00:00.53 | puckett_jw | However, I am NoOp'ing some stuffâ¦. Executing [s@subStandardExt:5] NoOp("IAX2/pbx-3221", "["incoming" != "spinen"] is 1") in new stack Executing [s@subStandardExt:6] NoOp("IAX2/pbx-3221", "{CUT(ARG1,:,2-)} is 10") in new stack |
00:01.13 | puckett_jw | Which shows that they are not null |
00:04.04 | puckett_jw | I am using the new BETA Adium, can someone acknowledge that my posts are coming through? |
00:04.34 | SaiSoma|AtHome | puckett_jw: i'm looking, but no expert here. i'm guessing a syntax error, still looking |
00:04.46 | SaiSoma|AtHome | can you post the NoOp, raw, no the output? |
00:04.48 | puckett_jw | Awesome, thanks |
00:05.20 | puckett_jw | Here are the lines... |
00:05.24 | puckett_jw | exten => s,n,NoOp(Call started in ${CDR(dcontext)} and is going to ${CDR(accountcode)}) |
00:05.24 | puckett_jw | exten => s,n,NoOp(["${CDR(dcontext)}" != "${CDR(accountcode)}"] is $["${CDR(dcontext)}" != "${CDR(accountcode)}"]) |
00:05.24 | puckett_jw | exten => s,n,NoOp({CUT(ARG1,:,2-)} is ${CUT(ARG1,:,2-)}) |
00:05.24 | puckett_jw | exten => s,n,Set(LOCAL(nextSet)=${IF($["${CDR(dcontext)}" != "${CDR(accountcode)}"])?${CUT(ARG1,:,2-)}}) |
00:05.40 | SaiSoma|AtHome | *nod* just FYI . .before fender does it |
00:05.42 | SaiSoma|AtHome | ~pastebin |
00:05.43 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
00:05.44 | SaiSoma|AtHome | :) |
00:07.42 | paulc | pucket_jw: I'm still not clear what your problem is - can we have the bullet point problem definition statement? |
00:07.48 | puckett_jw | BTW - I am running Asterisk 1.6.2.7 |
00:08.40 | puckett_jw | I am trying to set a varable to Null if the dcontext & accountcode are the same |
00:09.01 | puckett_jw | I set the accountcode the the context of the extension elsewhere in the call |
00:09.27 | puckett_jw | I am checking to see if the call is an internal to same context |
00:10.06 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
00:10.13 | puckett_jw | I am trying to use exten => s,n,Set(LOCAL(nextSet)=${IF($["${CDR(dcontext)}" != "${CDR(accountcode)}"])?${CUT(ARG1,:,2-)}}), but I am getting the error⦠Syntax IF(<expr>?[<true>][:<false>]) (expr must be non-null, and either <true> or <false> must be non-null) |
00:11.16 | puckett_jw | I know my expression is not null⦠NoOp("IAX2/pbx-3221", "["incoming" != "spinen"] is 1") |
00:11.45 | puckett_jw | Also the true is not null⦠NoOp("IAX2/pbx-3221", "{CUT(ARG1,:,2-)} is 10") |
00:12.26 | puckett_jw | Therefore, I cannot figure out what I have done wrong. I am sure that I have probably left a character out somewhere, but I cannot find it |
00:13.10 | puckett_jw | I was also woundering if the ":" in the CUT function could be throughing off the IF function? |
00:13.21 | paulc | What's your CUT line all about? Describe its use? |
00:14.36 | puckett_jw | So I am wanting to make the subStandardExt sub get called in a way that it would be recursive, where ${ARG1} could be⦠.<extension>[|dialtimeout][|device][:<extension>[|dialtimeout][|device]:...] |
00:15.31 | puckett_jw | My plan is to have the sub call its self on certain DIALSTATUS' when additional extensions are passed |
00:16.04 | puckett_jw | This way a call could go to a device, and the rollover to another device if not answered |
00:17.52 | puckett_jw | Here is the start of the sub⦠http://pastebin.com/BdG6VsQi |
00:18.25 | *** join/#asterisk ketema (~ketema@2001:470:5:138:217:f2ff:fe05:1e70) |
00:18.59 | puckett_jw | paulc: Does that make sense? |
00:26.41 | paulc | Maybe it's late in the day and my brain is fried, but I still don't get it |
00:27.00 | paulc | mixing stuff passed in (arguments, multiple destinations?) with destinations vs contexts? |
00:28.41 | paulc | backs away slowly - it's almost home time |
00:29.11 | puckett_jw | paulc: OK. Thanks |
00:36.31 | puckett_jw | I found it! I had a ) in the wrong place |
00:37.06 | puckett_jw | It should have been exten => s,n,Set(LOCAL(nextSet)=${IF(["${CDR(dcontext)}" != "${CDR(accountcode)}"]?${CUT(ARG1,:,2-)})}) |
00:38.09 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
00:38.44 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
00:55.13 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
00:56.21 | *** join/#asterisk pkecastillo (~pirruar@190.113.141.122) |
00:56.26 | pkecastillo | hello guys! |
00:57.02 | pkecastillo | I need the "exten number", with AMI commands... any help? |
00:57.09 | pkecastillo | not with the extension inbound, else with the anexo of ring group, while your state is ringing... |
00:58.40 | pkecastillo | I test with "ExtensionStatus", but nothing! :( |
00:58.45 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
01:01.24 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
01:12.04 | *** join/#asterisk sourcode (~code@ppp-61-90-15-9.revip.asianet.co.th) |
01:16.28 | *** join/#asterisk sourcode_ (~code@ppp-58-8-130-79.revip2.asianet.co.th) |
01:31.47 | *** join/#asterisk Kumbang (~kumbang@167.205.24.69) |
01:32.05 | *** join/#asterisk Fairman (~Fairman@c-24-23-61-12.hsd1.ca.comcast.net) |
01:35.51 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
01:40.19 | *** join/#asterisk Kumbang (~unknown@125.163.83.153) |
01:45.32 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
01:45.53 | *** join/#asterisk vutamhoan (~chatzilla@113.22.67.103) |
02:02.45 | *** part/#asterisk ketema (~ketema@2001:470:5:138:217:f2ff:fe05:1e70) |
