| 00:00.21 | em_pleh | WIMPy http://pastebin.com/uhVJ4rZF |
| 00:02.21 | WIMPy | Maybe you've got one-way communication? |
| 00:02.53 | WIMPy | I see Data from the client, but no raction to what Asterisk sends. |
| 00:03.53 | WIMPy | Do you have some firewall running on the client? |
| 00:14.51 | *** join/#asterisk talntid (~cl0se@c-98-232-117-119.hsd1.wa.comcast.net) |
| 00:25.15 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
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| 00:53.02 | *** join/#asterisk seraphie (~erin@207.98.195.107) |
| 00:55.10 | *** join/#asterisk m_tadeu (~quassel@89.180.30.224) |
| 00:56.00 | *** join/#asterisk raden (~tanning@66-191-96-74.static.eucl.wi.charter.com) |
| 00:56.18 | raden | I have someone try to brute force register asterisk how the hell do i stop this ? |
| 00:56.28 | raden | r '50.23.184.74:5076' - No matching peer found |
| 00:56.28 | raden | [May 16 19:56:23] NOTICE[17792]: chan_sip.c:23511 handle_request_register: Registration from '"cedric" <sip:cedric@66.191.96.76>' failed for '50.23.184.74:5076' - No matching peer found |
| 00:56.30 | raden | [May 16 19:56:23] NOTICE[17792]: chan_sip.c:23511 handle_request_register: Registration from '"cedric" <sip:cedric@66.191.96.76>' failed for '50.23.184.74:5076' - No matching peer found |
| 00:56.33 | raden | [May 16 19:56:23] NOTICE[17792]: chan_sip.c:23511 handle_request_register: Registration from '"cedric" <sip:cedric@66.191.96.76>' failed for '50.23.184.74:5076' - No matching peer found |
| 00:56.46 | WIMPy | No need to flood the channel as well. |
| 00:57.01 | raden | didnt mean to |
| 00:57.05 | raden | you should see my console |
| 00:57.10 | raden | like 10 reghs a second |
| 00:57.32 | WIMPy | Time to review sip.conf |
| 00:57.48 | m_tadeu | hi...I'm not receiving the bridge event in my AMI application. what do I have to do to receive it? |
| 00:58.26 | raden | eny=50.0.0.0/255.0.0.0 |
| 00:58.30 | raden | deny |
| 01:03.35 | raden | deny=0.0.0.0/0.0.0.0 |
| 01:03.35 | raden | permit=192.168.1.0/255.255.255.0 |
| 01:03.35 | raden | permit=66.168.15.0/255.255.255.0 |
| 01:03.39 | raden | what am i doing wrong ? |
| 01:04.53 | WIMPy | Hmm. I'm not sure, you can have multiple permits. |
| 01:06.14 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 01:07.59 | m_tadeu | I'm using asterisk 1.8...as I figure, it should be using AMI 1.1, right? |
| 01:08.03 | raden | OMG this flood is ridiculous |
| 01:12.29 | _pll | Do a iptables drop? |
| 01:14.00 | _pll | If you are only allowing those two networks I would consider iptables. |
| 01:25.31 | *** join/#asterisk Aut0Exec (~Joe@65.75.65.130) |
| 01:25.51 | Aut0Exec | anyone know how to get google voice going in asterisk... ? |
| 01:27.13 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 01:31.13 | Aut0Exec | can someone please help with google voice.. I have an account setup but where do I foward my calls to get them coming into asterisk? |
| 01:33.13 | raden | _pll, not working |
| 01:44.27 | _pll | iptables not working? Something is wrong in your iptables script then. You would be blocking at packet level. |
| 01:44.30 | raden | i found the ISP contacted them |
| 01:45.25 | _pll | Do this: /sbin/iptables -I INPUT -s 50.23.184.74 -j DROP |
| 01:52.59 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:53.00 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:56.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:19.25 | *** part/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
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| 02:37.23 | *** join/#asterisk raden_work (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 02:41.57 | raden_work | HOW can i stop a brute force asterisk attack ? |
| 02:43.10 | raden_work | how can i log everything on asterisk console ? |
| 02:49.45 | WIMPy | use iptables |
| 02:50.05 | WIMPy | Some like to get that done by fail2ban |
| 02:59.30 | _pll | raden_work: Do this: /sbin/iptables -I INPUT -s 50.23.184.74 -j DROP |
| 03:00.14 | *** join/#asterisk coppice (~chatzilla@m180-219-225-189.smartone-vodafone.com) |
| 03:03.51 | *** join/#asterisk p3nguin (~xwQ5kwYl6@2001:4978:202:beef:20c:29ff:fe62:be33) |
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| 03:19.10 | *** join/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 03:21.20 | Docfxit | Hi, |
| 03:21.43 | Docfxit | I tried to do an install apt-get install fail2ban I'm getting an error saying: |
| 03:21.59 | Docfxit | Couldn't find package fail2ban |
| 03:22.40 | Docfxit | Any ideas how I can get fail2ban? |
| 03:39.58 | *** join/#asterisk g00gle (~thameema@c-98-248-232-219.hsd1.ca.comcast.net) |
| 03:44.53 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 03:47.13 | g_r_eek | Docfxit: :) i just came to the channel for the same |
| 03:47.44 | g_r_eek | i just installed all ok it starts but it does not seem to work |
| 03:48.05 | g_r_eek | i followed http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk |
| 03:50.41 | Docfxit | g_r_eek I found the same page. I was trying to follow it. I got stuck on Couldn't find package fail2ban |
| 03:51.50 | Docfxit | I thought Maybe a re-boot would fix it. I lost my gnome menu. Now I can't do anything. Not even a terminal screen. |
| 03:54.42 | g_r_eek | Docfxit: sorry to hear that, i used this script: http://www.voip-info.org/liberty/view/file/3399 |
| 03:55.08 | g_r_eek | http://www.voip-info.org/liberty/download/file/3399 |
| 03:55.51 | *** join/#asterisk gh0st3r (~gh0st@123.243.120.136) |
| 03:58.08 | WiretapWork | Docfxit, ctrl+alt+f1 |
| 03:58.13 | WiretapWork | get yourself out of X |
| 03:59.19 | Docfxit | g_r_eek Thanks for the script. |
| 03:59.58 | Docfxit | WiretapWork Thanks. I tried that. It didn't work. |
| 04:00.26 | *** join/#asterisk oliver1 (~oliver@manz-5f74a9a4.pool.mediaWays.net) |
| 04:00.33 | WiretapWork | Docfxit, if that didn't take you to a text console, your machine is seriously screwed |
| 04:02.49 | Docfxit | WiretapWork Thanks. I needed that right now. |
| 04:03.11 | WiretapWork | Docfxit, what do you mean by 'lost my gnome menu' |
| 04:07.55 | Docfxit | WiretapWork I have the background wall paper. Not bar on top or bottom. |
| 04:08.07 | Docfxit | Not = no |
| 04:08.17 | WiretapWork | Docfxit, your gnome is broken, or at least your user profile |
| 04:08.23 | WiretapWork | delete your gnome prefs from your home directory |
| 04:08.57 | Docfxit | WiretapWork how would you suggest I get to the home directory? |
| 04:09.15 | WiretapWork | Docfxit, get yourself to a text console and log in |
| 04:09.25 | WiretapWork | CTRL+ALT+F[123456] |
| 04:10.39 | Docfxit | WiretapWork I tried all of those. It didn't do anything. |
| 04:10.52 | WiretapWork | you sure your system isn't locked up/ |
| 04:11.03 | WiretapWork | ctrl+alt+backspace |
| 04:11.08 | WiretapWork | that should kill X |
| 04:11.12 | WiretapWork | and force it to restart |
| 04:12.45 | *** part/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 04:13.01 | Docfxit | WiretapWork I tried that. It didn't do anything. |
| 04:13.23 | WiretapWork | Docfxit, your machine is hard locked, reboot it |
| 04:13.32 | WiretapWork | I'm assuming you don't have a cursor either/ |
| 04:13.41 | Docfxit | I called an extension that forwards the call back to me. That worked fine. So the system isn't locked up. |
| 04:13.52 | WiretapWork | I'm gonna hazard a guess that this is ubuntu |
| 04:14.05 | WiretapWork | well, X is locked up good and proper at the very least |
| 04:14.17 | WiretapWork | why you're mixing desktop and pbx I don't know |
| 04:14.19 | Docfxit | yes. It's Ubuntu. |
| 04:15.01 | Docfxit | I do have a cursor. |
| 04:15.40 | WiretapWork | I don't understand why you haven't rebooted yet... |
| 04:16.35 | Docfxit | WiretapWork I'm remote. It's hard to reach the button. It's an hour drive from here. |
| 04:16.48 | Docfxit | I wish I could reboot. |
| 04:17.34 | WiretapWork | I feel that I'm being trolled |
| 04:17.53 | Docfxit | What does that mean? |
| 04:18.25 | WiretapWork | you're an hour's drive from a machine you don't know how to use properly, and you were trying a bunch of local-only commands |
| 04:19.10 | Docfxit | I'm running VNC so I can control the display remotely. |
| 04:19.56 | WiretapWork | you don't have SSH? |
| 04:20.29 | Docfxit | I think I do but I'm not as familiar with it. |
| 04:20.33 | Maliuta | VNC??? 'sif use a GUI, let alone remotely |
| 04:20.51 | Docfxit | Now that you mention that. Wouldn't that give me a terminal window? |
| 04:21.11 | Maliuta | ssh gives you a remote shell |
| 04:21.20 | WiretapWork | Docfxit, 'secure shell' |
| 04:21.26 | Maliuta | you can run it _in_ a terminal window |
| 04:21.28 | WiretapWork | you honestly have no idea how to use a linux computer? |
| 04:21.40 | Maliuta | ssh my_user@my.host |
| 04:22.04 | WiretapWork | if that's the case I'd suggest you stop fucking with it... you're going to cause yourself and anyone else who relies on that PBX a world of hurt |
| 04:22.08 | Docfxit | WiretapWork. I'm trying my best. It's not my strong sute. |
| 04:22.33 | WiretapWork | learning is what you do on a lab box, in your spare room |
| 04:22.38 | WiretapWork | not on a bloomin production unit |
| 04:23.55 | Docfxit | WiretapWork. I've had it running for over two years. I've never had to know everything about Ubuntu in order to keep it running. |
| 04:24.15 | WiretapWork | the problem is nothing to do with 'ubuntu' |
| 04:24.23 | WiretapWork | but more to do with a lack of general unix knowledge |
| 04:25.25 | Docfxit | WiretapWork That's why I'm here when I get in trouble. |
| 04:26.41 | el3slave | get putty and try to ssh to the box |
| 04:27.39 | el3slave | er whatever system youre on, ssh to asterisk box; hopefully its available |
| 04:29.38 | Docfxit | el3slave Thanks. I have putty. I need to change the router to direct port 22 to Ubuntu. |
| 04:30.50 | el3slave | well thats your in as long as your ubuntu machine has ssh enabled |
| 04:41.16 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-edxuhqimhzecmnmd) |
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| 04:52.22 | ppc | hey didn't someone in here have a dial in box |
| 04:52.38 | ppc | like a pbx system you could call in and test etc |
| 04:52.43 | ppc | or conf with people |
| 04:56.24 | *** part/#asterisk Docfxit (~none@netblock-75-79-6-149.dslextreme.com) |
| 04:58.10 | WiretapWork | yep |
| 04:58.25 | WiretapWork | depends how you want to get at it |
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| 05:19.37 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 06:08.31 | *** join/#asterisk aiksa[LV] (~aiksaLV]@212.70.182.16) |
| 06:08.38 | aiksa[LV] | Hi everyone. |
| 06:09.28 | aiksa[LV] | after a number of years using asterisk queues I just noticed that while periodic queue announcment is playing, asterisk is not trying to connect the caller to any agent |
| 06:09.38 | aiksa[LV] | has it always been this way? |
| 06:17.47 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 06:17.50 | schmidts | good morning |
| 06:18.56 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 06:20.26 | nellyb | morning! |
| 06:22.05 | *** join/#asterisk e7e5 (~rudenko@77.239.239.27) |
| 06:25.47 | *** join/#asterisk nicola_pav (~chatzilla@mail2.tikalnetworks.com) |
| 06:28.08 | g00gle | morning guys |
| 06:28.19 | g00gle | anyone up for some brainstorming discussion |
| 06:29.19 | nicola_pav | hello. sometimes asterisk freezes and and when i try to login to asterisk cli it pops out msgs and returns me to the shell. I want to write a script to monitor when this happens. |
