| 15:02.28 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:02.28 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.1 (2012/03/15), 1.8.10.1 (2012/03/15), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:09.17 | *** part/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 15:16.02 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:17.43 | p3nguin | I have a couple channels that have been stuck for many days. I tried to bridge them and then hangup, but it turned them into zombies. Now I have two zombie channels for a while. Is there any way to purge channels without restarting asterisk? |
| 15:22.19 | anonymouz666 | channel request hangup? |
| 15:22.40 | anonymouz666 | if I remember correctly, there's an aliases for soft hangup |
| 15:22.59 | *** join/#asterisk davidduffy (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 15:23.49 | *** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226) |
| 15:40.59 | p3nguin | Let me rephrase... |
| 15:41.31 | *** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2) |
| 15:41.52 | p3nguin | I have a couple channels that have been stuck for many days. channel request hangup and channel redirect do not affect them. I tried to bridge them and then hangup, but it turned them into zombies. Now I have two zombie channels for a while. Is there any way to purge channels without restarting asterisk? |
| 15:46.57 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 15:47.56 | MrTelephone | Is there a way to force sip trunks from another asterisk to authenticate by HOST only no matter what the callerid of the call is? I had it configured but it doesn't work when there are matching peers on both sides. |
| 15:48.16 | p3nguin | Perhaps you mean sip peers. |
| 15:48.28 | MrTelephone | penguin whats up buddy? |
| 15:48.36 | p3nguin | Peers never authenticate by caller id. |
| 15:48.53 | MrTelephone | should I set all my sip accounts to peer instead of friends? |
| 15:49.01 | *** join/#asterisk TSM (~the_softw@78-105-6-158.zone3.bethere.co.uk) |
| 15:49.05 | p3nguin | Choices are username, IP/port, or no auth check. |
| 15:50.02 | p3nguin | Well, wait, I really shouldn't call it "auth" at this stage. |
| 15:50.14 | MrTelephone | ip/port auth doesn't take priority |
| 15:50.30 | p3nguin | Once the call matches one of your peers by username or IP/port, then asterisk asks for authentication if needed. |
| 15:50.33 | MrTelephone | Maybe I should be trying to use redirect for redundancy? |
| 15:51.26 | MrTelephone | I don't even need the call being trunked through asterisk. I could tell the client to send the invite to another proxy.. |
| 15:52.22 | p3nguin | In the peer entry, set the host to the correct IP address for the peer, don't set a secret. |
| 15:52.36 | p3nguin | And use type=peer. |
| 15:53.10 | MrTelephone | I did but when peer 1111 was on asteriskA and asteriskB, asteriskB was asking for auth and the sip trunk was failing |
| 15:53.29 | MrTelephone | i can try using peer again for user 1111 |
| 15:54.12 | MrTelephone | Forward() seems like it would work and then there won't be any extra rtp traffic |
| 15:54.23 | p3nguin | I'd imagine it was because there is no such thing as sip trunking. |
| 15:54.29 | p3nguin | That always messes up things. |
| 15:55.05 | MrTelephone | TDMoe might be the best for clustering then |
| 15:56.36 | MrTelephone | Forward wouldn't work because I wouldn't be able to redirect the call to PRI#2 |
| 16:02.20 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v006-207.mobile.uci.edu) |
| 16:02.32 | *** join/#asterisk cvance (~cvance@lafreniere.mygvllc.com) |
| 16:14.25 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 16:28.16 | *** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net) |