02:05.18 | *** join/#asterisk tkrn (~tkrn@WS1-DSL-208-102-253-13.fuse.net) |
02:07.58 | *** join/#asterisk sat-man (~jlupresto@74-81-241-158.static.sdyl005.digis.net) |
02:28.34 | *** join/#asterisk kartik (~koolkarti@117.199.113.196) |
02:45.02 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
02:47.28 | *** join/#asterisk jtodd (eh4tauhq8e@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
02:47.28 | *** mode/#asterisk [+o jtodd] by ChanServ |
02:48.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
02:56.27 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
03:09.16 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
03:13.22 | *** join/#asterisk devdvd (~myemail@173-31-160-214.client.mchsi.com) |
03:13.34 | devdvd | hi all |
03:16.36 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-prmcqxkhnnjbzuie) |
03:16.44 | sprite-- | I'm running into a very weird issue. Moving my development environment over to a production environment. Asterisk does not detect hangup from app_conference, versions of everything are the same. No errors or warnings are being thrown. Also tried app_konference which has a bunch of updates and latest asterisk from 1.6 branch. Neither fixed the issue. |
03:19.16 | [TK]D-Fender | sprite--: Wheres the failure for us to examine? |
03:19.58 | sprite-- | [TK]D-Fender: I can generate some logs real quick from working vs non-working. But like I said it's not throwing any errors. After I hang up both parties. They channels are still showing as active if I do sip show channels. |
03:20.10 | sprite-- | Asterisk is never generating a hangup event. |
03:22.26 | pabelanger | sprite--: indications loaded? |
03:22.27 | [TK]D-Fender | sprite--: So far that has precisely nothing to with with app_conference. |
03:23.01 | [TK]D-Fender | sprite--: If * is not told by the remote client "Hey I'm done with this call" then you have ANOTHER problem altogether |
03:23.34 | sprite-- | pabelanger: Nope seems I am missing my indications.conf let me copy that over and see if it fixes the issue. |
03:23.47 | [TK]D-Fender | indications also has nothing to do with SIP. |
03:23.52 | sprite-- | Earlier when I tried with ast 1.6 I had indications.conf though and it didn't solve anything |
03:23.58 | [TK]D-Fender | SIP is afull-progress signalled protocol |
03:24.45 | sprite-- | [TK]D-Fender: It detects hangup when it is not a conference call. It is only when a party is in a conference it does not work. |
03:25.14 | sprite-- | I will get some logs real quick of events working vs non-working server. |
03:25.38 | pabelanger | [TK]D-Fender: :) didn't see the part about it being SIP |
03:25.39 | *** join/#asterisk philippel (~p_lindhei@pool-98-117-128-192.sttlwa.fios.verizon.net) |
03:26.36 | philippel | hey all - does anyone recall the specific 1.4.X version that queues added the additional device state fields? |
03:26.41 | [TK]D-Fender | pabelanger : People always miss the big print |
03:27.08 | *** join/#asterisk Asaph (rob@unaffiliated/robdgreat) |
03:27.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
03:27.45 | [TK]D-Fender | philippel: Thtought that should be 1.6. branches aren't supposed to get new features mid-stream like that |
03:28.18 | philippel | [TK]D-Fender you actually don't want to get me 'started' on that one, but it was back ported into 1.4 somewhere aroudn 1.4.25ish |
03:29.19 | [TK]D-Fender | philippel: Must have taken a lot of arm-bending and favours |
03:29.20 | philippel | and the sad thing was that the 'fix' was not a proper fix and hasn't gotten proper until trunk (1.8) when they finally added HINTS as an option (from a 1.4 patch I submited over a year ago) |
03:29.40 | *** part/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
03:29.48 | [TK]D-Fender | philippel: Hate to say it... hit the changelogs <- |
03:29.50 | philippel | I wasn't part of it happening, just part of trying to get it 'right' with hints |
03:30.09 | philippel | I am - just thought someowne might have known off the top of their head :) |
03:31.18 | philippel | r184980 | mmichelson | 2009-03-30 08:23:59 -0700 (Mon, 30 Mar 2009) | 22 lines |
03:31.25 | philippel | so now to translate that into a versoin number |
03:33.06 | [TK]D-Fender | Should be easy enough. |
03:36.22 | [TK]D-Fender | philippel: 1.4.25-rc1 |
03:36.34 | [TK]D-Fender | 2009-05-13 |
03:36.55 | [TK]D-Fender | philippel: Or the full release following : 2009-05-21 - 1.4.25 |
03:37.08 | philippel | I was just honing in on that :) |
03:37.35 | philippel | thanks |
03:37.42 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
03:39.02 | sprite-- | [TK]D-Fender: Ok the issue was somehow related to trying to start MixMonitor with a recording directory that did not exist. It seems to be working now. Very weird. |
03:41.14 | [TK]D-Fender | sprite--: Problems always disappear when you look at them.. they're like faeries! |
03:42.13 | sprite-- | [TK]D-Fender: I don't see how MixMonitor failing to start should completely break hangup decection in asterisk though. |
03:42.22 | sprite-- | detection |
03:44.00 | sprite-- | but creating the missing directory for my mixmonitor fixed the hangup issue in conference and it now properly detects a hangup. |
03:47.23 | *** join/#asterisk soman (~somnath@118.102.130.6) |
03:50.24 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
03:58.42 | sprite-- | [TK]D-Fender: I lied. It randomly worked for some reason once. Now it's not working again. I will get a log real quick. |