| 06:29.27 | nicola_pav | any ideas hints please? |
| 06:29.29 | schmidts | g00gle this depends what you want to talk about ;) |
| 06:29.57 | g00gle | ok everyone knows asterisk 1.8 natively supports google talk |
| 06:30.00 | schmidts | nicola_pav recompile with DEBUG_THREADS enabled in menuselect and take a look at core show locks |
| 06:30.10 | schmidts | g00gle yes |
| 06:30.28 | g00gle | from asterisk you can terminate a call to google chat user |
| 06:30.46 | g00gle | but is there any way that the user can initiate a call with asterisk? |
| 06:31.11 | nicola_pav | schmidts: its not what i want exactly. the script i want to write will send me an email when asterisk is stuck, i will put it in crontab to run every 1 minute |
| 06:31.21 | g00gle | lets say that we have a buddy configured with asterisk.. user A adding this asterisk buddy |
| 06:31.34 | nicola_pav | my question is how to monitor when it freezes? |
| 06:31.39 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:c62c:3ff:fe2d:b7db) |
| 06:31.50 | g00gle | nicola_pav -> why don't you use SNMP |
| 06:32.02 | g00gle | nicola_pav -> asterisk supports it |
| 06:32.04 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 06:32.08 | schmidts | nicola_pav take a look at sipsak, this will send an option message to asterisk and if you dont get a response its stuck |
| 06:32.16 | schmidts | or use snmp ;) |
| 06:32.36 | nicola_pav | schmidts: i can write a script that runs the following: asterisk -rx'sip show peers' and put it in a variable. and if this variable is empty i know that asterisk is not running. |
| 06:32.37 | schmidts | g00gle i didnt use google talk so i didnt know how this could work |
| 06:32.45 | nicola_pav | the above does not apply when asterisk is stuck. |
| 06:32.57 | nicola_pav | i need another parameter or variable to look at |
| 06:33.00 | g00gle | nicola_pav -> exactly right |
| 06:33.09 | schmidts | nicola_pav sipsak -N IMHO :D |
| 06:33.27 | nicola_pav | schmidts: sipsak? |
| 06:33.30 | schmidts | sipsak |
| 06:33.36 | schmidts | ;) |
| 06:33.52 | nicola_pav | schmidts: i am afraid i am not familiar with |
| 06:33.55 | nicola_pav | is it a command? |
| 06:34.17 | g00gle | nicola -> can you define what do you mean by 'stuck' |
| 06:34.27 | nicola_pav | i am googling ot |
| 06:34.31 | schmidts | http://sipsak.org/ its a program which sends sip OPTIONS messages to the destination ip you give him and if the reply doesnt come in time you can see that asterisk is dead |
| 06:34.37 | g00gle | does it mean the safe_asterisk is running but no response from asterisk? |
| 06:34.51 | schmidts | i guess its deadlocked |
| 06:34.59 | nicola_pav | g00gle: with stuck i mean when u try to login to asterisk cli but u return to the terminal instead of logging to the cli |
| 06:35.11 | schmidts | safe_asterisk only checks if the process still runs, not if asterisk is deadlocked |
| 06:35.24 | g00gle | which means process dead but .pid file exists |
| 06:35.45 | schmidts | not really, maybe only one or some threads of the process is death not the whole process |
| 06:35.48 | nicola_pav | sorry for the terminology, maybe what i am looking for is called deadlocked |
| 06:36.24 | g00gle | you can better use the snmp |
| 06:36.31 | g00gle | with some publicly available snmp servers |
| 06:36.44 | nicola_pav | i want to integrate my script in zabbix monitoring |
| 06:37.24 | nicola_pav | i will try sipsak. but of course i need to test it :) |
| 06:37.25 | g00gle | i believe zabbix has snmp agent |
| 06:37.39 | nicola_pav | how can i freeze, stuck or get asterisk into a deadlock? :) |
| 06:37.45 | g00gle | yes it does |
| 06:38.00 | g00gle | so you just configure res_snmp.conf and provide zabbix information |
| 06:38.13 | g00gle | it can receive snmp traps |
| 06:38.23 | schmidts | nicola_pav search for the backtrace.txt file in your asterisk source there you can find information how to debug such an issue |
| 06:38.49 | nicola_pav | g00gle: so asterisk sends via snmp info to zabbix that asterisk is in deadlock? |
| 06:39.24 | g00gle | yes i believe |
| 06:39.33 | g00gle | i haven't tested though |
| 06:39.41 | g00gle | http://www.mzintegration.com/how-to-1/monitoring-asterisk |
| 06:40.31 | schmidts | nicola_pav it will not send that its deadlocked but it will not response to snmp requests, so you will see that its dead ;) |
| 06:40.45 | nicola_pav | g00gle schmidts: thanks, i will take a look on what u suggested |
| 06:40.51 | nicola_pav | i got it |
| 06:41.01 | nicola_pav | its ok unless i can test it |
| 06:41.11 | schmidts | nicola_pav btw which version do you use? |
| 06:41.17 | g00gle | here is another link may be useful for you |
| 06:41.19 | g00gle | http://voxilla.com/2009/02/03/configuring-asterisk-snmp-support-1131 |
| 06:41.31 | nicola_pav | schmidts i use 1.4 |
| 06:42.27 | *** join/#asterisk JT (~j@unaffiliated/jt) |
| 06:42.38 | schmidts | have 1.4 snmp support? i thought its only in 1.6? |
| 06:42.48 | wdoekes2 | older 1.4 perhaps? it is known to stop responding to cli commands |
| 06:43.02 | g00gle | yes it does have |
| 06:43.54 | g00gle | http://www.voip-info.org/wiki/view/Asterisk+Documentation+1.4+snmp.txt |
| 06:46.17 | g00gle | ok guys back to my topic.. who used gtalk with asterisk? |
| 06:46.51 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 06:46.55 | jacc0 | good morning all |
| 06:47.23 | g00gle | morning jacc0 |
| 06:47.31 | g00gle | where u from? |
| 06:47.55 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 06:47.57 | kaushal | Hi |
| 06:48.21 | kaushal | when i run dahdi_cfg -vv, i get DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 06:48.26 | kaushal | Please suggest |
| 06:48.41 | schmidts | morning jacc0 |
| 06:51.57 | kaushal | shall i pastebin the configs ? |
| 06:52.49 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 06:59.08 | *** join/#asterisk manji (~manjiki@84.254.40.156) |
| 07:01.04 | jacc0 | hi schmidts; I'm still waiting for the license to be accepted :S |
| 07:01.24 | *** join/#asterisk jkroon (~jkroon@dsl-242-4-65.telkomadsl.co.za) |
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| 07:03.45 | schmidts | jacc0 :D i think this taked some time for me too |
| 07:06.24 | wdoekes2 | kaushal: many of us aren't familiar with dahdi, but pb'ing the configs shan't hurt |
| 07:06.38 | wdoekes2 | does your dmesg say anything, btw? |
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| 07:13.15 | dan__t | This is a terrible way to describe what I'm looking for, I know, but is there a channel variable that represents the 'dst' portion of a cdr record stored in sql? |
| 07:13.27 | *** join/#asterisk sphenxes (~sphenxes@84.119.95.241) |
| 07:13.59 | aiksa[LV] | ${EXTEN} ? |
| 07:14.36 | dan__t | That would be it. |
| 07:14.37 | dan__t | Actually. |
| 07:15.10 | aiksa[LV] | :) |
| 07:15.41 | schmidts | dan__t IMHO ${CDR(dst)} should also work |
| 07:16.03 | dan__t | Very cool. Wasn't thinking straight there, my apologies. |
| 07:16.03 | aiksa[LV] | yup |
| 07:16.09 | aiksa[LV] | yet beware |
| 07:16.10 | aiksa[LV] | ! |
| 07:16.22 | aiksa[LV] | CDR gets written at the time of completition of the call |
| 07:16.59 | aiksa[LV] | so if your dialplan goes through hops and hoops jumping contexts, extensions and prioerities, I suppose that dst hold the last extension the call was at |
| 07:17.05 | aiksa[LV] | IMHO |
| 07:17.08 | dan__t | Understood. |
| 07:17.33 | dan__t | wasn't looking for a CDR-specific example, tha was just the best I could describe it. |
| 07:17.44 | dan__t | But it's nice to know I can reference cdr vars that way. |
| 07:18.02 | schmidts | CDR is a function, take a look at core show function CDR ;) |
| 07:18.04 | aiksa[LV] | yeah. Now if only we could write them that easy from ami |
| 07:18.20 | dan__t | hehe |
| 07:18.51 | kaldemar | dan__t: CDR(dst) |
| 07:19.29 | kaldemar | looks like schmidts already said it. |
| 07:20.13 | aiksa[LV] | kaldemar: actually any difference betwwen ${EXTEN} and CDR(dst) besides the obvious differences after ForkCDR |
| 07:20.20 | aiksa[LV] | ? |
| 07:23.46 | kaldemar | EXTEN is always the current extension, but CD(dst) is what was dialled in the first place. a goto does not change it. |
| 07:35.40 | aiksa[LV] | ok |
| 07:37.17 | *** join/#asterisk rich_ (~rich@193.220.59.2) |
| 07:39.26 | *** join/#asterisk Aas (~aas@static-dsl-88.87-197-156.telecom.sk) |
| 07:39.59 | Aas | hello all, can you recommend some call billing system? ...something you have used and have good experiences with |
| 07:52.58 | dan__t | So I guess function arguments can be separated by periods, as well as comments? I vaguely remember reading this. Can I override this behavior for a specific part of my dialplan? |
| 07:53.41 | dan__t | Trying to MixMonitor this channel, giving the resulting filename that of ${UNIQUEID}. Since ${UNIQUEID} contains a period, MixMonitor interprets data immediately after that as additional arguments. |
| 07:55.08 | wdoekes2 | dan__t: FUNC("${UNIQUEID}",abc) |
| 07:55.21 | dan__t | oh |
| 07:55.34 | wdoekes2 | but periods shouldn't separate arguments |
| 07:55.44 | wdoekes2 | in the old days pipes did |
| 07:56.28 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 07:56.39 | dan__t | huh |
| 07:59.49 | dan__t | Nice. An extra double quote. Asterisk wasn't doing what I thought it was doing. It was behaving pefectly. |
| 07:59.53 | dan__t | perfectly, too. |
| 08:03.50 | *** join/#asterisk rethus (~suther@p5087BBAF.dip.t-dialin.net) |
| 08:04.19 | rethus | i get asterisk-events over AMI. Is it possible to get the EVENT-Response in XML ? |
| 08:06.56 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 08:13.16 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 08:13.26 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 08:15.43 | *** join/#asterisk markit (~marco@adsl-156-149.38-151.net24.it) |
| 08:16.19 | markit | hi, what is "asterisk gui 2.0"? I thought asterisk was using FreePBX web interface... is it the same just "rebranded"? |
| 08:16.37 | markit | (newbie, command line only configurator here :)) |
| 08:17.44 | Chainsaw | markit: A strong recommendation to stay with the terminal. |
| 08:18.41 | markit | Chainsaw: :) but I'm curious |
| 08:20.53 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:23.14 | *** join/#asterisk ruyo (~psantos@a83-132-152-91.cpe.netcabo.pt) |
| 08:23.15 | jkroon | hi guys, what would be my chances of getting someone from Digium in here that would be able to assist with an issue I've (on recommendation from this #) logged with Digium but I'm not getting any joy on? |
| 08:23.38 | *** join/#asterisk awk (~awk@equipoise.hack.co.za) |
| 08:24.17 | awk | Hi guys, I have been focused on Asterisk 1.4 for some time now and been left out of the mailing lists and changes to new development, is there any articles anyone has on new features from 1,4 right up till 2.0 that you can direct me at? |
| 08:24.36 | jkroon | markit, i have to agree with Chainsaw on that one. I find that even though I'm one of those people that write my own management interfaces I consistently keep dropping back to the terminal to get things done. My recommendation: stick with the terminal. |
| 08:25.30 | dan__t | Using MixMonitor, I see this: == Executing ["/var/recordedcalls/processrecordedcall.sh 1305620668.3240"] |
| 08:25.37 | dan__t | It's as if the script doesn't ever get ran. |
| 08:25.48 | dan__t | I can, however, run that same comand on the command-line, as the asterisk user, and things work properly. |
