh: the problem is the config... not the install. rtfm, do what tk said. |
| 23:24.12 | citywok | if you need somebody to do it for you there are lots of people that would be happy to consult for you (i'll do it for 125/hr) |
| 23:25.11 | Kyosh | i have a bunch of asterisk boxes i am test clustering for fun to see how well it works and it does. but my father in law's company wants trixbox cause of all the open source pbx' out there, he was told its the easiest to work with. some truth to some degree. |
| 23:25.51 | Kyosh | ive made my asterisk do flips and really nice things, but again, throwing the whole gui thing in there kinda throws me off, especially with their custom scripts |
| 23:26.09 | p3nguin | I'm not sure what Trixbox has to do with open source. |
| 23:26.16 | Kyosh | im just hoping i dont have to reinstall again during the week |
| 23:26.17 | citywok | that's why this channel hates trixbox... b/c it sucks. |
| 23:26.32 | citywok | and we're all really confused why you need to keep reinstalling it. |
| 23:26.37 | Kyosh | trixbox is based on asterisk, thats the only thing that connects the 2 |
| 23:26.46 | p3nguin | I know what trixbox is. |
| 23:27.03 | Kyosh | because its only 5 mins to reinstall instead of figuring what got screwed when making a change that prevents the system from working at all |
| 23:27.06 | citywok | lol trixbox IS asterisk, and we all know that. it's just a gui to asterisk. and mostly it's just freepbx that is usfeul. |
| 23:27.19 | p3nguin | It's not "just a gui to asterisk." |
| 23:27.24 | p3nguin | ~trixbox |
| 23:27.24 | infobot | rumour has it, trixbox is unable to be supported here. It is a closed source distribution of Asterisk which its users don't have access to, making it difficult to support. Trying joining #tribox and asking your questions there. |
| 23:27.37 | citywok | yea, it's a distribution. i always think of trixbox as freepbx. |
| 23:27.39 | p3nguin | CLOSED SOURCE SHIT on top of a perfectly good asterisk. |
| 23:27.53 | p3nguin | Trixbox uses FreePBX just like it uses Asterisk. |
| 23:27.54 | Kyosh | AAH was better i think |
| 23:27.57 | citywok | oh, do they actually mod the asterisk source themselves / fork off of it? |
| 23:28.11 | p3nguin | They patch asterisk, they patch freepbx. |
| 23:28.15 | p3nguin | And they don't share. |
| 23:28.31 | citywok | interesting, i didn't know that. i've installed asterisk+freepbx for small businesses |
| 23:28.44 | p3nguin | That's better than trixbox. |
| 23:28.45 | Kyosh | they have custom scripts, that i know for sure, but not much else about it. as i said, wasnt my choice. but the pappy inlaw wants it so he can mess around with the gui |
| 23:28.52 | citywok | although at my call center we use vanilla asterisk & lots of custom work |
| 23:29.03 | citywok | but for an 8 person law firm freepbx is much easier :P |
| 23:29.12 | citywok | even if it's annoying as hell to work with |
| 23:29.17 | Kyosh | 8 person law firm? |
| 23:29.17 | p3nguin | I'd recommend AsteriskNOX if you insist on FreePBX; it's certainly better than Trixbox. |
| 23:29.30 | p3nguin | If that's all he cares about, he'll love it. |
| 23:29.47 | citywok | Kyosh: yes, 8 person law firm |
| 23:29.53 | p3nguin | AsteriskNOW, even. |
| 23:29.56 | citywok | or 25 person not for profit |
| 23:30.06 | Kyosh | p3nguin, again, he insisted that he wanted asterisk |
| 23:30.16 | citywok | Kyosh: all of them are asterisk |
| 23:30.20 | p3nguin | So instead you gave him Trixbox. Way to go. |
| 23:30.32 | citywok | asterisknow being rolled and distributed by digium, the company that "makes" asterisk so to speak |
| 23:30.53 | p3nguin | I'll say it again: If all he cares about is asterisk and a fancy gui, AsteriskNOW is better than Trixbox. |
| 23:30.53 | citywok | if you have questions about asterisknow i believe qwell is the one that packages it? |
| 23:30.58 | dijib | is this a good enough list for my asterisk package needs ? http://downloads.openwrt.org/backfire/10.03.1-rc5/brcm-2.4/packages/ |
| 23:31.09 | corretico | <[TK]D-Fender>i'm using trixbox. i'm not sure where the dialplan is locate |