04:04.16 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
04:05.25 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
04:05.40 | *** join/#asterisk jks (jks@193.189.93.254) |
04:06.25 | sprite-- | [TK]D-Fender: http://gist.github.com/404572 This time Asterisk detected one of the hangups. |
04:07.57 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
04:07.59 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
04:09.07 | *** join/#asterisk shmaltz (~chatzilla@mail.dmaven.com) |
04:09.14 | shmaltz | hi everyone |
04:09.25 | shmaltz | ~sleep |
04:10.13 | infobot | sleep is probably overrated, and a poor substitute for caffeine. |
04:10.14 | *** join/#asterisk jksM (jks@193.189.93.254) |
04:10.16 | shmaltz | is infobot asleep? |
04:11.33 | shmaltz | ~google asterisk |
04:14.30 | shmaltz | does anyone know what the market share of asterisk is? |
04:16.05 | *** join/#asterisk p3nguin (gpz5GvdFkf@2001:4978:202:beef:210:4bff:fe2b:9074) |
04:19.52 | vutamhoan | I got this issue - https://issues.asterisk.org/view.php?id=15915 - Any suggestion is appreciated. (Static build fail) |
04:22.05 | sprite-- | [TK]D-Fender: Hit me up when you get back. Updated the gist with a 3rd attempt where needed party is detected. Willing to pay if you can figure out a solution to get this sorted. |
04:23.41 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
04:24.58 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
04:38.36 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
04:42.37 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
04:45.28 | *** join/#asterisk supa_disko (~bleh@secure27.lnk.telstra.net) |
04:49.00 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
04:49.41 | *** join/#asterisk vutamhoan (~chatzilla@113.22.67.103) |
04:52.06 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
05:07.18 | *** join/#asterisk BugKhaM (~BugKhaM@125.25.83.238.adsl.dynamic.totbb.net) |
05:08.24 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
05:08.30 | BugKhaM | Hi, what's controlling the order of cards loaded in /proc/zaptel? Mine keeps changing every time the system reboots |
05:10.06 | BugKhaM | try changing the order of modules loaded in /etc/sysconfig/zaptel or changing priority of script loading in /etc/init.d/zaptel but the problem still persists |
05:14.58 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
05:15.01 | spenguin[work] | TEST |
05:17.47 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
05:17.57 | k-man | anyone ever tried to make a poe injector for a linksys phone? |
05:20.55 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
05:22.27 | p3nguin | I made one for my Cisco desk phone. |
05:23.34 | drmessano | I try to inject poe wherever I go |
05:23.56 | drmessano | QUOTH THE RAVEN, NEVERMORE |
05:24.34 | *** join/#asterisk _gm (~quassel@203.215.176.22) |
05:27.09 | k-man | p3nguin: can you point me to some good instructions for doing it? |
05:27.26 | k-man | can i use the power adaptor that came woth the phone? |
05:27.51 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
05:28.57 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
05:29.25 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
05:29.40 | p3nguin | k-man: http://pinouts.ru/Net/poe_pinout.shtml |
05:30.14 | p3nguin | k-man: I used a regular power brick that I did have plugged into the phone. Now it just plugs in at the jack rather than on the phone. |
05:30.31 | p3nguin | k-man: http://www.interfacebus.com/Power_Over_Ethernet.html |
05:30.41 | k-man | p3nguin: yeah, thats what I want to do |
05:31.51 | p3nguin | k-man: Does your cable come out of a wall jack already? |
05:32.08 | k-man | p3nguin: yes |
05:32.32 | k-man | but i plan to move the ethernet jack to a new location, and add poe at the same time |
05:32.49 | k-man | currently the voip phone is next to the TV which is useless as I'm always on the couch when it rings |
05:32.52 | k-man | i need it next to the couch |
05:32.54 | k-man | hehe |
05:34.18 | p3nguin | k-man: I used a Radio Shack 274-1576 dc power jack, mounted it in the wall plate, disconnected the brown and blue pairs from the back of the rj-45, and wired the power from the dc jack to those free pins on the rj-45. |
05:35.00 | p3nguin | I did it so I only have one cable for the phone instead of two. |
05:36.35 | k-man | yeah, i could do something like that |
05:36.37 | p3nguin | You need to make sure your device is able to be powered by PoE mode B, which powers the unused pairs rather than powering the data pairs. |
05:36.52 | k-man | ah, its a linksys SPA942 |
05:36.57 | k-man | not sure which mode of POE it uses |
05:38.30 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
05:38.39 | k-man | hmm.. i could just buy a PoE switch for about $50 off ebay |
05:38.45 | k-man | that might be a much easier option |
05:39.06 | *** join/#asterisk blaines (~blaines@ip68-106-24-21.ph.ph.cox.net) |
05:41.02 | spenguin[work] | k-man: should be easy |
05:41.07 | spenguin[work] | if you arent following standards |
05:41.08 | k-man | yeah |
05:41.16 | spenguin[work] | only 4 wires are normally used |
05:41.21 | spenguin[work] | for a 100mbit line |
05:41.24 | k-man | hmm... ok |
05:41.29 | k-man | ill look more into it |
05:41.47 | spenguin[work] | leave out 1236 |
05:41.56 | spenguin[work] | the rest are for you to play with |
05:43.43 | k-man | hmm.. i need to buy a roll of networking wire so i can run this new network cable under the house |