| 16:32.26 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:46.15 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-113-188.cust.dsl.vodafone.it) |
| 16:47.15 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:56.53 | *** join/#asterisk ipengineer (~ipenginee@static-71-252-134-63.dllstx.fios.verizon.net) |
| 16:57.25 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 17:01.24 | p3nguin | Does Linksys SPA-942 do BLF? |
| 17:04.40 | *** join/#asterisk mahaD (mahaD@123.238.235.241) |
| 17:07.34 | p3nguin | Who is a good wholesale provider with a reseller interface? |
| 17:09.14 | talntid | for meetme() it requires zaptel hardware? |
| 17:09.19 | p3nguin | no |
| 17:09.56 | talntid | didn't think it should, but the voip-info page says "The MeetMe application needs a timer to work. There are different ways to get the timer to work, but it won't work by default if you haven't got a Digium Zaptel hardware interface card installed. At this time only zaptel devices may be used. If you do not have a Zaptel device see the ztdummy instructions for timing." |
| 17:10.30 | p3nguin | That's outdated, for one. And for two, you only need the timer that it mentions. |
| 17:10.48 | p3nguin | Just install dahdi and you'll get the dahdi timer. |
| 17:10.56 | talntid | roger that |
| 17:13.06 | talntid | thanks |
| 17:20.27 | *** join/#asterisk timahvo1 (~rogue@197.178.39.194) |
| 17:28.13 | ke-esc | OK- I do have an issue after all. Trying to use multirow func_odbc to get a list of devices which are associated with a particular dialed extension, and concatenate the returned values to produce a string to send to the Dial command. I've based my configs off page 361 of the Definitive Guide... configs and console output at http://pastebin.me/6d3b7cfcd79b35615b41c2b8b30eb9ae |
| 17:32.58 | *** part/#asterisk NotHere (~steve@geozix.static.otenet.gr) |
| 17:45.08 | ke-esc | I forgot to mention, I'm running Asterisk 1.8.9.2 |
| 17:47.47 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
| 17:50.43 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 18:01.47 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 18:04.04 | *** join/#asterisk chasing`Sol (~cS@197.132.189.193) |
| 18:05.04 | tzanger | I'm ahving a moment... what the heck is the "echo" of yoru own voice bouncing off the phone hybrid called? there's a specific term for it |
| 18:05.59 | kaldemar | ke-esc: you set LOCAL(ext) but give ${ext} instead of ${LOCAL(ext)} to the ODBC function. |
| 18:06.59 | kaldemar | ke-esc: check rest of LOCAL() usage aswell. |
| 18:07.55 | ke-esc | awesome- that seemed to do it! The LOCAL() was included as part of a canned stdexten that I started from... I may get rid of it all together, don't think it should be needed |
| 18:08.28 | ke-esc | tzanger, feedback? |
| 18:08.36 | p3nguin | Echo, probably. |
| 18:08.56 | tzanger | no, it's a very fast echo that you actually want |
| 18:09.03 | tzanger | so it doesn't sound like you're talking into dead air |
| 18:09.16 | tzanger | I just can't remember the damned term |
| 18:09.18 | p3nguin | uhhhh.... what? |
| 18:10.17 | tzanger | p3nguin: the hybrid in the phone you're using. it injects some of your own voice energy into the earpiece. that is not the echo you want to get rid of. the echo you want to get rid of is your voice energy bouncing off the far-end hybrid |
| 18:10.28 | ke-esc | "Comfort Noise Generator |
| 18:10.28 | ke-esc | Optionally, the echo canceller can use the Comfort Noise Generator (CNG) to inject spectrally matched comfort noise during nonlinear processing to avoid a dead air effect." |