| 08:25.51 | schmidts | jkroon at this time you will not see anyone from digium, they still sleep ;) but maybe we can help you too |
| 08:26.02 | kaldemar | markit: asterisk doesn't use anything. a GUI uses asterisk. there are many GUIs, freepbx and asterisk gui being two of those. |
| 08:26.23 | jkroon | schmidts, got a B410P where one of the ports won't come UP. All alarms are cleared but L2 stays Down. |
| 08:26.37 | jkroon | the port staying down follows the NTU, but even after replacing the NTU ... no change. |
| 08:26.42 | jkroon | what info would you need? |
| 08:27.23 | schmidts | jkroon what do you see in dmesg or your syslog? |
| 08:28.55 | *** join/#asterisk frawd (~francois@242.Red-81-38-28.dynamicIP.rima-tde.net) |
| 08:29.08 | wdoekes2 | dan__t: does asterisk have access to /bin/sh? (is it possibly chrooted?) |
| 08:31.24 | markit | jkroon: I've asked here to know something more about that GUI, why all telling me to use command line? I mean, I could ask "how are bananas like?" and still love apples and not going to eat anything else |
| 08:31.31 | dan__t | It does, I edited /etc/passwd appropriately. |
| 08:31.53 | dan__t | As the AST_USER and AST_GROUP |
| 08:32.10 | dan__t | uh, before that last part - as the asterisk user, I was able to run that command without incident. |
| 08:32.18 | markit | I remember asterisk was starting "asteriskGui" project with a "self made web interface", then if I recall correctly they "switched" to freepbx, so wondering now what are they doing |
| 08:34.24 | dan__t | https://issues.asterisk.org/view.php?id=16534 - hmmm |
| 08:34.29 | kaldemar | markit: more like digium was starting the project. unless i'm mistaken, someone is maintaining the asterisk gui (visit #asterisk-gui), but for example asterisknow uses freepbx by default. |
| 08:34.33 | *** part/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 08:35.25 | markit | kaldemar: oh, correct, AsteriskNow was the project |
| 08:38.11 | dan__t | any other ideas there, wdoekes2 ? |
| 08:38.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 08:42.12 | dan__t | 1:38 |
| 08:43.02 | rethus | Is there a overview, in which case - wich event-type returnd if i execute a AMI-Command? |
| 08:43.14 | rethus | if i do konference list, i got _clilist as first event-type |
| 08:43.29 | rethus | if i execute konference list MyConference, i got _cliresponse |
| 08:48.42 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 08:52.18 | rich_ | how do i dial from the cli? |
| 08:52.29 | rich_ | I do not have that command listed, is it a module to be added? |
| 08:53.09 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85fc.bcn.adamo.es) |
| 08:54.49 | kaldemar | rich_: console dial ... |
| 08:57.40 | kaldemar | rich_: the commands are provided by chan_oss, chan_alsa or chan_console. you need one of those modules. |
| 09:01.19 | *** join/#asterisk pif (~ldm@zenon.apartia.fr) |
| 09:01.36 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59) |
| 09:10.12 | rich_ | cool |
| 09:10.14 | rich_ | thanks |
| 09:18.49 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 09:19.21 | puzzled | hi |
| 09:22.46 | *** part/#asterisk Aas (~aas@static-dsl-88.87-197-156.telecom.sk) |
| 09:28.52 | Lantizia | Lo, can anyone give me some ideas as to why this is happening? http://asterisk.pastebin.ca/2062236 |
| 09:29.16 | Lantizia | it's an incoming call from my ITSP - was working yesterday just fine... |
| 09:29.24 | Lantizia | full sip and rtp debug is on |
| 09:32.49 | Chainsaw | Lantizia: You're telling them to go away. 401 Unauthorized. |
| 09:32.58 | Chainsaw | Lantizia: i.e. "You should have sent me some credentials and didn't" |
| 09:33.12 | Chainsaw | Lantizia: Or "dial plan says no". |
| 09:33.32 | Chainsaw | Lantizia: core set verbose 10 & core set debug 10 would probably help more, as that shows the dial plan code being executed. |
| 09:33.43 | Lantizia | lemme try that... |
| 09:35.26 | Lantizia | shows about the same logs tbh |
| 09:35.36 | Lantizia | it doesn't get to the dial plan |
| 09:36.04 | Wiretap | wow... this is a weird issue my mate is having |
| 09:36.16 | Lantizia | i'm at a loss why it would choose to 401 inbound calls now but yesterday it wasn't :S |
| 09:36.28 | Chainsaw | Lantizia: Generally, due to a configuration change in sip.conf |
| 09:36.29 | wdoekes2 | Lantizia: did you remove the insecure=invite this morning? |
| 09:36.35 | Wiretap | he gets audio if he calls an internal extension from an external extension (not trunk) |
| 09:36.45 | Wiretap | but if he calls from the external extension out a trunk, no audio |
| 09:37.02 | Lantizia | wdoekes2, insecure is still set to very |
| 09:38.05 | wdoekes2 | I doubt it |
| 09:38.14 | wdoekes2 | sip show peer 811223530 |
| 09:38.31 | kaldemar | Lantizia: very is not a supported option in recent versions. what version are you using? |
| 09:38.44 | Lantizia | wdoekes2, was only on the outbound bit - not the inbound - fixed now...i bet my colleague has foxed with it |
| 09:38.49 | Lantizia | 1.8.4 |
| 09:39.50 | kaldemar | valid values are invite and port. |
| 10:05.52 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 10:16.01 | *** part/#asterisk markit (~marco@adsl-156-149.38-151.net24.it) |
| 10:17.21 | rich_ | how do i console dial an external line? |
| 10:18.58 | kaldemar | you can only dial extension[@context]. it is up to your dialplan to define where the call goes. |
| 10:20.44 | rich_ | can i plae real call using the cli |
| 10:21.28 | kaldemar | that is a real call. maybe originate is what you're looking for? |
| 10:22.52 | kaldemar | "core show help channel originate" in CLI will tell you more (if you have the cli originate module, res_clioriginate). |
| 10:23.58 | *** join/#asterisk Syrex (~Syrex@dsl-241-198-253.telkomadsl.co.za) |
| 10:24.18 | Syrex | Hello everybody... |
| 10:24.38 | *** part/#asterisk rethus (~suther@p5087BBAF.dip.t-dialin.net) |
| 10:24.40 | Chainsaw | Hi Syrex. |
| 10:26.31 | Syrex | Anyone here know in which source code file I could modify the 'from: ' header sent for notify messages? Polycom phones expect notify message to have a 'from: "1904" <1904@sip.server.com>' header instead of 'from: <1904@sip.server.com>' Directed call pickup isn't working :( |
| 10:27.09 | Syrex | Stupid Polycom is simply sending ** instead of **1904 |
| 10:28.16 | WiretapSeven | Syrex, aside from patching and recompiling |
| 10:29.13 | Syrex | WiretapSeven: Don't mind patching and recompiling, unless the 'from' header can be modified in configuration files? It's essentially the watcher notify messages which Asterisk is sending to the phones I want to modify slightly... |
| 10:29.48 | WiretapSeven | Syrex, don't think you can do it from config |
| 10:31.49 | Syrex | WiretapSeven: Any idea of which file I could poke around in? Have a decoded tcpdump ( similar to sip debug) for a notify for a dialog-info+xml hint subscription... |
| 10:32.30 | WiretapSeven | Syrex, you might wanna take a look at the Cisco Unified patches, (gareth-1.8.3.patch is the latest), as they had to make similar changes for these phones |
| 10:35.21 | *** join/#asterisk chalcedony (~llhull@unaffiliated/chalcedony) |
| 10:53.15 | Syrex | WiretapSeven: Doesn't appear to relate directly... Found a patch which tells Asterisk to speak xpidf+xml although the phone claims pidf+xml... |
| 10:53.43 | WiretapSeven | Syrex, it makes a lot of extra changes, and no, its not a direct relation since its a completely different type of phone |
| 10:53.54 | WiretapSeven | anyway, I'm off |
| 10:54.06 | Syrex | WiretapSeven: Thanks anyway... Will go through the patch in greater detail... |
| 10:58.41 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-62-5.in-addr.btopenworld.com) |
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| 11:37.01 | rich_ | I would like to digimap 101 -> *97 |
| 11:37.05 | rich_ | how can i do this please? |
| 11:37.13 | rich_ | i.e. dial 101 = *97 |
| 11:41.21 | kaldemar | be more precise. you press digits 101 on something, then what? |
| 11:41.34 | *** join/#asterisk devil_evoxxx (~d3v1l@host137-92-dynamic.0-79-r.retail.telecomitalia.it) |
| 11:41.47 | devil_evoxxx | hi all |
| 11:45.27 | devil_evoxxx | i have a question a bout |
| 11:45.30 | devil_evoxxx | music on hold |
| 11:45.31 | devil_evoxxx | or |
| 11:45.43 | devil_evoxxx | playback cmd |
| 11:46.31 | devil_evoxxx | when a call comes from gsm phone to my voip number, and my voip number is not reacheable is possibile to play a message without answer? |
| 11:46.59 | devil_evoxxx | (in this mode, the gsm phone does not spend money) |
| 11:47.02 | rich_ | kaldemar, I want to dial 101 for the phone to dial *97 for voicemail |
| 11:47.02 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:47.05 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 11:47.28 | rich_ | At the momen the messges button is mapped to the extension number of the phone, i cannot manually overirde that so i wanted to setup a digimap to translate, 101 to *97 |
| 11:47.49 | kaldemar | rich_: what does *97 do in your system? |
| 11:47.54 | binbash_ | devil_evoxxx there will be charged anyway. If your Asterisk answeres the call to playback the audio file.. that's a active call |
| 11:49.14 | kaldemar | devil_evoxxx: try the playback application with option noanswer. the sip channel should then send early media. |
| 11:49.41 | kaldemar | devil_evoxxx: assuming that it is sip you use. |
| 11:49.46 | devil_evoxxx | binbash: there are no ways ? in example, when i call a gsm phone not reachable i can hear the voice that say to me that the user is not available. I need ss7 signaling? |
| 11:49.53 | devil_evoxxx | yes, i'use sip |
| 11:55.08 | *** join/#asterisk rich_ (~rich@193.220.59.2) |
| 11:55.47 | rich_ | kaldemar, is that didimap possible? |
| 11:57.25 | kaldemar | rich_: there is no such thing as digimap in asterisk. is *97 an extension in your dialplan? |
| 11:57.57 | *** join/#asterisk fish-bulb (~qcstewart@nat/digium/x-mcjhnfqponwjkiec) |
| 11:58.27 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 11:59.48 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:03.26 | kaldemar | devil_evoxxx: you might need to call the Progress application before Playback for early media to work. |
| 12:04.40 | devil_evoxxx | yea!!! is working!!! beautifull |
| 12:05.38 | *** join/#asterisk Ryushin (~Ryushin@184.248.68.85) |
| 12:07.44 | jacc0 | woei! siemens HiPath + HG1500 interface interop. test with astrisk 1.8: it works!! |
| 12:10.07 | jacc0 | is happy |
| 12:10.31 | Lantizia | Lo, what version did pipes not work (and thus need commas) |
| 12:14.21 | *** join/#asterisk MarKsaitis (~MarKsaiti@host81-137-62-5.in-addr.btopenworld.com) |
| 12:15.21 | leifmadsen | Lantizia: 1.6.0 and later |
| 12:15.28 | Lantizia | thought so, thanks leifmadsen :P |
| 12:15.29 | leifmadsen | commas work in 1.2 and 1.4 |
| 12:15.43 | Lantizia | but pipes _stop_ working 1.6+ yeah? |
| 12:15.54 | leifmadsen | yes, unless you change something in asterisk.conf |
| 12:16.17 | Lantizia | nah it's ok - I'm having a raging argument with the a2billing guys on how awful their installation instructions are |
| 12:16.27 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 12:16.28 | leifmadsen | [compat] |
| 12:16.29 | leifmadsen | app_set=1.6 |
| 12:16.44 | leifmadsen | Lantizia: that's because no one likes to write documentation |
| 12:17.18 | Lantizia | that'd be cool - I've offered to rewrite their documentation ... if they can clarify the points I've raised about what _might_ be wrong with the original documentation |