05:44.21 | k-man | can you harm a non poe device if you plut it into a powered port? |
05:46.58 | k-man | s/plut/plug |
05:47.38 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
05:48.07 | p3nguin | Not much easier, really. It should take you a couple minutes to drill a hole in the wall plate and mount the power jack, then another couple minutes to solder and wire the power to the ethernet jack. |
05:53.21 | p3nguin | k-man: I guess it depends on if you want to spend $50 or $3. |
05:55.36 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
05:55.58 | p3nguin | I'm seeing info saying the phone supports 802.3af mode A, but nothing saying it doesn't support mode B as well. |
05:58.03 | k-man | whats the difference? any idea? |
05:58.16 | p3nguin | Yes. I thought I already mentioned it. |
05:58.26 | k-man | oh, maybe i missed it |
05:58.31 | p3nguin | Mode A powers over the data pair, mode B powers over the unused pair. |
05:59.13 | k-man | ah, its just the pins |
05:59.15 | p3nguin | For wiring up a dumb injector like I did, I wouldn't advise trying to power over the data pair. |
05:59.24 | k-man | ok |
05:59.26 | k-man | fair enough |
06:00.00 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
06:00.11 | p3nguin | Mode B uses the pairs not needed for data, so it's okay to cut them from the jack to disconnect it from the switch. |
06:00.46 | k-man | ah, isee |
06:01.27 | p3nguin | So what I did was disconnect the blue and brown pairs from the back of the rj-45. |
06:01.48 | p3nguin | That allows the data pairs to still be connected from the switch to the jack. |
06:02.10 | p3nguin | Then mount the power jack in the wall plate and wire the power to the unused pairs. |
06:02.33 | p3nguin | Now power is only on the jack and not the switch. |
06:03.08 | k-man | i was thinking to put the power plug in the cupboard where i have my switch, and inject the power there to the relevant ethernet cable |
06:03.23 | k-man | but same diff, i can do the same thing in the cupboard |
06:03.31 | p3nguin | I don't know if the switch ports can handle power being applied when it isn't supposed to have it, so I just disconnect it entirely. |
06:03.44 | k-man | yeah, i see what you mean |
06:04.32 | p3nguin | I'll have to get some pictures of mine. |
06:04.40 | p3nguin | (tomorrow) |
06:05.31 | Jumpie | p3nguin...was scrolling up on your convo |
06:05.42 | Jumpie | so basically you're having him split away the power pins |
06:05.45 | Jumpie | and just connect data? |
06:06.04 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
06:06.59 | *** join/#asterisk Trixboxer (~praju@datacenter3.supportdepartment.net) |
06:07.03 | p3nguin | Pull the wall plate off and look at the back of the jack. You've got 8 wires, but only 4 are used. Take off the unused pair from the back of the jack and move them out of the way. Now wire the dc power jack to those unused pins on the back of the rj-45. |
06:07.17 | p3nguin | unused pairs, rather. |
06:08.05 | p3nguin | Leave the two data pairs connected, power the unused pins. This makes a "dumb" mode B power injector. |
06:08.38 | Jumpie | but i thought the whole idea was, him wanting to use a poe switch to a non poe device |
06:08.44 | Jumpie | why even bother with the powered wires? |
06:08.53 | Jumpie | just so safety's sake? |
06:09.03 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
06:09.04 | p3nguin | a single Ethernet cable as opposed to an Ethernet cable and a power cable |
06:09.17 | p3nguin | He wanted it for mobility. |
06:09.18 | Jumpie | oooooooh |
06:09.23 | Jumpie | ou're talking ghetto riggin it |
06:09.49 | p3nguin | If that's what you call it, but it's not really rigging at all. |
06:10.08 | p3nguin | It's the same technology that all those $25 PoE devices use. |
06:10.09 | Jumpie | well you are changing the way it was originally intended |
06:10.10 | Jumpie | :D |
06:10.12 | florz | Jumpie: you do have at least a basic understanding of electrical engineering, don't you? |
06:10.23 | Jumpie | im a telcom guy, and yes |
06:10.30 | Jumpie | i just hvent done a seperaton of a poe device/line like that |
06:10.31 | Jumpie | havent had a need |
06:10.41 | p3nguin | If the device supports mode B, then it was intended to be used that way. |
06:10.52 | Jumpie | im just saying, most people use poe devices with poe switches |
06:11.03 | Jumpie | your explanation is great, i was just sayin it was a rig |
06:11.04 | Jumpie | hhe |
06:11.29 | Jumpie | when i meant 'intended' i guess i meant, anything other than using a single cable for power/data |
06:11.43 | p3nguin | You aren't making sense to me. |
06:11.52 | drmessano | He IS using a single cable |
06:12.03 | Jumpie | er well, sorry |
06:12.16 | Jumpie | i guess what i meant was, instead of just plugging the end right into the phone, as a poe device normally |
06:12.16 | drmessano | Just injecting the power at the wall |
06:12.21 | Jumpie | yea... |
06:12.24 | k-man | no, the SPA942 is a poe device |
06:12.27 | Jumpie | thast not what the manuf intended |
06:12.33 | Jumpie | im not sayin its bad lol it was just a comment |
06:12.33 | p3nguin | Sure it is. |
06:12.40 | k-man | i have a non POE switch, so i want to inject power between the switch and the spa942 |
06:12.47 | drmessano | Who cares where the power is injected? |
06:12.50 | *** join/#asterisk Gopal (~Miranda@61.12.17.170) |