| 18:10.40 | p3nguin | Comfort noise is something entirely different. |
| 18:12.04 | p3nguin | In not all phones can I hear myself in the ear piece. Polycom HD phones, for example, I haven't been able to hear myself talk. |
| 18:12.31 | Naikrovek | none of them |
| 18:12.33 | Naikrovek | ? |
| 18:12.53 | tzanger | ke-esc: kind of yes |
| 18:13.07 | p3nguin | I've only used like six, and I don't hear myself in the ear piece when I talk into the mic on the handset. |
| 18:13.12 | tzanger | ke-esc: CNG is kidn of the high tech injection of "ambient noise" |
| 18:13.17 | ke-esc | p3nguin, I've got a polycom soundpoint ip 550 hd and i can hear myself? |
| 18:13.29 | Naikrovek | p3nguin: that's.. bizarre. i can hear myself in all polycom hd handsets i have |
| 18:13.43 | p3nguin | ke-esc: Are you asking me if you can hear yourself? 'Cause I sure as heck don't know if you can. |
| 18:14.17 | ke-esc | p3nguin, sorry- didn't mean for a question mark.... statement- i can hear myself. |
| 18:14.32 | p3nguin | Maybe it is just too quiet for me to notice since I am used to hearing that sound much louder in other phones. |
| 18:25.02 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 18:26.02 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:31.23 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 18:37.39 | kaldemar | tzanger: hybrid echo :) |
| 18:39.52 | tzanger | kaldemar: well it's near-end vs far-end |
| 18:39.59 | tzanger | there's a term that I just can't remember |
| 19:11.18 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 19:11.30 | asteriskmonkey | are macros completley broken in 1.6? |
| 19:11.45 | p3nguin | Broken? No. |
| 19:11.48 | asteriskmonkey | :/ |
| 19:12.04 | asteriskmonkey | im trying to do a gototoif statment in a macro and its ignoring the results |
| 19:12.17 | asteriskmonkey | its just processing on to the next line :/ |
| 19:12.42 | [TK]D-Fender | asteriskmonkey, Show us |
| 19:12.46 | p3nguin | ^ |
| 19:13.06 | asteriskmonkey | http://pastebin.ca/2128892 |
| 19:13.19 | [TK]D-Fender | asteriskmonkey, And the failure? |
| 19:13.38 | [TK]D-Fender | asteriskmonkey, } fail in your syntax |
| 19:13.41 | [TK]D-Fender | ] <- |
| 19:13.53 | asteriskmonkey | Gotoif($[${DB(${ARG1}/2)}] = "ON"]?yes:no) ? |
| 19:13.56 | [TK]D-Fender | ${DB(${ARG1}/2)}] = <-------- |
| 19:14.08 | [TK]D-Fender | you added the expression close brace wrong |
| 19:14.17 | asteriskmonkey | ah |
| 19:14.18 | asteriskmonkey | thanks |
| 19:14.33 | asteriskmonkey | let me try that.. been bashing myself for the longest on this one |
| 19:15.07 | p3nguin | I'm surprised there wasn't some verbose output describing that problem. |
| 19:15.13 | asteriskmonkey | no thats not working :/ |
| 19:15.25 | p3nguin | Show us the rest. |
| 19:15.27 | asteriskmonkey | GotoIf("SIP/phil-00001369", ""OFF"] = "ON"?yes:no") |
| 19:15.29 | [TK]D-Fender | asteriskmonkey, that was just the first problem I found immediately. |
| 19:15.31 | asteriskmonkey | Dial("SIP/phil-00001369", "SIP/ |
| 19:15.32 | p3nguin | All of it, not just a piece of it. |
| 19:15.44 | asteriskmonkey | http://pastebin.ca/2128892 |
| 19:15.52 | asteriskmonkey | well there was the original pastbin |
| 19:15.56 | [TK]D-Fender | asteriskmonkey, Next I see quotes on one side, but not the other.... unless your DB has literal quotes in it <- |
| 19:15.57 | asteriskmonkey | i corrected the bracket placing |
| 19:16.08 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 19:16.26 | asteriskmonkey | no, I thought I had to wrap in quotes |
| 19:16.29 | asteriskmonkey | not the case? |
| 19:16.32 | [TK]D-Fender | BOTH sides |
| 19:16.35 | p3nguin | You should, but not ONE side. |
| 19:16.44 | [TK]D-Fender | * does literal content comparisons |