| 12:17.21 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:17.21 | leifmadsen | russellb, jimvanmeggelen, and myself seem to be the only crazy ones |
| 12:17.39 | Lantizia | http://forum.asterisk2billing.org/viewtopic.php?f=33&t=8849 |
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| 12:28.28 | QuantumSchema | Good morinin' all! |
| 12:29.23 | kleszcz | hi |
| 12:30.24 | *** join/#asterisk jkroon (~jkroon@dsl-242-4-65.telkomadsl.co.za) |
| 12:32.00 | jacc0 | hi |
| 12:32.06 | jacc0 | good afternoon |
| 12:33.46 | *** join/#asterisk JerJer (~jj@asterisk/original-h323-guy/JerJer) |
| 12:38.05 | leifmadsen | morning |
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| 13:05.23 | Syrex | hey... |
| 13:09.00 | jacc0 | hey!! |
| 13:09.02 | jacc0 | :p |
| 13:09.30 | Syrex | anyone here use Polycom with BLF and directed call pickup? |
| 13:13.59 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 13:14.31 | Aut0ExeC | hi guys... i'm trying to set up google voice... but how to I foward my google voice account thats linked to a physical phone to my asterisk box? |
| 13:14.45 | Chainsaw | Syrex: Just to show that people are here, I do use Polycom 670 handsets, but not the full BLF features. |
| 13:14.48 | Aut0ExeC | that part is not covered in the docs |
| 13:16.15 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:16.15 | leifmadsen | Aut0ExeC: you don't? |
| 13:16.40 | leifmadsen | Aut0ExeC: Google Voice has to be setup to send calls to Asterisk directly, or a number that is associated with Asterisk as the end point |
| 13:16.42 | Syrex | Chainsaw: I have had BLF (resource list style) with directed call pickup working on my Polycom 670s... It broke when I upgraded the firmware to 3.3.x but I'd also upgraded Asterisk at the same time... What Asterisk version do you run? Here are the steps for BLF + directed call pickup for Polycom: http://www.excaliburtech.net/archives/147 |
| 13:17.15 | Chainsaw | Syrex: I run 1.8.4 with the TCP SIP crash fix on top. |
| 13:17.52 | Chainsaw | Syrex: I'm on SIP firmware 3.2.5; the advantages of the newer style firmware were not apparent to me. |
| 13:18.26 | Aut0ExeC | leifmadsen: I dont get it ... in the google voice options... there is only a foward to a physical phone |
| 13:19.29 | Aut0ExeC | leifmadsen: how can it be possible to send calls to asterisk directly? |
| 13:19.43 | Syrex | Chainsaw: Instructions at the Excaliburtech site details extending BLF to get ringing/in use/busy status lights working right... Any chance you would try see if it works with your Asterisk version (I'm running 1.6.2.17.2). |
| 13:20.28 | Chainsaw | Syrex: It's a production deployment, I tend to avoid unnecessary changes. |
| 13:20.36 | Syrex | Chainsaw: I opened a call with Polycom and they say it's my Asterisk... I tried downgrading to 3.2.2, 3.2.3, 3.2.4 and directed call pickup just sends the pickup prefix instead of appending the extension... |
| 13:20.41 | leifmadsen | Aut0ExeC: then I must not get what you're reading or saying |
| 13:22.31 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 13:22.33 | Chainsaw | Syrex: I do use custom device states to show whether an agent is logged into a handset or not. |
| 13:22.34 | leifmadsen | runs off to write a script to generate polycom configurations |
| 13:22.38 | Chainsaw | Syrex: And those seem to work. |
| 13:22.57 | Chainsaw | leifmadsen: I have one. The Polycom Automatic Provisioning Agent framework. |
| 13:23.11 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:23.27 | c0rnoTa | hello all |
| 13:23.50 | leifmadsen | Chainsaw: nice -- I need to just build a simple script that will take the CSV file created by the scan gun system my buddy uses |
| 13:23.52 | Aut0ExeC | leifmadsen: ok can you just give me an example as to what to put in the options under google voice settings... where it says "forward calls to:"... what should I put there to get the calls to go to asterisk box? |
| 13:23.58 | leifmadsen | Aut0ExeC: sorry I can't |
| 13:24.02 | Aut0ExeC | ok |
| 13:24.05 | Aut0ExeC | np |
| 13:24.27 | leifmadsen | Chainsaw: is that a project you wrote? |
| 13:24.43 | Chainsaw | leifmadsen: I use mod_rewrite; either there is a MAChere.cnf, or it calls papa.php with the MAC of the requesting phone. |
| 13:24.56 | Chainsaw | leifmadsen: Which will write the config, and then echo it out. |
| 13:25.08 | leifmadsen | interesting, that sounds neat |
| 13:25.14 | leifmadsen | so you just provision via HTTPS then? |
| 13:25.27 | Chainsaw | leifmadsen: http; the boot loader will not support https, only the SIP application does. |
| 13:25.30 | leifmadsen | then have apache echo back the configs and rewrite the mac address that came in? |
| 13:25.34 | leifmadsen | ah gotcha |
| 13:25.37 | Chainsaw | leifmadsen: So you'd have to split amongst HTTP & HTTPS anyway. |
| 13:25.47 | c0rnoTa | please help me how I can set TON (type of number) and NPI in outgoing SETUP frame, that send by my asterisk over E1 trunk, when I call a number ? |
| 13:25.55 | leifmadsen | Chainsaw: coolio |
| 13:26.00 | Chainsaw | leifmadsen: The PHP code just echos out the config it wrote at the end. That was easiest to do :) |
| 13:26.12 | leifmadsen | yep that's what I'm going to be doing too, but generating configs for the phones |
| 13:26.50 | Chainsaw | leifmadsen: In my case rolling out new phones is rather easy; it's just about assigning station IDs in a linear fashion and checking which of the two offices it is in. |
| 13:26.59 | leifmadsen | nice nice |
| 13:27.07 | *** join/#asterisk Ryushin (~Ryushin@68-241-1-169.pools.spcsdns.net) |
| 13:27.10 | Chainsaw | leifmadsen: It then adds the relevant statement to the SIP config and reloads SIP just before echoing the config out. |
| 13:27.16 | leifmadsen | this is going to get deployed to several locations and companies, so it needs to be generic, but I like what you're doing |
| 13:27.30 | Chainsaw | leifmadsen: I'm happy to share what I have with you, perhaps it is of use. |
| 13:27.32 | leifmadsen | rock on -- I use #exec to generate the configs for the phones |
| 13:27.38 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 13:27.40 | leifmadsen | sure, want to email me? |
| 13:27.58 | Chainsaw | Will do, feel free to privmsg the address. |
| 13:28.13 | leifmadsen | mine is very public already: firstname@leifmadsen.com |
| 13:28.21 | leifmadsen | s/firstname/<firstname> |
| 13:29.48 | c0rnoTa | guys, how I can set type of outgoing number in PRI? |
| 13:30.47 | leifmadsen | maybe we don't know? |
| 13:32.19 | Chainsaw | leifmadsen: Sent. |
| 13:32.32 | leifmadsen | Chainsaw: thanks! I'll certainly check it out and send back any features I may add |
| 13:32.38 | c0rnoTa | yeap, could be :) |
| 13:32.40 | Chainsaw | leifmadsen: It's appreciated :) |
| 13:32.53 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 13:32.54 | Polysics | hello |
| 13:32.55 | leifmadsen | open source at its finest :) |
| 13:33.09 | Polysics | which dialplan variable holds the SIP name of the calling user, please? |
| 13:33.24 | Polysics | such as 10006 for SIP/10006-blah123etc |
| 13:33.57 | leifmadsen | Polysics: it wouldn't really because that's another channel -- you'd have to set it before calling the other end point and make it inherited |
| 13:34.12 | leifmadsen | Set(_MyChannel=${CHANNEL(name)}) |
| 13:34.50 | Polysics | but ${CHANNEL} does hold the correct channel, just in the full form |
| 13:36.01 | leifmadsen | so strip off the junk you don't want |
| 13:36.03 | leifmadsen | use CUT() |
| 13:36.07 | leifmadsen | problem solved |
| 13:36.09 | leifmadsen | NEXT!!! |
| 13:36.42 | Polysics | ok, got it, it's CHANNEL(peername) :-D |
| 13:37.06 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 13:37.06 | Polysics | another little thing |
| 13:37.25 | Polysics | for RECORD, i have 5 as maxduration, but it only lasts a split second |
| 13:37.39 | Polysics | it does record for that however, it's ast 1.8 if it matters |
| 13:38.12 | QuantumSchema | Ahhh! I don't know what happened! Asterisk stoped writing to queue_log. |
| 13:38.20 | QuantumSchema | ::scratches head:: |
| 13:38.24 | leifmadsen | hard disk full? |
| 13:38.59 | QuantumSchema | Nope.. 6% used, 54G free. |
| 13:39.54 | Chainsaw | QuantumSchema: Did you run out of inodes on your ext2/3/4 filesystem? |
| 13:40.17 | Chainsaw | QuantumSchema: Try using touch to create files in the relevant directory. -ENOSPC can mean two things; you're only accounting for the former, not the latter. |
| 13:40.22 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 13:40.45 | ferdna | how to limit registration only from local network? |
| 13:41.15 | kaldemar | ferdna: permit/deny |
| 13:41.18 | leifmadsen | use iptables to block outside connections, or permit/deny |
| 13:41.30 | ferdna | thank you |
| 13:42.01 | *** join/#asterisk imox1234_ (~imox1234@78.173.64.30) |
| 13:42.24 | ferdna | i got a call about 20 mins ago... from 256-428-6175 |
| 13:42.30 | ferdna | is that digium number? |
| 13:42.39 | ferdna | i googled nothing came up |
| 13:43.11 | ferdna | .. |
| 13:43.15 | leifmadsen | try calling it? |
| 13:43.18 | ferdna | and thgis is in my logs |
| 13:43.19 | ferdna | [May 17 07:20:20] NOTICE[15686] chan_sip.c: Call from 'XXXX' to extension '92564286175' rejected because extension not found in context 'phones'. |
| 13:43.28 | ferdna | leifmadsen, no i havent yet |
| 13:46.47 | pabelanger | ferdna: *CLI> dialplan show 92564286175@phones |
| 13:47.13 | QuantumSchema | Chainsaw: I was able to move the old queue_log to .old and touch a new queue_log. |
| 13:47.18 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 13:47.30 | QuantumSchema | Chainsaw: I did change the owner back to asterisk.asterisk too. |
| 13:47.32 | Chainsaw | QuantumSchema: Okay, that's the inode theory out of the window then. |
| 13:47.59 | ferdna | pabelanger, localhost*CLI> dialplan show 92564286175@phones There is no existence of 92564286175 phones extension Command 'dialplan show 92564286175@phones' failed. |
| 13:48.06 | ferdna | and that is correct |
| 13:48.09 | ferdna | that number is not mine |
| 13:48.56 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 13:49.07 | pabelanger | ferdna: So, XXXX is dialling an extension that does not exist |
| 13:49.40 | ferdna | pabelanger, i really have no clue |
| 13:49.41 | ferdna | :( |
| 13:49.45 | ferdna | i got a call this morning |
| 13:49.53 | ferdna | from that number |
| 13:50.17 | afink | Hello everyone, when you setup with a pstn SIP provider "s" should handle any call that comes in on that context correct? |
| 13:50.43 | leifmadsen | incorrect |
| 13:50.49 | QuantumSchema | ::scratches head:: this isn't good. |
| 13:51.07 | leifmadsen | "s" is mostly just for analog phones -- you can tell the ITSP to call whatever you want in the register line |
| 13:51.20 | leifmadsen | register => username:password@itsp.com/4165551212 |
| 13:51.34 | leifmadsen | then when the provider calls you, they will request 4165551212 |
| 13:51.39 | leifmadsen | you match that in your dialplan |
| 13:51.53 | leifmadsen | you could of course add /s to the end but that's not very descriptive and is not a best practice |
| 13:52.27 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 13:53.08 | QuantumSchema | Does this correlate to it not writing to the queue_log ? May 17 09:53:00] ERROR[29768] utils.c: fwrite() returned error: Broken pipe |
| 13:54.14 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 13:54.14 | billmania | I'm looking for an external device for creating WORM removable media. I need to create two copies simultaneously and will sacrifice write speed for increased data capacity and media durability. Anyone have any experience with such a device? |