06:12.51 | p3nguin | If they built it to use mode B PoE, then that was how the mfg intended for it to be used. |
06:12.56 | Jumpie | k-man i could have sworn you said you had a poe switch going to a non poe device |
06:12.56 | *** join/#asterisk lowlevel (~Stuart@lowlevel.ca) |
06:13.02 | drmessano | At the wall plate, in an inline injector, or at the switch |
06:13.15 | Jumpie | ok, sorry for confusion |
06:13.22 | p3nguin | It's still a midspan PoE injector, so what's the difference? |
06:13.32 | k-man | Jumpie: i was saying that sometimes PoE switches go cheapish on ebay, it might be worth trying to pick one of those up instead of fiddling around |
06:13.35 | drmessano | If he has a poe switch and a non-poe device, wouldn't he just make a sandwich? |
06:14.07 | Jumpie | p3nguin, i guess i meant, most people use poe devices without the intention of having to do any punching, soldering |
06:14.11 | *** join/#asterisk soman (~somnath@118.102.130.6) |
06:14.19 | p3nguin | Why does it have be manufactured rather than fabricated? |
06:14.21 | Jumpie | again, i wasnt bagging on your explanation |
06:15.02 | drmessano | Most people run Windows too. Doesn't make it right |
06:15.15 | p3nguin | It took me 5 minutes and $3 to inject PoE. I think that's pretty frugal. |
06:15.51 | Jumpie | heeh |
06:15.52 | Jumpie | ok ok |
06:16.03 | Jumpie | i think you're gettin overvly defensive about something i wasnt saying was bad at all |
06:16.13 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
06:16.17 | Jumpie | i was simply saying, most people buy poe devices/switches to just plug and play |
06:16.17 | drmessano | I kinda like the idea of cutting the end off of a wall wart, punching down the ends on the unused pins inside the plate, and running to the nearest receptacle |
06:16.18 | p3nguin | You called it a ghetto rig. |
06:16.19 | Jumpie | and youre method was great |
06:16.35 | drmessano | Its actually very clean |
06:16.50 | k-man | yes, its a good plan, i'll have a good look at doing it when i get home |
06:17.42 | drmessano | and you could grab any 5v 1A+ power supply from ebay.. especially one with a zip cord and not the round cable on the linksys supplies |
06:17.49 | drmessano | Make it easy to cut and punch |
06:18.32 | p3nguin | Probably ought to get a 48V one instead of 5V. |
06:19.00 | Jumpie | drmessano that somethin chepa you could get at like, radio shack too? |
06:19.03 | drmessano | Do the Linksys phones use 48V standard PoE? |
06:19.05 | Jumpie | or graybar, grainger |
06:19.06 | BugKhaM | Hi, how to change the order of the zaptel's channel drivers loaded in the kernel |
06:19.07 | p3nguin | I guess it might depend on what you're powering, though. |
06:19.18 | drmessano | Jumpie: If you want to pay 5x as much |
06:19.49 | p3nguin | I have to assume that since they are 802.3af compliant, they'll want 48V over Ethernet. |
06:21.24 | p3nguin | I did learn that my older (legacy) Cisco phone is designed with Cisco's old standards (non-802.3af) and mode B is reverse polarity. |
06:21.40 | *** join/#asterisk d00gster (~dt@87.109.238.2) |
06:22.48 | p3nguin | I'm not brave enough to piggy-back power onto the data pairs, so everything worked out just fine. |
06:23.20 | p3nguin | I potentially could have blown up my phone and switch port. |
06:23.51 | Jumpie | hehe |
06:27.06 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
06:27.57 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-kfmfuxyzvhgqrrqk) |
06:29.09 | *** join/#asterisk BANSAL (~BANSAL@117.199.123.207) |
06:30.24 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
06:35.58 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
06:37.21 | *** join/#asterisk pinoyskull (~pinoyskul@122.55.80.205) |
06:38.41 | maxagaz | can I include a directory in asterisk to manage users.conf and extensions.conf ? |
06:39.07 | p3nguin | #include ? |
06:39.39 | p3nguin | I think you're supposed to include files, not directories. |
06:41.06 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
06:41.18 | maxagaz | p3nguin, directories would be much convenient |
06:41.49 | p3nguin | I don't know if directories are able to be included since I have never done it. |
06:42.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
06:50.20 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
06:51.09 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
06:55.45 | *** join/#asterisk gelo (~gelo@mx01.quobis.com) |
07:02.43 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
07:11.00 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
07:12.03 | *** join/#asterisk jrz (~jrz@a190165.upc-a.chello.nl) |
07:16.04 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
07:19.23 | *** join/#asterisk ksn (~ksn@93-35-11-18.ip52.fastwebnet.it) |
07:22.30 | *** join/#asterisk amunir (~ahmedmuni@203.215.176.22) |
07:24.11 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
07:25.35 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
07:25.43 | *** join/#asterisk p3nguin (~xwQ5kwYl6@mtop-mpls.a2infotech.com) |
07:25.50 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
07:27.34 | *** join/#asterisk Get_The_Fish (~Get_The_F@173-14-4-113-Colorado.hfc.comcastbusiness.net) |
07:28.21 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
07:28.27 | eject_ck | Hi all |
07:29.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