| 19:17.32 | [TK]D-Fender | asteriskmonkey, BTW your barces are a FUBAR'd on the 3 dialplan lines that follow as well.... |
| 19:17.36 | [TK]D-Fender | braces* |
| 19:18.26 | p3nguin | Did I miss a pastebin with more info in it? |
| 19:19.24 | [TK]D-Fender | the same one he first posted |
| 19:19.45 | asteriskmonkey | cleaned it up a bit now getting ouput like OFF] = ON?yes:no |
| 19:19.54 | asteriskmonkey | but its still just going to nextline |
| 19:19.56 | asteriskmonkey | :/ |
| 19:20.13 | asteriskmonkey | Gotoif(${DB(${ARG1}/2)}] = ON?yes:no) |
| 19:20.16 | [TK]D-Fender | asteriskmonkey, New real PB please... |
| 19:20.25 | [TK]D-Fender | And that is still wrong |
| 19:20.32 | [TK]D-Fender | now you have no start to your expression |
| 19:21.05 | p3nguin | Gotoif($["${DB(${ARG1}/2)}" = "ON"]?:no) |
| 19:21.25 | [TK]D-Fender | asteriskmonkey, What is the / 2? |
| 19:21.35 | [TK]D-Fender | trust[-1] |
| 19:21.40 | asteriskmonkey | a key in the astdb |
| 19:21.43 | p3nguin | keys 1 and 2 |
| 19:21.46 | asteriskmonkey | it stores ON or OFF |
| 19:21.49 | [TK]D-Fender | Better be.... |
| 19:22.35 | asteriskmonkey | thanks |
| 19:22.37 | asteriskmonkey | works now |
| 19:22.48 | asteriskmonkey | gah, i fiddle with that think for a good 30mins |
| 19:22.55 | asteriskmonkey | now i feel wicked dumb :/ |
| 19:23.23 | asteriskmonkey | guess its one of those days where to much coffee and no sleep cause brain to stop functioning.. thanks alot guys for pointing that out |
| 19:23.25 | p3nguin | I'd feel dumb for not asking for help sooner, not for making the mistake in the first place. |
| 19:23.27 | [TK]D-Fender | Gotoif($["${DB(${ARG1}/2)}"!="ON"]?no) |
| 19:23.43 | [TK]D-Fender | asteriskmonkey, ^ |
| 19:23.56 | p3nguin | I like that even better! I hate using ?:no in Ifs. |
| 19:24.25 | [TK]D-Fender | p3nguin, Yeah your way was an attempt to remove the pointless jump, but fell to the "all = else" fail |
| 19:24.58 | asteriskmonkey | yeah better to keep that |
| 19:25.02 | p3nguin | I usually correct those when I think to. I actually have like one or two in my own system that I've been meaning to fix, but just haven't done it. |
| 19:25.26 | [TK]D-Fender | p3nguin, Well I'm sure you've trimmed so much else that it really doesn't amtter in the big scheme |
| 19:25.41 | asteriskmonkey | hey can someone answer me this in asterisk 10 has the realtime changed so that you can create contexts in the db and refer to them or do you still need the switch statements and the flat file? |
| 19:26.13 | *** join/#asterisk retentiveboy (~retentive@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 19:26.31 | talntid | [Mar 16 12:26:07] NOTICE[956]: chan_sip.c:11502 sip_reg_timeout: -- Registration for '00065529@70.167.153.130' timed out, trying again (Attempt #16) |
| 19:27.03 | talntid | what are some causes for this? I am registering to two different providers, and both are saying the same thing.. |
| 19:27.22 | talntid | correction, same provider, in 2 different locations (flowroute.. one server in LV, other in LA) |
| 19:27.40 | [TK]D-Fender | talntid, I'd go look at what you're sending them before wondering why they aren't answering back |
| 19:28.08 | talntid | if it matters, this works perfect most of the time, and every once in a while.. they just "drop" |
| 19:28.09 | *** join/#asterisk iulhk (iulhk@119.154.63.148) |
| 19:28.21 | talntid | i'll read the sip debug though |
| 19:30.14 | [TK]D-Fender | talntid, "something else works" and "something similar works" are not valid comparisons as a substitute to looking at the actual problem. Go get some direct debug to look at and see what it says. |