| 23:31.15 | Kyosh | he wants it, he gets it. too much to stress over his ass being a stubborn fool |
| 23:31.22 | dijib | or are a lot handles but asteris18-core or something |
| 23:31.30 | dijib | handled |
| 23:31.30 | p3nguin | corretico: I doubt he said that, so don't quote him as saying that. |
| 23:31.54 | citywok | only if it was satirical |
| 23:32.03 | p3nguin | I'll say it again again: If all he cares about is asterisk and a fancy gui, AsteriskNOW is better than Trixbox. You should give him what he asked for instead of Trixbox. |
| 23:32.04 | Kyosh | think of an old-school hong kong style chinese business man who is not questioned, just given what he asks for. |
| 23:32.23 | citywok | he asked for asterisk. he didn't ask for trixbox. |
| 23:32.29 | Kyosh | no, he cares about trixbox cause he read it in a fukin magazine. i never mentioned asterisk to him or stated that he mentioned it |
| 23:32.30 | p3nguin | EXACTLY |
| 23:32.35 | p3nguin | Trixbox IS NOT Asterisk. |
| 23:32.36 | citywok | he's too stupid to know what he's getting anyways |
| 23:32.44 | p3nguin | It's fucking trixbox for crying out loud. |
| 23:32.45 | Kyosh | he asked for trixbox, not asterisk |
| 23:32.46 | citywok | you could give him mud and tell him it was asterisk |
| 23:32.57 | p3nguin | (1830.08) <Kyosh> p3nguin, again, he insisted that he wanted asterisk |
| 23:33.02 | p3nguin | Make up your mind. |
| 23:33.04 | [TK]D-Fender | corretico: I'm talking about your CLI output |
| 23:33.06 | pabelanger | language please |
| 23:33.12 | Kyosh | oh no, i threw asterisknow on a server for him and he saw it didnt have the trixbox logo and flipped on me |
| 23:33.31 | citywok | find a new job :) |
| 23:33.35 | [TK]D-Fender | corretico: in the first 2 we could see what was being exectued. But no SIP debug. When you enabled that you seemed to have gone and lowered the verbose level so it hid the other half. |
| 23:33.37 | citywok | working for stupid people sucks lol |
| 23:33.41 | p3nguin | haha |
| 23:33.46 | p3nguin | AMEN, BROTHER! |
| 23:33.51 | [TK]D-Fender | corretico: thus you are only showing half at a time. we need the whole thing at once. |
| 23:34.01 | Kyosh | [19:20] <Kyosh> i have a bunch of asterisk boxes i am test clustering for fun to see how well it works and it does. but my father in law's company wants trixbox cause of all the open source pbx' out there, he was told its the easiest to work with. |
| 23:34.04 | citywok | okay FINALLY going to go get that beer, and laugh about your boss. |
| 23:34.20 | citywok | Kyosh: clearly you've proven it isn't easy to work with, now haven't you? |
| 23:34.30 | p3nguin | I didn't understand that statement hte first time you said it. |
| 23:34.33 | Kyosh | i rather work with trixbox than my father in law |
| 23:34.42 | p3nguin | father in law's company wants trixbox cause of all the open source pbx out there... does not compute. |
| 23:34.52 | Kyosh | his words, not mine |
| 23:34.53 | citywok | lol no kidding |
| 23:34.57 | citywok | new job :P |
| 23:34.58 | p3nguin | Does not compute. |
| 23:35.18 | p3nguin | I'd like to have Windows because of all the open source OSs out there. |
| 23:35.22 | Kyosh | citywok, im a consultant for him, not his bitch employee. he fired the bitch cause the bitch kept using pirated software |
| 23:35.34 | Kyosh | well he thinks free == OSS |
| 23:35.37 | Kyosh | dunno why |
| 23:35.48 | p3nguin | You should have informed him otherwise. |
| 23:35.57 | rdegges | Anyone know what app_dahdiras.so (DAHDI ISDN Remote Access Server) is for? :o |
| 23:36.09 | p3nguin | And told him at the first mention of wanting open source that Trixbox isn't. |
| 23:36.33 | Kyosh | p3nguin, to him, a magazine is a more credible source than the inventor of the invention |
| 23:37.05 | corretico | <[TK]D-Fender>thanks. now I have sip set debug enable |
| 23:37.15 | p3nguin | Another quote of something he didn't say? |
| 23:37.23 | corretico | let me try to get a complete copy of the CLI |
| 23:38.17 | dijib | why is the digium asterisk repo giving me a 404? |
| 23:40.07 | p3nguin | What did you use for it? |
| 23:40.43 | cusco | cisco will ask for money/contract |
| 23:40.48 | cusco | oops |