07:29.47 | eject_ck | I have 3 sip clients (SPA3102, Nokia E52, X-lite) behind NAT. Asterisk is in internet :). When I'm trying to register 3rd device - then one of already connected peers goes unreachable. WHat the best practice to deal with this ? |
07:30.21 | Ziaeon | i dont know if its good practice |
07:30.29 | Ziaeon | but you could try changing registration port |
07:30.49 | *** join/#asterisk Yonn (~Yon@212.247.19.244) |
07:30.52 | Ziaeon | I've seen certain ISP's/modem combos dislike various instances of 5060 |
07:31.38 | kaldemar | ~sipnat |
07:31.39 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
07:32.00 | eject_ck | kaldemar: tahnks! |
07:32.53 | kaldemar | so tell asterisk that the clients are behind a nat and turn on qualify. |
07:33.21 | eject_ck | did it already |
07:33.29 | eject_ck | no any additional configs on clients ? |
07:35.26 | kaldemar | not really. |
07:39.01 | *** part/#asterisk amunir (~ahmedmuni@203.215.176.22) |
07:39.18 | *** join/#asterisk amunir (~ahmedmuni@203.215.176.22) |
07:40.03 | ksn | any idea of this error? http://pastebin.org/246635 should be kernel module related, but the modules are loading fine |
07:42.21 | *** join/#asterisk infobot (~infobot@rikers.org) |
07:42.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.7, 1.6.1.19, 1.6.0.27 (2010/05/04), 1.4.31 (2010/05/04), *-Addons 1.6.2.1, 1.6.1.3, 1.6.0.5, 1.4.11 (2010/04/15), dahdi-linux 2.3.0 + dahdi-tools 2.3.0 (2010/04/13), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs |
07:42.36 | Ziaeon | customers are tired of hearing "you need a better network" |
07:48.19 | kaldemar | ksn: which modules? |
07:49.13 | ksn | kaldemar, wctdm, now it loads in asterisk too, but continue to tell me "channel 1,2 ignored" and i don't receive calls :\ |
07:49.36 | ksn | seems i had just to rmmod and modprobe again a couple of time |
07:50.35 | *** join/#asterisk soman (~somnath@118.102.130.6) |
07:51.11 | *** join/#asterisk af_ (~getsmart@88-149-241-148.dynamic.ngi.it) |
07:51.13 | *** join/#asterisk visioconf (~FSC@196.203.15.214) |
07:51.26 | *** join/#asterisk Xt0f (~d54c8bc2@gateway/web/freenode/x-hioushtxffwxjpdo) |
07:51.29 | visioconf | Good morning everybody |
07:51.47 | eject_ck | kaldemar: what do you mean ? |
07:53.31 | kaldemar | eject_ck: that you shouldn't need any additional configuration on the clients. |
07:54.17 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
07:54.57 | kaldemar | ksn: seems that dahdi is not properly configured or 1-2 are not valid channels. |
07:55.32 | eject_ck | ^) |
07:55.44 | eject_ck | kaldemar: I mean NAT specific settings |
07:55.49 | eject_ck | ok, got it working :) |
07:56.46 | ksn | kaldemar, where can i debug more the problem? the config generated seems fine (i just had to change country code) |
07:57.10 | eject_ck | btw, I have SPA-3102 as VoIP gateway to PSTN and vice versa on FXS line. Does it supports to use 2nd Fxs port simultaneously (make to simultaneously calls) as well ? |
07:57.39 | visioconf | any ideas how to be a voip minutes seller ? |
07:58.00 | visioconf | I mean what must I have |
07:58.51 | kaldemar | visioconf: a clear vision of what you want your business to be. |
07:59.45 | Xt0f | Good Morning, Good Morning. I have implemented TLS with AMI but unfortunately I got the msg tcptls.c:218 handle_tcptls_connection: FILE * open failed! when trying to connect to port 5039. I am using CentOS 5.5 and Asterisk 1.6. Any ideas ? |
07:59.51 | kaldemar | ksn: what does your /etc/dahdi/system.conf look like? |
08:00.39 | ksn | http://pastebin.org/246705 <- :p |
08:00.40 | *** join/#asterisk BANSAL (~BANSAL@117.199.127.93) |
08:01.02 | visioconf | yeah, I want to sell voip minutes to other reselling companies. But i'm serching for the tech side |
08:01.30 | kaldemar | ksn: and the output of lsdahdi (or cat /proc/dahdi/*)? |
08:01.39 | visioconf | kaldemar: I need any docs, cases studies , requirements docs...etc |
08:01.45 | BANSAL | can anybody help me .. I am unable to install asterisk ... |
08:01.48 | BANSAL | http://pastebin.org/246706 |
08:02.04 | BANSAL | I tried and got these errors .. |
08:02.19 | BANSAL | can anybody comment on this ? |
08:02.21 | ksn | kaldemar, http://pastebin.org/246708 |
08:02.27 | kaldemar | visioconf: selling voip minutes is quite vague. you must be more specific. |
08:03.50 | *** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-sjacdenhdzfuxifq) |
08:03.59 | kaldemar | ksn: looks like the channels are configured and even in use. how did you configure the channels in /etc/asterisk/chan_dahdi.conf ? |
08:04.37 | visioconf | Kaldemar, OK I'll try to specify it. I need to have an idea about the needed technology and infrastructure to have a voip server and a billing system in order to sale minutes on that server to a specific user |
08:05.36 | ksn | kaldemar, pretty much standard stuff, want a pastebin? |
08:05.41 | kaldemar | visioconf: voip only? no pstn interfacing? |
08:05.51 | kaldemar | ksn: sure, i'll take a look. |
08:06.32 | visioconf | kaldemar, as a first step only VOIP |
08:07.09 | ksn | kaldemar, http://pastebin.org/246718 |
08:07.26 | visioconf | kaldemar: however f there is always pssibility to interface with pstn it would be excellent |
08:09.00 | kaldemar | visioconf: well, you obviously need something to do the VoIP part for you. what is the most suitable, depends on your needs. it might be a B2BUAS like asterisk or a proxy like kamailio or opensips. the billing part is done by some other software. for asterisk, a2billing seems to be used quite widely for an example. many people also write their own interfaces to asterisk's call data records. |