| 13:55.07 | pabelanger | QuantumSchema: looks like an issue with your AMI script looking connection with asterisk |
| 13:57.10 | afink | ok, thanks leifmadsen |
| 13:58.32 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:58.32 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:03.12 | *** join/#asterisk zircote (~zircote@63.250.239.58) |
| 14:05.46 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:05.46 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:09.37 | *** join/#asterisk Polysics (~luca@host210-142-static.228-95-b.business.telecomitalia.it) |
| 14:09.38 | Polysics | how odd |
| 14:10.00 | Polysics | http://pastie.org/1916434 |
| 14:10.05 | Polysics | trying to get recording to work |
| 14:10.20 | Polysics | but apparently $RECORDED_FILE is empty |
| 14:10.20 | leifmadsen | what version of asterisk are you using |
| 14:10.25 | leifmadsen | that looks like Asterisk 1.2 |
| 14:10.28 | Polysics | me? 1.8 |
| 14:10.39 | leifmadsen | oh ya your dialplan is significantly out of date |
| 14:10.56 | leifmadsen | that's certainly not the right format for Record() anymore |
| 14:11.17 | Polysics | nice :-D |
| 14:11.25 | leifmadsen | [Syntax] |
| 14:11.25 | leifmadsen | Record(filename.format[,silence[,maxduration[,options]]]) |
| 14:11.40 | leifmadsen | ${RECORDED_FILE}: Will be set to the final filename of the recording. |
| 14:12.02 | leifmadsen | and you should not be using priority numbers, and you don't even need to specify the extension each time |
| 14:12.07 | leifmadsen | exten => 205,1,NoOp() |
| 14:12.11 | leifmadsen | same => n,App() |
| 14:12.15 | leifmadsen | same => n,App() |
| 14:12.16 | Polysics | so the problem is the dot for the format? |
| 14:12.26 | Polysics | that will save a lot of work |
| 14:12.26 | leifmadsen | Polysics: you tell me based on the information I have given you |
| 14:13.06 | Polysics | i need to learn how to read asterisk docs |
| 14:13.36 | leifmadsen | indeed |
| 14:13.37 | leifmadsen | ~thebook |
| 14:13.37 | 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 |
| 14:13.43 | leifmadsen | start with the dialplan chapters |
| 14:13.45 | raden_work | is it common for people to brute force attack asterisk ? |
| 14:13.46 | leifmadsen | lots of good info there |
| 14:13.53 | leifmadsen | raden_work: very |
| 14:14.03 | raden_work | were getting hit like mad lately :( |
| 14:14.19 | zkn | i'm sure some if not most of you here know how to overcome firewall/nat issues with SIP clients accessing Asterisk from different locations/networks.. could you help on the right track here with tip on what it takes to not having to configure every firewall/router to allow SIP signaling and voip traffic so it is possible to connect to the Asterisk server wherever you are and have internet connection? |
| 14:14.39 | leifmadsen | raden_work: that's why I use fail2ban |
| 14:14.58 | leifmadsen | zkn: I just enable nat=yes on the peer and that's it |
| 14:15.18 | leifmadsen | I never have to configure the firewall or router on the client side |
| 14:15.32 | leifmadsen | it just works |
| 14:15.32 | zkn | well, that does not seem to be enough |
| 14:15.35 | Polysics | hmm |
| 14:15.44 | Polysics | still getting Playback requires a filename |
| 14:15.58 | leifmadsen | use DumpChan() to see what channel variables you have access to |
| 14:16.12 | sezuan | leifmadsen: I assume it is sometimes necessary to use directmedia=yes... |
| 14:16.16 | leifmadsen | why not just save the filename you're going to record to into a channel variable, then use that for both Record() and Playback() ? |
| 14:16.26 | leifmadsen | sezuan: nope |
| 14:16.38 | leifmadsen | not for me -- I always send the media through asterisk |
| 14:17.01 | leifmadsen | the phone only ever has to worry about communicating with asterisk and never another device directly |
| 14:17.03 | sezuan | eh.. I mean = =no |
| 14:17.17 | leifmadsen | well I set directmedia=no in [general] |
| 14:17.20 | leifmadsen | I never use directmedia |
| 14:17.21 | Polysics | http://pastie.org/1916434 |
| 14:17.45 | leifmadsen | Polysics: use my suggestion per a few lines ago |
| 14:18.01 | leifmadsen | goes back to writing a blog post |
| 14:18.24 | sezuan | I guess default is directmedia=yes. I was a bit confused when I started with a clean config and the media was send directly, but all peers were nat=yes. |
| 14:19.09 | zkn | ok, so directmedia=no in general section and nat=yes for all peers |
| 14:20.33 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:22.10 | sezuan | zkn: I guess that should work in the most cases. |
| 14:23.34 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:24.38 | zkn | well, for some reason I still see errors about Packet timed out / SIP Retransmission timeout reached errors...and that has become especially frequent with 1.8 versions... not claiming that it is 1.8 issue, most probably still somewhere in configs |
| 14:26.43 | QuantumSchema | Arg... I've gotten no where... :-( |
| 14:27.03 | QuantumSchema | queue_log is still not being populated with a utils.c fwrite() error still being logged. |
| 14:28.42 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:29.50 | *** join/#asterisk hensema (~erik@scrat.hensema.net) |
| 14:30.32 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:31.11 | jacc0 | @leidmadsen: is there going to be a 1.8.5-rc1 today :) ? |
| 14:31.33 | jacc0 | *@leifmadsen |
| 14:31.53 | leifmadsen | jacc0: depends on if the issues blocking it get resolved |
| 14:31.59 | leifmadsen | you can start using 1.8 from the branch anytime you want though |
| 14:32.28 | leifmadsen | https://issues.asterisk.org/roadmap_page.php |
| 14:33.32 | *** join/#asterisk Khratos (~Khratos@66.128.60.148) |
| 14:33.55 | *** join/#asterisk skyion (~bradley@tcs-gw.bulwer.thusa.net) |
| 14:34.04 | Polysics | hmm |
| 14:34.14 | Polysics | still not working and i really do not get the rason |
| 14:34.22 | Polysics | i have moved the name of the file to a variable |
| 14:34.28 | Polysics | the file IS there but does not get playes |
| 14:34.32 | QuantumSchema | Is there a way to reload the module that is using util.c? |
| 14:34.32 | Polysics | *d |
| 14:34.42 | QuantumSchema | Or is that part of the core? |
| 14:35.21 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 14:35.33 | hensema | (possible faq) Does anybody have a script to transcode a mp3 to a supported format for music on hold? (my distro doesn't do mp3) |
| 14:36.30 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:37.46 | ferdna | hensema, add mp3 dupport |
| 14:37.51 | ferdna | support |
| 14:38.21 | hensema | I'd rather use standard packages in order to have automatic security updates |
| 14:38.41 | leifmadsen | use sox to convert the mp3 to ulaw or something supported natively by asterisk |
| 14:39.38 | skyion | Hey, I have a sip peer on asterisk server A with its callerid set in the sip.conf, there is also a sip trunk between server A and server B where calls breakout, however the source callerid is coming up as the source sip peer name at server B & the incorrect source callerid is being forwarded out to the telecoms provider.... |
| 14:40.00 | *** join/#asterisk Khratos (~Khratos@66.128.60.148) |
| 14:40.09 | Khratos | Good morning |
| 14:40.14 | skyion | any idea on how to ensure the source callerid stays the same? |
| 14:40.38 | leifmadsen | skyion: set it in the dialplan before the Dial() through the provider |
| 14:40.46 | *** join/#asterisk Exten (~Exten@mail.gdc.co.il) |
| 14:41.00 | leifmadsen | and use sendrpid=yes in sip.conf |
| 14:41.10 | Polysics | what's the proper wav setting for asterisk, please? 8khz 16bit mono? |
| 14:41.24 | leifmadsen | 8khz, 8-bit, mono |
| 14:43.09 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 14:46.00 | QuantumSchema | Weird... |
| 14:46.29 | QuantumSchema | removed my queue_log. touched a new queue_log. chown'ed the new queue_log to asterisk.asterisk. CLI> logger reload |
| 14:47.07 | QuantumSchema | I'm still getting an fwrite() error in the log but I'm logging to queue_log again. |
| 14:48.34 | *** join/#asterisk Exten (~Exten@mail.gdc.co.il) |
| 14:49.07 | coppice | leifmadsen: you should know better than that |
| 14:49.15 | Exten | its not 8khz 16bit mono ? |
| 14:49.26 | coppice | it is |
| 14:49.44 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:50.41 | leifmadsen | odd, I always convert to 8bit I'm pretty sure |
| 14:50.52 | leifmadsen | it's been months since I converted a file though |
| 14:51.02 | leifmadsen | and I've only had 1 cup of coffee and doing 8 things at once ;) |
| 14:51.15 | hensema | it's most probably 8 bit |
| 14:51.26 | leifmadsen | just try it and find out I guess :D |
| 14:51.31 | hensema | PSTN is 8 bit mono 8 khz ulaw (in europe anyways) |
| 14:53.03 | coppice | PSTN is 8 khz ulaw in the US and a few other places |
| 14:53.05 | coppice | PSTN is 8 khz Alaw in the Europe and most other places |
| 14:53.06 | coppice | If you want to record for wither of these you need to use 8k/second 16 bit mono |
| 14:55.26 | skyion | hmmm but I can't manually set it to one callerid as there are many behind that trunk.... |
| 14:55.29 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:56.37 | *** join/#asterisk zkn (~zkn@82.131.33.142.cable.starman.ee) |
| 14:56.49 | Polysics | is The Book about asterisk 1.8? |
| 14:57.23 | zkn | yes |
| 14:57.35 | Polysics | i need to basically re-learn it from scratch, and to actually learn something this go :-P |
| 14:58.08 | leifmadsen | Polysics: yes, it is based entirely around Asterisk 1.8 |
| 15:01.05 | Khratos | leifmadsen: good moring. Could you please remove these comments? (they have been inserted in the appropiated area) |
| 15:01.09 | Khratos | https://wiki.asterisk.org/wiki/display/AST/Changing+the+Verbose+and+Debug+Levels?focusedCommentId=13076653#comment-13076653 |
| 15:02.14 | Polysics | it is called a third edition, but it looks almost a different book to me :-D |
| 15:03.05 | leifmadsen | Polysics: it is :) totally rewrote from scratch |
| 15:03.22 | leifmadsen | O'Reilly liked using "3rd edition" for marketing purposes to show a time line |
| 15:04.03 | leifmadsen | Khratos: done |
| 15:06.11 | Khratos | Thanks. I could write down a couple of links which comments could be cleaned up while I'm reading the documentation and mail it to you. No problem with that? |
| 15:06.19 | leifmadsen | no problem with that |
| 15:06.22 | leifmadsen | lmadsen@digium.com |
| 15:07.12 | Khratos | Great |
| 15:12.21 | Polysics | is MeetMe still the recommended way to do conferences? |
| 15:13.50 | leifmadsen | Polysics: in 1.8 MeetMe() is more feature rich than ConfBridge(), but ConfBridge() requires less resources |
| 15:13.55 | leifmadsen | so depends what you need to do |
| 15:14.08 | leifmadsen | in asterisk trunk ConfBridge() has many more features |
| 15:14.17 | Polysics | i need to move two existing callers in a room and add a third |
| 15:14.23 | Polysics | basically turn a call into a conference |
| 15:14.33 | leifmadsen | you could use ConfBridge() for that |
| 15:16.28 | *** join/#asterisk diatonic (~diatonic_@mail.clearwater-research.com) |
| 15:18.10 | Polysics | i still need an extension for the room, and to transfer callers to that via AMI, i suppose |
| 15:18.21 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 15:20.40 | *** join/#asterisk Tim_Toady (~moi@188.4.51.59.dsl.dyn.forthnet.gr) |