| 23:40.54 | cusco | sorry buffer was high |
| 23:40.54 | p3nguin | There's no releasever 6, so that's probably where you went wrong. |
| 23:41.26 | dijib | im using http://packages.asterisk.org/centos/centos-asterisk.repo |
| 23:41.39 | p3nguin | That's a file, not a URL. |
| 23:41.52 | p3nguin | I mean... |
| 23:41.55 | p3nguin | That's a file, not a repo URL. |
| 23:42.24 | corretico | http://pastebin.com/4gayg8CH |
| 23:43.01 | dijib | thats what im using as my centos-asteerisk.repo |
| 23:43.11 | p3nguin | I'll rephrase. |
| 23:43.36 | p3nguin | When you have used that repo file, and you try to install something, what URL does it indicate has failed? |
| 23:44.38 | dijib | whats the utility that includes wget? wget? |
| 23:44.42 | p3nguin | Perhaps something like http://packages.digium.com/centos/6/current/i386/RPMS/some-package.rpm |
| 23:44.51 | [TK]D-Fender | corretico: 2899 is what you are sending to the Cisco |
| 23:45.06 | [TK]D-Fender | corretico: as we see here : -- Executing [s@macro-dialout-trunk:19] Dial("SIP/3214-08b41230", "SIP/to_ivr_alepo/2899|300|") in new stack |
| 23:45.10 | corretico | yes |
| 23:45.11 | dijib | im trying to use yum as my package handler |
| 23:45.14 | corretico | extension 2899 |
| 23:45.17 | [TK]D-Fender | corretico: And here : INVITE sip:2899@10.10.12.11 SIP/2.0 |
| 23:45.32 | [TK]D-Fender | corretico: the Cisco responds : SIP/2.0 404 Not Found |
| 23:45.36 | p3nguin | yum is a perfectly good package manager front-end, but you have to give it valid information. |
| 23:45.37 | corretico | the cisco ip address is 10.10.12.11 |
| 23:45.45 | [TK]D-Fender | corretico: which means it doesn't like that number and doesn't know what to do with it |
| 23:46.06 | corretico | <[TK]D-Fender>ohhhh great... |
| 23:46.14 | p3nguin | another quote. |
| 23:46.23 | corretico | possible a Cisco Configuratioin problem |
| 23:46.24 | corretico | !! |
| 23:47.03 | p3nguin | dijib: Since you don't seem to pick up on it, I'll just tell you in plain English. If you are using $releasever in your repo URL and you are using CentOS 6, IT WILL FAIL. Do you understand now? |
| 23:47.41 | p3nguin | There is no releasever 6 in the asterisk repo. |
| 23:48.53 | dijib | <PROTECTED> |
| 23:48.59 | p3nguin | Using 5 now? |
| 23:49.01 | dijib | im just hungry and not thinking perfectly. |
| 23:49.10 | dijib | it was the $releaserver variable\ |
| 23:49.13 | dijib | sorry to trouble you |
| 23:49.16 | p3nguin | sigh |
| 23:49.32 | p3nguin | Do you read the stuff I type for you? |
| 23:49.37 | p3nguin | I don't type it for my enjoyment. |
| 23:49.42 | dijib | yes, when im not in vi. |
| 23:49.57 | dijib | i only have a 12" screen on this laptop |
| 23:50.03 | dijib | limited realestate |
| 23:50.20 | p3nguin | I would have put my editor on another virtual desktop. |
| 23:50.41 | p3nguin | IRC on 1, editor on 2, browser on 3, etc. |
| 23:50.43 | dijib | im in 7.... |
| 23:50.51 | p3nguin | Virtual desktop 7? |
| 23:50.55 | dijib | dont want to install more windows crap |
| 23:51.09 | p3nguin | Now you've really lost me. |
| 23:51.38 | dijib | dont worry about it.. im not running linux. and i dont like things running as i play games and like to keep ram & cpu low. |
| 23:52.26 | dijib | ya so whats a good yum install asterisk line look like... what packages do i need |
| 23:52.39 | corretico | <[TK]D-Fender>i gonna check with the cisco guy |
| 23:52.55 | corretico | <[TK]D-Fender>thanks a lot 4 your time |
| 23:52.58 | p3nguin | yum install asterisk18 asterisk18-configs |
| 23:53.16 | p3nguin | corretico: Why do you keep quoting him as saying things he has not said? |
| 23:54.07 | dijib | nothing else? |
| 23:54.14 | dijib | im going to make dinner, ill be back |
| 23:54.28 | p3nguin | yum install dahdi-linux |
| 23:55.01 | p3nguin | corretico: If you're copying and pasting his nick including the angled brackets, stop it. If you want to address him, type in tk and press the tab key to complete his nick automatically. |
| 23:55.07 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 23:55.20 | p3nguin | corretico: It's like magic or something. |
| 23:58.29 | [TK]D-Fender | corretico: You're welcome |