| 19:32.03 | talntid | right. what I mean is... this works, in the same configuration, 90% of the time. just... every once in a while, it comes up as not registered to them. |
| 19:34.27 | talntid | I have 60 agents on the phones right now, placing calls... |
| 19:34.38 | talntid | any way to isolate this without telling them all to get off? |
| 19:37.11 | p3nguin | You are losing registration to your ITSP? |
| 19:37.49 | talntid | yes |
| 19:38.53 | p3nguin | What is your registertimeout set to? |
| 19:39.49 | talntid | http://pastebin.com/sLPnkq2Q |
| 19:40.09 | p3nguin | Wait, does the registertimeout have any effect on a peer that is currently registered, or is that only for retrying? |
| 19:41.28 | talntid | i don't know - but it was registered for about a week |
| 19:42.32 | p3nguin | Here's mine: http://pastebin.com/657mLaLm |
| 19:43.09 | talntid | your register string is in there |
| 19:43.27 | p3nguin | Correct, because it is part of the configuration for the peer. |
| 19:43.28 | talntid | unless you garbled it |
| 19:43.35 | talntid | i meant the secret |
| 19:43.43 | p3nguin | It's not real credentials. |
| 19:43.46 | talntid | roger |
| 19:44.43 | p3nguin | It's basically the same as what you're using, and I *rarely* ever lose registration with them. |
| 19:45.04 | p3nguin | Could be network related, I guess. |
| 19:45.11 | talntid | yeah, I'm wondering if my firewall is mad at them or something... |
| 19:45.35 | talntid | shouldn't be. |
| 19:49.39 | talntid | now it's registered properly |
| 19:49.40 | talntid | :\ |
| 19:51.28 | talntid | the general section: http://pastebin.com/Nq5H5sxY |
| 19:54.24 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:59.20 | p3nguin | allow=all, disallow=all? WTF kind of jacked up configuration is that? |
| 20:10.13 | [TK]D-Fender | talntid, You showed configs... for a PEER. You are having trouble registering. This has nothing to do with your peer. |
| 20:10.26 | [TK]D-Fender | talntid, And you didn't show the actual debug for your registrations attempt |
| 20:13.16 | ke-esc | Is there a good switchboard application for monitoring queues and whatnot that supports a realtime asterisk configuration? |
| 20:17.02 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 20:21.43 | paolosupino | Q: how do I set callerid on outbound SIP channel? |
| 20:26.12 | p3nguin | Set() and CALLERID() |
| 20:26.15 | *** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith) |
| 20:29.04 | paolosupino | p3nguin: I did _00.,1,Set(CALLERID(NUM)=3456787698) and the next step is the Dial command, but the callerid isn't passed. |
| 20:30.09 | *** join/#asterisk BJD10 (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 20:34.21 | jgowdy | Anybody else seen ExternalIVR report H (hangup) when fax is detected? |
| 20:34.22 | vinhdizzo | [Mar 16 13:33:52] WARNING[14934]: file.c:950 ast_streamfile: Unable to open /home/vinh/TNTTSP-Phone/main (format 0x4 (ulaw)): Permission denied |
| 20:34.27 | vinhdizzo | the file main.wav is there |
| 20:34.38 | vinhdizzo | and is properly encoded |
| 20:34.48 | vinhdizzo | because it worked on a different install of asterisk |
| 20:34.57 | vinhdizzo | does anyone know what's wrong now? |
| 20:35.18 | talntid | yeah, permissions are denied |
| 20:35.32 | talntid | ls /home/vinh/TNTTSP-Phone/main |
| 20:35.33 | vinhdizzo | talntid: why is it? |
| 20:35.36 | talntid | show us the output of that |
| 20:36.07 | vinhdizzo | -rwxr-xr-x 1 vinh vinh 902140 Mar 15 2011 /home/vinh/TNTTSP-Phone/main.wav |
| 20:36.21 | vinhdizzo | asterisk process is running as user asterisk |
| 20:36.36 | talntid | so you know what the issue is? |
| 20:36.45 | vinhdizzo | ino |
| 20:36.46 | vinhdizzo | no |