| 15:24.06 | diatonic | Can recent Polycom Soundpoint IP phones be configured to auto-answer on the headset instead of speakerphone? With the proper alert info sent? |
| 15:24.59 | _Corey_ | diatonic: yes, you need to set headset memory mode if I'm not mistaken |
| 15:25.48 | Syrex | Anyone here have Polycom BLF + directed call pickup working using resource lists (as detailed here http://www.excaliburtech.net/archives/147)? |
| 15:26.52 | diatonic | _Corey_, thanks |
| 15:34.39 | *** join/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 15:43.41 | Aut0ExeC | can someone please help me with google voice for asterisk? |
| 15:48.25 | keith4 | Aut0ExeC: http://www.catb.org/~esr/faqs/smart-questions.html |
| 15:49.29 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:53.56 | Aut0ExeC | keith4: oh really? thanks man |
| 15:54.33 | Aut0ExeC | looks at keith4 with anger in his eyes |
| 15:55.09 | Aut0ExeC | keith4: dude i meant like when i set up google voice it asks for a physical phone to forward to... so how do I now redirect that to asterisk? |
| 15:55.26 | leifmadsen | Aut0ExeC: turn down the attitude |
| 15:55.32 | keith4 | dude, you need a physical phone number for asterisk |
| 15:55.43 | Aut0ExeC | howso? |
| 15:55.54 | leifmadsen | you need a number from an ITSP |
| 15:55.54 | keith4 | there used to be a way to connect google voice to Gizmo, and Gizmo to asterisk, but i think they stopped allowing/supporting that |
| 15:55.58 | leifmadsen | then it forwards to that number |
| 15:56.03 | Aut0ExeC | ohhhhhhhh |
| 15:56.08 | keith4 | also, even when it worked, the performance was shite |
| 15:56.24 | Aut0ExeC | leifmadsen: thats what i wanted to know earlier |
| 15:56.41 | Aut0ExeC | kinda defeats the purpose cuz if i have a number from itsp then why would I need another free number? |
| 15:56.52 | Aut0ExeC | thats bs |
| 15:57.03 | keith4 | the "how to ask a question" link isn't intended to be flippant; it *will* help you get your question answered, instead of being ignored |
| 15:57.15 | Aut0ExeC | keith4: ok thanks |
| 15:58.21 | Aut0ExeC | is there an itsp that I can get a free number wif? |
| 15:58.38 | keith4 | that would have been Gizmo |
| 15:58.48 | Aut0ExeC | ahh |
| 15:59.00 | Aut0ExeC | sux |
| 15:59.27 | keith4 | you can get a DID for like $2/month, though |
| 15:59.32 | keith4 | ~itsp |
| 15:59.32 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 16:00.36 | Aut0ExeC | keith4: never seen 2 bucks a month bro |
| 16:00.39 | Aut0ExeC | thats crazy |
| 16:02.21 | keith4 | see, for example, flowroute |
| 16:02.41 | Aut0ExeC | ok ok |
| 16:02.44 | Aut0ExeC | i'll check it out now |
| 16:02.55 | keith4 | msg infobot "itsplist-us" for more ideas |
| 16:03.05 | *** join/#asterisk catphish (~catphish@vpn.atechmedia.net) |
| 16:03.49 | catphish | when making an outgoing call through a sip provider, how can i set the outgoing caller id? |
| 16:04.14 | Aut0ExeC | keith4: thnks |
| 16:04.36 | catphish | i have tried Set(CALLERID(all)="Name" <000000>) in the outgoing extension |
| 16:04.58 | catphish | which appears to set the callerid of the internal sip channel, but doesn't appear to be passed through to the sip provider |
| 16:05.32 | Chainsaw | catphish: Generally the outgoing caller ID is filtered. |
| 16:05.43 | catphish | mine isn't |
| 16:05.45 | Chainsaw | catphish: All telcos do this, and I would be most surprised if SIP providers are any exception. |
| 16:06.03 | catphish | unless it is, and my provider made a mistake |
| 16:06.15 | Chainsaw | catphish: Unless you have specifically negotiated this capability, you don't have it. |
| 16:06.25 | Chainsaw | catphish: And even if you did, I would expect you to be limited to a *very* specific range. |
| 16:06.42 | Chainsaw | catphish: So make sure you keep to that range, or they will throw away what you send. |
| 16:06.42 | catphish | Chainsaw: my provider told me that he hadn't yet implemented the restriction, and i was free to set it |
| 16:06.55 | Chainsaw | catphish: They said what you wanted to hear so you would pay them. |
| 16:07.05 | catphish | lol i doubt that |
| 16:07.27 | Chainsaw | catphish: Of course. All business is honest and 100% transparent. |
| 16:07.28 | catphish | can we get back to the question at hand anyway, if it turns out it's his mistake i'll get him to fix it |
| 16:08.07 | catphish | i'll get a packet trace of the sip session to see what asterisk is sending in the initiation |
| 16:09.36 | catphish | i don't see why my provider would lie about such a thing, he only knows i'd be back to whine :) |
| 16:10.30 | Chainsaw | catphish: Absolutely. Setting arbitrary caller ID is normal and common practice. That's why you get all those prank calls from the white house. |
| 16:10.39 | *** join/#asterisk marlowe (~marlowe@static-72-66-8-138.washdc.fios.verizon.net) |
| 16:11.01 | catphish | of course it's normal |
| 16:11.06 | catphish | that's how caller id works |
| 16:11.18 | catphish | if people couldn't send them, it would never work |
| 16:12.22 | Polysics | do you recommend using realtime for config files? |
| 16:12.38 | leifmadsen | I recommend implementing what is necessary to solve the problem |
| 16:12.44 | catphish | i think i see the problem, asterisk is setting my from address to my sip username instead if my callerid number |
| 16:13.39 | Polysics | then i would say "no, you do not recommend that" |
| 16:13.41 | Polysics | :-D |
| 16:13.59 | Chainsaw | Polysics: It is an additional thing to go wrong, yes. |
| 16:15.30 | leifmadsen | Polysics: I didn't say that |
| 16:15.36 | leifmadsen | in some cases I would certainly recommend it |
| 16:15.40 | leifmadsen | I wouldn't add it just for fun though |
| 16:15.43 | Polysics | Chainsaw, i will be having peers in realtime |
| 16:15.53 | Polysics | leifmadsen, can i ask for an use case, please? |
| 16:16.05 | leifmadsen | distributed systems |
| 16:16.28 | Polysics | such as load balanced situations? |
| 16:17.02 | leifmadsen | that could be one purpose of a distributed system, yes |
| 16:18.35 | Polysics | or failover |
| 16:18.45 | Polysics | hmm, it does look like i have to at least take a look |
| 16:18.51 | Polysics | dang .-) |
| 16:18.53 | keith4 | catphish: are you able to set the name? |
| 16:19.00 | keith4 | er.... catphish: are you able to set the number? |
| 16:19.11 | keith4 | some itsps only allow setting the CID number, not the name |
| 16:19.42 | catphish | keith4: the problem is probably that i don't understand SIP well though |
| 16:20.39 | catphish | currently asterisk is sending the string: From: "Charlie" <sip_username@sip.provider.com>;tag=nnnnnn |
| 16:20.59 | catphish | is that where the callerid number should be sent? |
| 16:21.09 | catphish | or is there an alternative field i should be setting? |
| 16:21.27 | catphish | if i *don't* send username@sip.provider.com in the number field the call gets rejected |
| 16:22.02 | keith4 | try something like: exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=1234567890) |
| 16:22.16 | catphish | that's exactly what i'm doing |
| 16:22.34 | paulc | catphish do you have a fromuser in your peer definition? |
| 16:22.46 | catphish | but it's being overridden by: fromuser= |
| 16:22.50 | catphish | yes i do |
| 16:22.55 | catphish | if i remove it, the call is rejected |
| 16:22.56 | *** join/#asterisk fauxalliance (~fauxallia@142.163.152.119) |
| 16:23.19 | keith4 | sounds like you should ask your provider some questions, then |
| 16:23.30 | catphish | how *should* this work? |
| 16:23.55 | catphish | am i right in thinking my provider should identify me by ip address instead on my from: sip string? |
| 16:26.52 | Polysics | is ODBC recommended over straight MySQL driver? |
| 16:27.09 | leifmadsen | I recommend it yes |
| 16:27.35 | Chainsaw | Oh, âI actually use the straight MySQL one. |
| 16:27.37 | Polysics | i am certain i will not be using any other db. is there any other reason? the book does not state a preference but only explains ODBC, so it is implied :-D |
| 16:28.08 | catphish | thanks keith4 my isp is taking a look |
| 16:28.43 | paulc | catphish: I think one of my providers relies on fromuser for auth, but lets me Set(CALLERID(num)=xxxxxx) (without touching the name part) - worth a try maybe |
| 16:29.16 | catphish | it seems that my provider uses the from number for auth |
| 16:29.42 | catphish | but maybe i can set the from name to a number |
| 16:29.46 | Polysics | in any case, i just configure res_mysql.conf, rest stays the same, right? |
| 16:31.02 | keith4 | catphish: dunno. my provider uses a username/secret for auth |
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| 16:48.32 | Polysics | does 1.8 not have res_mysql.conf by default? |
| 16:48.39 | Polysics | or the file does not exist? |
| 16:48.55 | Polysics | it is not reading the musiconhold.conf stuff i put in the db |
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| 16:50.13 | jacc0 | @leifmadsen:thnk (a bit late) |
| 16:50.14 | jacc0 | bye all |
| 16:51.10 | Polysics | or is the config in res_config_mysql? i am confused |
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| 17:05.00 | catphish | keith4: fixed it, my ISP uses the from header for auth and requires a 'Remote-Party-ID' header for callerid |
| 17:05.30 | catphish | and can someone tell chainsaw to stop being so arrogant when he gets back |
| 17:05.56 | catphish | most ISPs do filter caller ID, but they can't filter it if its not sent! |
| 17:06.09 | catphish | thanks all |
| 17:06.15 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:06.15 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:07.01 | Jcook_5xData | is it possible for asterisk not honor resetinterval=never I have set this but I still get random resets |
| 17:07.47 | *** join/#asterisk brainiac (~mhauss@c-68-52-248-20.hsd1.tn.comcast.net) |
| 17:14.52 | Jcook_5xData | I have check all over google but I have not found any good resources |
| 17:15.24 | *** join/#asterisk Diffen (~diffen@user8.85-195-0.netatonce.net) |
| 17:18.16 | malcolmd | if they're randomly occuring then there's probably a connection problem w/ the link. if that's a digium card you've got in there, you can contact our support team: http://www.digium.com/support |
| 17:19.38 | Jcook_5xData | I did see this when i did a pri debug > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1)> Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) (2) ] |
| 17:20.36 | Jcook_5xData | I thought it was weird to Network Congestion on a pri when there only about 5 people on the phone at a time |
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| 17:24.11 | WIMPy | You could ask your provider, what that temporary failure means. |
| 17:24.40 | keith4 | Jcook_5xData: you're not sharing that as a data T1, like with a Cisco ISR, are you? |
| 17:25.07 | Jcook_5xData | no straight pri |
| 17:26.50 | Jcook_5xData | I am wondering if I should replace the card as well as contact ATT |
| 17:31.40 | *** join/#asterisk f8l_0e (~jbarthle@12.190.216.66) |
| 17:35.16 | f8l_0e | I am setting up an * server with a 4 line analog card. The phones are 4 line cisco spa942. There are going to be more extensions than lines. Is there a way that I can setup shared line appearance for all the line keys so they can all access each line, but still set up extensions? |
| 17:35.17 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 17:44.27 | ChannelZ | Why use the concept of lines at all? |
| 17:45.28 | f8l_0e | This is the first time I'm working with an analog card. I'm used to SIP trunks. What would you suggest? |