08:10.09 | *** join/#asterisk Z_God (~julius@wlan234091.mobiel.utwente.nl) |
08:10.18 | BANSAL | please help me... |
08:11.03 | visioconf | kaldemar, and how could I specify that such user have such quota via such gateway, is it managed via asterisk ? |
08:11.03 | *** join/#asterisk sulex (~sulex@dynamic-adsl-78-14-171-203.clienti.tiscali.it) |
08:11.46 | kaldemar | ksn: have you tried configuring the channels under [channels]? with "channel => 1-2"? |
08:12.19 | ksn | mh actually not |
08:13.48 | ksn | mh still ignoring |
08:15.23 | kaldemar | check dahdi-channels.conf for some conflicting configuration. |
08:15.28 | *** join/#asterisk Janos (~cramos@190.10.52.113) |
08:15.58 | Janos | got a small question, is ael recommended for use with 1.4 ? or should i upgrade to 1.6 if i want to use it ? |
08:16.28 | *** join/#asterisk jrz (~jrz@c63047.upc-c.chello.nl) |
08:16.40 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
08:16.48 | kaldemar | Janos: you can use it just as well with 1.4. |
08:16.48 | *** join/#asterisk jrz (~jrz@c63047.upc-c.chello.nl) |
08:17.27 | *** join/#asterisk kaarlo (~kaarlo@unaffiliated/kaarlo/x-8749308) |
08:17.28 | kaarlo | hi |
08:17.32 | Janos | kaldemar, sweet, then i think i'm going ael, thanks a lot |
08:17.43 | kaarlo | is there any example how to use lua dialplan on asterisk? |
08:18.26 | kaldemar | kaarlo: configs/extensions.lua in the source package |
08:18.56 | kaarlo | ah, okay |
08:19.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
08:19.16 | kaarlo | thank you, nvm that there is a example lua configuration ;) |
08:19.53 | kaldemar | extensions.lua.sample is the correct filename.. |
08:20.16 | kaarlo | yep, found it at /usr/share/asterisk/conf/extensions.lua.sample |
08:20.42 | kaldemar | BANSAL: have you tried a newer version of libpri? |
08:21.16 | BANSAL | kaldemar: I am using 1.2.2 |
08:22.24 | Xt0f | Hello, any ideas relating to my TLS + AMI problem ? Thanks in advance. |
08:23.04 | kaldemar | BANSAL: any particular reason for not using a newer version? |
08:23.53 | BANSAL | <PROTECTED> |
08:24.26 | BANSAL | <PROTECTED> |
08:24.31 | kaldemar | BANSAL: http://downloads.asterisk.org/pub/telephony/libpri/ |
08:24.32 | Janos | another question, having a static queue(no agents, just sip channels) how does asterisk determines if the queue has someone to answer on it ? |
08:25.15 | Janos | i have this simple queue, and i have to add joinifemtpy=yes otherwise sip channels won't ring |
08:25.42 | ksn | kaldemar, dunno seems fine to me, it can be a kernel/hardware problem? |
08:25.53 | BANSAL | kaldemar: should I use libpri-1.2-current.tar.gz |
08:26.37 | kaldemar | ksn: really looks strange to me, can't see a reason for that, sorry. |
08:26.47 | kaldemar | BANSAL: what version of asterisk are you using? |
08:27.14 | BANSAL | <PROTECTED> |
08:27.16 | ksn | kaldemar, okaay thank anyway :) |
08:27.36 | BANSAL | newer one is supposed to be unstable I think ... |
08:29.12 | kaldemar | BANSAL: use 1.4.10.2 |
08:29.50 | BANSAL | <PROTECTED> |
08:30.09 | drmessano | Libpri 1.2.2 is 4 years old |
08:30.23 | drmessano | You need to be on 1.4.10.2 |
08:31.00 | Xt0f | Nobody for my TLS + AMI issue....? ;-(. ...thanks in advance... |
08:32.13 | kaldemar | BANSAL: i'd suggest that you move to the newest 1.4 release of asterisk and the newest dahdi (zaptel was renamed to dahdi ages ago). if you insist on using zaptel and asterisk 1.4.20, use the latest release which is 1.4.12.1. |
08:33.47 | visioconf | kaldemar, I need any doocuments about haw to setup an IP callshop |
08:33.55 | BANSAL | kaldemar: ok .. so the errors where just because of libpri version ... |
08:33.59 | kaldemar | BANSAL: if this is your first install, use one of the 1.6 branches. 1.6.0 and 1.6.1 branches will soon be switched to securify fixes only. 1.6.2 is the latest release branch. |
08:34.44 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
08:34.58 | kaldemar | visioconf: you probably won't find any before you specify what you need in more detail. |
08:35.53 | visioconf | Kaldemar, I'm newbie in such techs so could you please tell me what do I need to sepecify ? |
08:35.59 | BANSAL | kaldemar: well since I am totally new to asterisk so I am less concern about version ... if you say asterisk 1.4.20 zaptel 1.4.12.1 and libpri 1.4.10.2 will work fine .. then I can proceed with it. |
08:37.23 | *** join/#asterisk MiserySoft (~lee@kalamazoo.dreamhost.com) |
08:38.58 | kaldemar | visioconf: the service you want to provide. what exactly you're looking to sell and to whom and how. |
08:39.05 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
08:40.30 | kaldemar | BANSAL: if you're new to asterisk, don't take old versions. you'll be more likely to run into trouble with them. take the latest releases. |
08:41.02 | BANSAL | kaldemar: ok .. I'll use the latest one ... :) |
08:41.15 | BANSAL | kaldemar: thanks for helping me .. |
08:46.09 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
08:47.00 | Xt0f | kaldemar : could you help me with Open SSL and AMI ? ;-) |
08:49.55 | BANSAL | <PROTECTED> |
08:50.27 | BANSAL | kaldemar: make: *** [makeopts] Error 1 |
08:59.18 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
08:59.57 | *** join/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