| 20:36.53 | vinhdizzo | i changed the wav files to own by asterisk |
| 20:36.54 | talntid | chown asterisk:asterisk /home/vinh/TNTTSP-Phone/main.wav |
| 20:36.54 | vinhdizzo | still no go |
| 20:37.15 | vinhdizzo | let me retry again |
| 20:37.41 | vinhdizzo | same error |
| 20:37.42 | talntid | you must ensure that the file is in a place, and has proper permissions, to be read by the user asterisk |
| 20:38.00 | vinhdizzo | how so? |
| 20:38.00 | talntid | as a learning experience, type "su - asterisk" |
| 20:38.12 | talntid | and then navigate to where that file is |
| 20:38.29 | [TK]D-Fender | Checkout time, BBIAB |
| 20:40.20 | p3nguin | You can't su to asterisk because asterisk does not have a login shell.` |
| 20:40.33 | p3nguin | At least it isn't supposed to, so if it does, you've make a mistake. |
| 20:40.49 | talntid | root@pbx2:~# su - asterisk |
| 20:40.49 | talntid | $ ls |
| 20:40.49 | talntid | astdb moh sounds sqlite.db |
| 20:41.23 | talntid | plain jane install of asterisk, from repos, on ubuntu 10.04 |
| 20:42.32 | talntid | asterisk:x:104:107:Asterisk PBX daemon,,,:/var/lib/asterisk:/bin/sh |
| 20:42.40 | vinhdizzo | yea i cant su to asterisk |
| 20:42.50 | talntid | interesting |
| 20:43.38 | talntid | well, vinhdizzo, the point is... |
| 20:43.50 | talntid | asterisk doesn't have sufficient rights to read that file |
| 20:43.59 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 20:44.15 | vinhdizzo | talntid: yes, but how do i give it sufficient rights |
| 20:44.18 | vinhdizzo | is it the directory? |
| 20:44.25 | p3nguin | Should be /bin/false |
| 20:44.27 | vinhdizzo | bc the file is readbale by everyone |
| 20:44.50 | talntid | I agree with you p3nguin. I'm going to rip it out. |
| 20:45.01 | talntid | but it's odd that it is like that from the ubuntu repos |
| 20:45.46 | vinhdizzo | talntid: just changed shell to bash for asterisk |
| 20:45.50 | vinhdizzo | now time to test |
| 20:45.56 | talntid | :) |
| 20:46.10 | vinhdizzo | so it's the directory |
| 20:46.15 | talntid | you can always change it back, but this is the best test ;) |
| 20:46.24 | talntid | you'll learn more than if I just tell you the answer |
| 20:46.30 | vinhdizzo | heheh |
| 20:46.32 | vinhdizzo | thanks! |
| 20:46.44 | talntid | now before you just go giving that directory public permissions.. |
| 20:46.49 | talntid | consider moving that file to a more public place |
| 20:46.59 | talntid | like, maybe the asterisk sounds folder |
| 20:47.08 | vinhdizzo | i see |
| 20:47.10 | vinhdizzo | well |
| 20:47.13 | vinhdizzo | i want it in my home directory |
| 20:47.17 | talntid | ok |
| 20:47.20 | vinhdizzo | so i can migrate servers easily |
| 20:47.25 | vinhdizzo | =] |
| 20:47.49 | talntid | well, by exploring the solution this way, you will be able to see the implications of changing the file/directory permissions |
| 20:48.57 | vinhdizzo | back to /bin/false |
| 20:49.05 | talntid | and everything working? |
| 20:49.10 | vinhdizzo | testing now |
| 20:50.32 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:51.30 | vinhdizzo | looks like its working |
| 20:52.09 | talntid | :) |
| 20:52.17 | talntid | and now, you know how to troubleshoot it in the future |
| 20:52.19 | *** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey) |
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| 21:01.53 | vinhdizzo | thanks talntid! |
| 21:02.29 | michael-i | Hi all. I want to have asterisk send a sip notify message when a new cdr has been created. Is the cleanest way to modify the main cdr interface so it calls the SIPNotify AMI command? Alternatives? |