| 17:45.52 | ChannelZ | If you have more extensions that physical phone lines, I assume none of those phone numbers belong to any particular person |
| 17:46.22 | f8l_0e | Correct. They are just multiple lines for anyone's use. |
| 17:46.28 | ChannelZ | For dialing out, put all 4 channels in a group and dial the group. Asterisk will use the next free line |
| 17:48.06 | f8l_0e | They will need to transfer calls to an extension instead of being able to directly pick up a line then, correct? |
| 17:49.19 | ChannelZ | Yeah - or use call parking. How do people tell each other they have a call now? Page? Yell down the hall? |
| 17:49.45 | f8l_0e | They have an AT&T merlin pbx that uses line keys. |
| 17:49.58 | f8l_0e | Intercom |
| 17:51.43 | sxpert | f8l_0e: that's like 30+ years old ? |
| 17:51.48 | f8l_0e | Yep. |
| 17:52.14 | ChannelZ | I just personally think treating the physical phone lines in an old manner prevents you from future flexibility. |
| 17:52.30 | f8l_0e | My friend bought a doctors office with all the existing eqp. |
| 17:52.42 | f8l_0e | Than ChannelZ. |
| 17:52.46 | f8l_0e | Thanks |
| 17:52.53 | ChannelZ | You could mix those 4 lines with 10 more SIP channels in which case having 4 buttons on the phone doesn't mean a lot |
| 17:53.14 | f8l_0e | Putting the lines together in a group is done in the dahdi.conf, right? |
| 17:53.44 | ChannelZ | chan_dahdi.conf yes |
| 17:53.57 | puzzled | with 1.8 trunk is there a way I can build an individual module or chan without having to rebuild everything? |
| 17:54.10 | ChannelZ | In any event aren't those 4 buttons just 4 separate SIP accounts? (I'm using SPA922s) |
| 17:55.13 | ChannelZ | puzzled: if you have the source and build environment that matches how the rest of it was built in the first place it will only rebuild the bits you've tweaked on.. but if you're running a * package it's fairly difficult |
| 17:55.59 | puzzled | ChannelZ: I'm using src and the same build env |
| 17:56.30 | puzzled | can't imagine there isn't some neat make command to build an individual module |
| 17:56.48 | f8l_0e | No, I'm using a 4 port digium analog card. |
| 17:58.23 | ChannelZ | puzzled: there is.. 'make' |
| 17:58.47 | puzzled | ChannelZ: yup found that one :) |
| 17:58.54 | ChannelZ | Unless you modified every single module since the last time you built it |
| 18:00.08 | ChannelZ | And in a lot of cases it's not possible to just build one module, a bunch of stuff depends on each other. Unless you're on a 386 or something the whole thing doesn't take that long to build anyway, a few minutes maybe. From then on if you're hacking on a module it will only rebuild what has changed |
| 18:00.21 | f8l_0e | Wait, you're right. I created 4 sip accounts for the phones to register to. |
| 18:01.09 | ChannelZ | f8l_0e: so you actually have to go further out of your way to make those line keys behave like old analog line selection keys. So my thinking is, why bother :) |
| 18:01.58 | f8l_0e | Very true and thanks again. My friend is gonna kick my @$$ for making him buy 4 line phones. :) |
| 18:02.43 | ChannelZ | The only thing I think it would be useful for is to be able to glance at the phone and see that all 4 analog lines were 'in use' and that it'd be pointless to pick up the phone to try and call someone |
| 18:03.33 | *** join/#asterisk luckman212_ (~irc@2001:470:1f07:1225:6872:6022:77a6:8f23) |
| 18:04.39 | *** join/#asterisk fullstop (~fullstop@static-173-210-91-4.saucontech.com) |
| 18:04.46 | ChannelZ | Is there a secretary who answers all calls and directs them or will it be IVR driven? |
| 18:04.56 | fullstop | Good afternoon |
| 18:05.32 | f8l_0e | Anyone in the office that is available will pick up. No IVR. But I will be setting up VM. |
| 18:05.57 | f8l_0e | Hi fullstop. |
| 18:06.51 | *** join/#asterisk timahvo1 (~rogue@41.223.57.76) |
| 18:07.23 | fullstop | Any idea how peer selection in sip.conf works? I have a situation where my telco requires authentication for outbound calls but can not use authentication for inbound calls. |
| 18:08.01 | fullstop | When I configure an outgoing peer and an incoming user, asterisk matches the peer by ip address, and ends up using the wrong peer and rejects the call. |
| 18:08.27 | fullstop | I suspect that it is because there is no user information in the incoming call, so it matches by ip address. |
| 18:12.11 | f8l_0e | did you set up the context in your sip providers section of sip.conf |
| 18:12.13 | f8l_0e | ? |
| 18:12.31 | fullstop | f8l_0e: it rejects the call well before that. |
| 18:13.11 | f8l_0e | what do you see on the console when the attempted inbound call comes in? |
| 18:13.42 | fullstop | http://pastebin.com/2d1PhYQF |
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| 18:14.21 | fullstop | It picks the "best" peer, which happens to be the wrong one. |
| 18:14.40 | f8l_0e | Is that a wireshark capture? |
| 18:14.46 | fullstop | since I must have 2 defined with the same ip address -- one for incoming calls w/out digest auth and one for outbound w/digest auth. |
| 18:18.10 | fullstop | f8l_0e: no. sip debug from asterisk |
| 18:19.18 | kaldemar | fullstop: use a single peer and set insecure=port,invite for it to not authenticate incoming calls. |
| 18:19.28 | f8l_0e | I don't think you should need two peer entries in the sip.conf for one provider. |
| 18:20.32 | fullstop | kaldemar: I will try that. |
| 18:20.54 | kaldemar | more information on how the peer matching works can be found in the sample sip.conf foe example. |
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| 18:26.46 | f8l_0e | fullstop: http://pastebin.com/HpEsde0d |
| 18:27.37 | f8l_0e | obviously there are other settings under your peer and dialplan that you'll want to do. |
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| 18:32.42 | fullstop | f8l_0e: I tried that -- outgoing works fine.. incoming not so much. |
| 18:34.22 | f8l_0e | but once you set up just that one and it connects to your voip provider, they know which ip to send back to and asterisk routes those back to the same peer entry. |
| 18:34.55 | f8l_0e | are you behind a firewall on the asterisk server? |
| 18:36.00 | f8l_0e | nevermind. a firewall isn't a problem if you see the incoming attempt. |
| 18:36.33 | kaldemar | fullstop: how does it not work? what do you see in the CLI? |
| 18:36.41 | fullstop | one moment |
| 18:37.03 | fullstop | I am behind a firewall, yes. A Cisco ASA5510 |
| 18:37.13 | Sertys | yo |
| 18:37.34 | Sertys | i wanna build up a pool of mobile phones to use via chan_mobile |
| 18:37.46 | Sertys | the last time i tested chan_mobile, it was kinda crappy |
| 18:37.57 | fullstop | http://pastebin.com/1sBqkb7f |
| 18:38.07 | *** join/#asterisk m4xx (~m4xx@75-144-154-165-NewEngland.hfc.comcastbusiness.net) |
| 18:38.11 | ChannelZ | Get iPhones. They belong at the bottom of a pool |
| 18:38.20 | *** join/#asterisk oliver1 (~oliver@manz-5f74a9a4.pool.mediaWays.net) |
| 18:38.23 | WIMPy | What about gateways? |
| 18:38.32 | fullstop | I'll be back in a minute |
| 18:38.38 | *** join/#asterisk grabes (~grabes@209.183.177.102) |
| 18:38.53 | Sertys | lol |
| 18:39.00 | grabes | Anyone know if SLA is working with Yealink phones on 1.8 before I get too into this? |
| 18:39.00 | Sertys | iphones rock |
| 18:40.45 | _Corey_ | Sertys: We had a bluetooth gizmo that plugged into an FXO port at one point, I'm not sure what brand it was |
| 18:41.09 | _Corey_ | you could theoretically pair it to anything and plug in as many of those as you want w/o chan_mobile |
| 18:41.29 | f8l_0e | ChannelZ: I temporarily configure the 942s manually through the web interface. I want to provision them from ftp in the future, but I'm having a difficult time finding the documentation for the xml files, do you have a link? |
| 18:42.15 | Sertys | _Corey_: have u got any idea what's the cheapest sim box for like 4-8 sim cards? |
| 18:42.40 | WIMPy | Goto(ebay) |
| 18:42.48 | _Corey_ | not really, like i said it was a bluetooth device so it was 1-1 |
| 18:42.55 | Sertys | yeah |
| 18:43.02 | Sertys | i got the point of it |
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| 19:16.07 | f8l_0e | fullstop: you there? |
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| 19:18.34 | drfreeze | Anyone know how to set LBO on a WCTDM card? |
| 19:23.29 | f8l_0e | freeze: Does this help? http://www.cadvision.com/blanchas/Asterisk/ConfiguringT1LineDahdi.html |
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| 19:24.03 | fullstop | f8l_0e: yes |
| 19:24.53 | f8l_0e | fullstop: did you get your inbound working? |
| 19:25.30 | drfreeze | f8l_0e: that is for a T1. Any idea how to set that for a TDM analog line? |
| 19:26.07 | fullstop | f8l_0e: negative |
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| 19:26.43 | fullstop | if I am reading it correctly, my telco's peer is rejecting me. |
| 19:26.55 | fullstop | or, rejecting my response |
| 19:27.27 | f8l_0e | instead of the debug view, have you seen what the cli shows? |
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| 19:31.28 | fullstop | f8l_0e: sometimes looking at things too closely has its disadvantages. |
| 19:31.45 | fullstop | f8l_0e: my problem became excruciatingly clear after doing that. |
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| 19:32.43 | f8l_0e | figured it out fullstop? |
| 19:33.20 | fullstop | yes. Many thanks. |
| 19:33.27 | f8l_0e | you're welcome |
| 19:34.19 | f8l_0e | freeze: everything I see show LBO being specific to PRI interfaces, not PSTN. |
| 19:34.38 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:36.36 | f8l_0e | or maybe not. |
| 19:36.52 | citywok | who loves microsoft dynamics crm?!?!? <-- not this guy |
| 19:36.58 | f8l_0e | freeze: what do you want to set the LBO to? |
| 19:39.28 | *** part/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 19:41.12 | f8l_0e | it should just be lbo = x under the span that you are configuring in /etc/dahdi/system.conf |
| 19:41.42 | f8l_0e | how many feet from the demarcation point are you? |
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| 19:50.05 | malcolmd | line build out doesn't apply to the analog cards. if you've got a gain issue, you can set them up or down using rxgain and txgain in chan_dahdi.conf |
| 19:51.30 | malcolmd | there are also module parameters for the wctdm24xxp driver for tx and rx gains for fxs and fxo interfaces, if you prefer to go that route |
| 19:52.08 | f8l_0e | malcolmd: I believe you, but why do I get the following on dahdi show status |
| 19:52.13 | f8l_0e | phonesrv*CLI> dahdi show status |
| 19:52.13 | f8l_0e | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 19:52.13 | f8l_0e | Wildcard AEX410 Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 19:52.53 | malcolmd | doesn't matter; line build out doesn't apply to the analog cards. |
| 19:53.26 | f8l_0e | ok. good to know. hopefully it helps drfreeze out. |
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| 20:09.03 | drfreeze | malcolmd: ok. I thought that might be the case |
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| 20:19.04 | malcolmd | no worries :) |
| 20:19.48 | *** join/#asterisk ideaman (~ihaveapla@c-98-202-74-138.hsd1.ut.comcast.net) |
| 20:20.57 | ideaman | anyone have frequent registration issues with Flowroute? |
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| 20:31.18 | leifmadsen | never heard of Flowroute :) |
| 20:32.13 | f8l_0e | i'm using them. i haven't had a registration problem in a couple of months. |
| 20:32.42 | ideaman | They just mentioned a SIP ALG setting on my router being on possibly causing that. I've never had to mess with that one. |