| 21:03.34 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 21:10.17 | [TK]D-Fender | michael-i: what "main cdr interface"? |
| 21:10.37 | michael-i | [TK]D-Fender: main/cdr.c |
| 21:11.43 | [TK]D-Fender | michael-i: copy one of the CDR modules and mod it and load alongside the rest |
| 21:11.49 | michael-i | or maybe create a faux cdr backend: cdr_sipnotify.c |
| 21:11.57 | [TK]D-Fender | MiYes, that |
| 21:12.00 | michael-i | [TK]D-Fender: yup, just clicked :) |
| 21:15.13 | scubes13 | dsanyone here have asterisk and polycoms running over TLS transport? |
| 21:15.23 | scubes13 | s/ds/does/g |
| 21:15.38 | scubes13 | aingt that niftty |
| 21:15.59 | scubes13 | s/aingt/aignt/g |
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| 21:43.07 | michael-i | It seems I can call manager events from modules with manager_event() in manager.c but there is no manager_action()? |
| 21:43.10 | *** join/#asterisk iprouteth0 (~iprouteth@unaffiliated/iprouteth0) |
| 21:43.16 | talntid | uh, something really.. odd just happened |
| 21:43.49 | talntid | pbx2*CLI> sip show peers |
| 21:43.49 | talntid | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 21:44.17 | [TK]D-Fender | chan_sip isn't even loaded |
| 21:44.34 | iprouteth0 | quick question. Is there a way to put the Wait() application into the alternate google voice inbound extension config |
| 21:44.36 | iprouteth0 | exten => username@gmail.com, n, Dial(SIP/101, 180, D(:1)) |
| 21:44.43 | talntid | everyone was making calls... then.. i got a call saying "nobody can dial" |
| 21:44.51 | talntid | checked, and that's how it is |
| 21:44.56 | talntid | i reloaded *, and same thing |
| 21:44.59 | iprouteth0 | somehow put delay into the D(:1)) portion |
| 21:45.16 | iprouteth0 | ? |
| 21:50.01 | talntid | rebooted whole machine, and asterisk came up properly... |
| 21:50.05 | talntid | but, still, wtf. |
| 21:50.45 | kaldemar | iprouteth0: remove the spaces from your extension btw.. "w" is 500 msec pause. |
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| 21:54.30 | iprouteth0 | the spaces matter? It functions with them currently |
| 21:55.43 | iprouteth0 | so it would be something like D(w:1))? |
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| 22:00.44 | kaldemar | iprouteth0: spaces do matter in some places. you better not get used to adding them inside apps. D(w:1) is probably not what you want, look at the syntax in "core show application Dial". |
| 22:02.02 | kaldemar | D(:ww1) would add a 1 s pause before the 1. |
| 22:06.11 | [TK]D-Fender | iprouteth0: No ":" |
| 22:06.21 | [TK]D-Fender | iprouteth0: every "w" char = 500ms |
| 22:06.27 | [TK]D-Fender | iprouteth0: it is not a delimiter |
| 22:06.29 | talntid | [TK]D-Fender, do you feel Asterisk 10 is production ready? or should someone stick to 1.8? |
| 22:06.36 | [TK]D-Fender | 12345wwww67890 |
| 22:06.39 | [TK]D-Fender | 2s wait |
| 22:06.50 | [TK]D-Fender | talntid: Either is fine. |
| 22:07.07 | [TK]D-Fender | not production = pre-release branch. |
| 22:07.24 | talntid | roger that. I'm gonna upgrade from 1.6. |
| 22:10.21 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 22:17.42 | talntid | hm. the move from 1.6 to 1.8 killed my callerids database |
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| 22:43.43 | jgowdy | If anybody is interested in a github based Asterisk SVN mirror |
| 22:43.45 | jgowdy | https://github.com/freedomvoice/asterisk |
| 22:48.30 | scubes13 | asterisk + tls + polycom -> anyone ever setup all three? I have been struggling through google the last 4 days and beating head against wall… any and all assistance greatly appreciated! |
| 22:48.39 | carrar | I'm interested in less gov |
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