| 20:33.34 | f8l_0e | is your router sip aware? |
| 20:35.34 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 20:36.03 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:36.05 | f8l_0e | if you set the failover route on your flowroute admin page to your asterisk server's ip, the calls will still come through even if the registration failed. |
| 20:49.41 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 20:50.00 | anonymouz666 | no lucky using asterisk-1.8.4... it is crashing when loading dynamic modules! |
| 20:50.14 | anonymouz666 | it's...bizarre :/ |
| 20:50.57 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 20:54.21 | f8l_0e | 666: you could always set up a noload => whatever.so for each module then comment out one at a time |
| 20:54.54 | WiretapWork | anonymouz666, did you remove the old modules before you installed the new version? |
| 20:55.05 | WiretapWork | I found that sometimes, failing to do this can cause thatr |
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| 21:02.31 | anonymouz666 | WiretapWork: you are probably right |
| 21:02.35 | anonymouz666 | will remove |
| 21:03.21 | WiretapWork | anonymouz666, also, I'm not sure if its still important, but noload the IMAP and voicemail ODBC modules. |
| 21:09.17 | *** part/#asterisk grabes (~grabes@209.183.177.102) |
| 21:11.20 | leifmadsen | WiretapWork: well that would only be applicable in prebuilt packages like RPMs where you have separate modules, but when compiling app_voicemail you only get one module and you either get ODBC, or IMAP, or FILE support in voicemail (not separate modules) |
| 21:12.52 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 21:13.25 | WiretapWork_ | leifmadsen, yep, well, not just RPM but the factory .deb install from the debian repos too |
| 21:13.32 | WiretapWork_ | stupid IRC client borked on me |
| 21:13.42 | WiretapWork_ | I got your message but the main window was frozen XD |
| 21:14.05 | *** join/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 21:14.16 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 21:14.33 | leifmadsen | WiretapWork_: right that was my point |
| 21:14.45 | WiretapWork_ | yep |
| 21:14.51 | WiretapWork_ | and anon installed from source? |
| 21:15.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:19.30 | gavimobile | folks, im upgrading my asterisk, and when in shell I typed " remove asterisk16 asterisk16-core" however I get in return Package(s) asterisk16 available, but not installed. |
| 21:20.07 | gavimobile | how can I find out the correct package name to remove? |
| 21:20.25 | ferdna | gavimobile, what distro? |
| 21:20.34 | gavimobile | centos5 |
| 21:21.26 | ferdna | gavimobile, are you in the console or graphical mode? |
| 21:21.52 | gavimobile | consol |
| 21:21.59 | gavimobile | via ssh |
| 21:22.27 | gavimobile | ferdna: this is what im trying to do. http://www.1oc.com/blog/?p=334 |
| 21:23.06 | ferdna | gavimobile, do this: rpm -qa | less |
| 21:23.14 | ferdna | look for package name |
| 21:23.17 | gavimobile | ferdna: in shell? |
| 21:23.23 | ferdna | yes |
| 21:24.04 | gavimobile | it gives me a list of comman ds |
| 21:24.17 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:24.23 | ferdna | gavimobile, not in yum shell |
| 21:24.25 | ferdna | in console |
| 21:24.29 | ferdna | shell |
| 21:25.28 | gavimobile | ok I got a list of packages |
| 21:25.58 | ferdna | look for the one you want |
| 21:26.03 | ferdna | asterisk something |
| 21:27.35 | tzafrir | gavimobile, rpm -qf /usr/sbin/asterisk |
| 21:28.43 | ferdna | gavimobile, once you know the package name |
| 21:28.52 | gavimobile | tzafrir: hey, thanks. I found that package in the list, asterisk16-core-1.6.2.17.3-1_centos5. however that list is forever long. im still looking for the other package |
| 21:29.00 | ferdna | just do "rpm -e foobar" |
| 21:29.10 | ferdna | foobar your package |
| 21:29.20 | gavimobile | ferdna: can I use a wildcard? |
| 21:29.26 | gavimobile | rpm -e foo* |
| 21:29.52 | ferdna | gavimobile, i am not sure... i run slackware... and asterisk on top... different package tools |
| 21:29.53 | *** join/#asterisk jkroon (~jkroon@dsl-242-4-146.telkomadsl.co.za) |
| 21:30.07 | gavimobile | I see :-/ |
| 21:30.08 | tzafrir | gavimobile, in grep |
| 21:30.28 | gavimobile | tzafrir: could you give me an example bro? |
| 21:30.29 | tzafrir | gavimobile, rpm -qf checks the package a file should belong to |
| 21:30.41 | ferdna | gavimobile, http://linux.die.net/man/8/rpm |
| 21:31.29 | ferdna | look at: Erase Options |
| 21:32.48 | gavimobile | ferdna: im sure this manual is really simple to understand, I just don't have enough brainpower to put it together right now.. maybe cause the hour :-p |
| 21:32.53 | gavimobile | thanks man! |
| 21:35.05 | ferdna | your welcome :) |
| 21:36.07 | leifmadsen | you're* :) |
| 21:36.37 | WiretapWork_ | ahahaha, so glad I'm not the only one who does that |
| 21:42.22 | gavimobile | is this the second package which I need ? asterisknow-version-1.7.1-2_centos5 |
| 21:44.08 | gavimobile | I think it was it |
| 21:46.49 | *** join/#asterisk oliver1 (~oliver@manz-5f74a9a4.pool.mediaWays.net) |
| 21:49.40 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 21:50.22 | Micc | Is there any advantage to running x64 for the OS that asterisk will run on? I've always done 32 bit before. It never uses up all the addressable memory anyways. |
| 21:54.18 | *** part/#asterisk gavimobile (~user@bzq-84-108-104-165.cablep.bezeqint.net) |
| 21:58.59 | pigpen | Hi all. I am having an issue with an Audiocodes issue. Reference pastbin: http://pastebin.com/7uPEJeQA |
| 21:59.39 | pigpen | Setup is polycom 650 -->Asterisk 1.8.3.3---> Audiocodes MP114 FXO---> 4 PSTN lines (ATT) |
| 22:00.02 | pigpen | Now and then, while dialing out, calls will get a busy tone. |
| 22:00.12 | pigpen | it will last for around 2 minutes. |
| 22:00.26 | pigpen | but no more. |
| 22:00.45 | pigpen | After this time, the same number, from the same polycom works fine. |
| 22:01.33 | pigpen | During the busy tone "time", usage of a cell phone verifies the remote end is available. |
| 22:01.59 | pigpen | For that matter, the remote is a very large corporation and busy signals would be very unlikely. |
| 22:04.58 | Chainsaw | pigpen: Missing disconnect supervision resulting in stuck ports? |
| 22:05.08 | Chainsaw | pigpen: With busydetect, it would take 3 or so goes for a port to clear. |
| 22:05.40 | *** join/#asterisk rogersja (~RogersJa@S0106000f6695039f.gv.shawcable.net) |
| 22:06.55 | pigpen | when this happens, all channels are not in use. |
| 22:07.24 | pigpen | by verifying "core show channels verbose" in asterisk and by visual confirmation physically at the unit. |
| 22:08.08 | pigpen | I have pulled full log files and the asterisk event and sent to the distributor, but things are taking much longer than it should. |
| 22:08.31 | pigpen | hence why I am here hoping to expedite the resolution. |
| 22:08.44 | pigpen | Chainsaw, thanks for the idea though. |
| 22:09.03 | Chainsaw | pigpen: So you have port status LEDs? |
| 22:09.18 | pigpen | yeah, all dark when this happnes (ie: on hook) |
| 22:09.26 | Chainsaw | pigpen: Right, so it doesn't even dial out. |
| 22:09.36 | pigpen | right. it fails at the audiocodes. |
| 22:09.45 | pigpen | like it can't even grab an fxo. |
| 22:09.49 | Chainsaw | pigpen: Do you have SNMP MIBs with port status perhaps? |
| 22:10.09 | pigpen | that would be dam handy... |
| 22:10.17 | pigpen | hmm.... |
| 22:10.30 | Chainsaw | pigpen: My Pattons give me that for the ISDN BRIs. I rely on it for knowing when Openreach engineers tug on my wires. |
| 22:10.43 | Chainsaw | pigpen: Or when my circuit erodes away again. It does that about once a year. |
| 22:11.17 | pigpen | yeah, not a bad idea. |
| 22:11.35 | pigpen | I am sending all debug to syslog, which is dam nice compared to the web access view. |
| 22:19.40 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 22:20.59 | leifmadsen | Asterisk 1.4.42-rc1 Now Available | Asterisk http://j.mp/jmRNQX |
| 22:25.07 | *** join/#asterisk g_r_eek (~g_r_eek@p50894F48.dip0.t-ipconnect.de) |
| 22:31.22 | Jcook_5xData | anyone have a good explaining pri debuging |
| 22:31.30 | Jcook_5xData | link* |
| 22:34.59 | Chainsaw | Jcook_5xData: Red alarm, you can't see them. Yellow alarm, they can't see you. Blue alarm, they can't talk to the network. |
| 22:36.13 | Chainsaw | Jcook_5xData: Once you're out of alarm state, route calls. |
| 22:36.25 | Jcook_5xData | Chainsaw, lol no i mean this stuff http://pastebin.com/ZMhVM572 |
| 22:36.30 | Chainsaw | That's ISDN PRI in two sentences. Enjoy. |
| 22:36.59 | Chainsaw | Jcook_5xData: You've redacted your trace, so I can't help you. |
| 22:37.14 | Chainsaw | That's enough for today. Thank you very much, good night. |
| 22:37.25 | Jcook_5xData | my pri random reset |
| 22:41.35 | *** join/#asterisk lcat (~lcat@187.45.253.214) |
| 22:42.57 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:6872:6022:77a6:8f23) |
| 22:45.08 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:03.23 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 23:03.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 23:09.34 | *** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar) |
| 23:11.24 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v015-215.mobile.uci.edu) |
| 23:11.56 | fofware | hello, any one play with JabberSend from manager API, I'm trying to send msgs, but it not work, If I send they pbx_lua they work |
| 23:12.26 | fofware | can any one give a tip? |
| 23:15.09 | paulc | fofware: what does your dialplan for sending look like (when it doesn't work) and what about jabber.conf? (use pastebin) |
| 23:15.20 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 23:18.08 | fofware | paulc: I'm trying send it from a script out from dial plan |
| 23:18.45 | fofware | paulc: because in dial_plan Only know if i receive a call |
| 23:19.43 | paulc | fofware: So show us your dial plan and maybe someone can help.... without seeing it I can't comment on what you're doing right or wrong |
| 23:20.28 | fofware | paulc: so with script connected to manager at port 5038 I know when someone write in his/her client |
| 23:20.49 | fofware | paulc: ok |
| 23:21.20 | paulc | fofware: You're saying you want Asterisk to receive Jabber messages from other people, and output them via AMI? |
| 23:21.44 | fofware | paulc: Yes, right |
| 23:22.48 | paulc | fofware: I'm not sure that's supported. I saw a demo once of reacting to messages in the dialplan but it may have been after a JabberSend initiated the conversation. |
| 23:23.16 | paulc | fofware: you might be better off with something external talking to the Jabber server, then triggering Asterisk as needed when something relevant happens (either via call file, AMI, whatever) |
| 23:23.26 | fofware | paulc: AMI have a JabberSend function |
| 23:24.05 | paulc | That's for sending messages to Jabber end points, from Asterisk. You're asking for the other way round I thought? |
| 23:24.13 | fofware | paulc: yes, maybe better run a bot |
| 23:26.45 | paulc | yeah, that's probably a better way to go I think |
| 23:28.58 | fofware | paulc: any way here is my script in lua http://pastebin.ca/2062868 |
| 23:38.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:42.10 | fofware | well any one know some bot for read jabber messages... before googling it :)} |
| 23:44.31 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053225.ics.uci.edu) |