09:01.29 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
09:05.27 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
09:06.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
09:09.35 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
09:25.37 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
09:29.00 | carrar | YAWN |
09:30.23 | *** join/#asterisk Tim_Toady (~moi@77.49.63.231.dsl.dyn.forthnet.gr) |
09:41.06 | af_ | I am connnecting a patton 4110 (vesion 5.x firmware) to an asterisk box. do you know of any example? (obly 4.x, that is different) |
09:41.56 | Chainsaw | af_: I have a 4118 connected to Asterisk 1.6.2.6 |
09:42.00 | Chainsaw | af_: Let me know what you need. |
09:42.20 | Chainsaw | af_: Mine only has FXS ports, so I can't help you with FXO matters. |
09:42.31 | af_ | well just a working example. it looks like there isnt' of firmware 5.x |
09:42.39 | af_ | plenty of firmware 4.x |
09:42.52 | Chainsaw | af_: I'm glad I'm not the only one who found that. |
09:42.53 | af_ | syntax differs. no comments on that |
09:43.08 | Chainsaw | af_: And the 400-page software configuration guide is mostly "aren't we great" waffle. Yes. |
09:43.09 | af_ | it looks it's impossible to downgrade, also |
09:43.39 | Chainsaw | af_: Tell you what; send me an e-mail and I'll mail you my 4118 config back. I have 4634 config as well if that's helpful. |
09:43.46 | af_ | and, the reset to factory defaults button does not work |
09:43.48 | Chainsaw | af_: I just need some time to scrub passwords etc. |
09:44.03 | af_ | you are really kind |
09:44.04 | Chainsaw | af_: tony@linx.net |
09:45.31 | af_ | sent |
09:45.45 | af_ | just going to eat something |
09:46.13 | af_ | I will send you back my working conf, if I will have success |
09:46.18 | af_ | see ya |
09:48.35 | *** join/#asterisk Trixboxer (~praju@datacenter3.supportdepartment.net) |
09:54.22 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
10:07.12 | *** join/#asterisk imcdona (imcdona@173.160.189.77) |
10:07.25 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
10:08.09 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
10:11.02 | *** join/#asterisk bonbon (~b49740fa@gateway/web/freenode/x-ljjbwdzzemvzxivz) |
10:12.35 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
10:13.24 | bonbon | hello all, I'd like to know if Asterisk is the right product I should be evaluating for a system that would tie a phone number with a customer. The customer would enter the phone number on our website and would be issued a 6-digit code. He/she will then call a toll-free number from the registered phone and enter the code. An application will then match the incoming phone number and the entered code with an entry in a backend data |
10:13.51 | carrar | sure |
10:13.56 | carrar | you'll need to write that |
10:14.01 | carrar | but isn't very hard |
10:14.24 | bonbon | that's fine. but for call handling, tracking the entered code and the incoming phone number etc - all right up asterisk's alley? |
10:14.32 | carrar | yup |
10:14.37 | bonbon | cool, thanks |
10:15.11 | carrar | & dinner |
10:15.48 | bonbon | issues a six-digit code for dinner |
10:16.20 | *** join/#asterisk joobie (~joobie@CPE-121-220-3-162.lnse1.win.bigpond.net.au) |
10:22.01 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
10:23.39 | *** join/#asterisk Researcher (~user@unaffiliated/unafilliate) |
10:27.28 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
10:29.14 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
10:42.13 | *** join/#asterisk aidinb (~Aidin@71-94-148-218.static.mtpk.ca.charter.com) |
10:44.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
10:46.20 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
10:49.09 | *** join/#asterisk oktay (~oktay@81.215.202.193) |
10:49.35 | oktay | Hello guys. Is there a VOIP phone/router that has OpenVPN builtin? |
10:50.26 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
10:51.23 | florz | oktay: for at least some snoms there is firmware with openvpn available from snom |
10:52.43 | WIMPy | Snom 370 and 8xx AFAIK |
10:53.43 | WIMPy | Some Draytec Vigors also have openvpn support, but I don't know if you can route your voip traffic that way. |
10:55.40 | oktay | i was looking at snom too .. their official forums made it sound like it might not work. |
10:55.56 | oktay | WIMPy: 370 is the minimum that supports openVPN? |
10:56.10 | WIMPy | Yes |
10:56.30 | oktay | this is for Dubai by the way. so I am open to suggestions against blocking. |
11:06.08 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
11:08.20 | oktay | let me see if i can buy snom here.. |
11:09.12 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
11:11.12 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
11:14.16 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
11:16.18 | oktay | a cheapo .. zycoo zp302 seems to support it |
11:17.47 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
11:17.47 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
11:22.21 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
11:22.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
11:22.23 | oktay | anybody know at what level Dubai blocks VOIP ? |
11:23.04 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-jtofmidxnvrhzrzu) |
11:28.39 | eject_ck | oktay: OSI7 I guess ;) |
11:29.21 | oktay | i hope |
11:29.47 | eject_ck | oktay: and even on OSI 7 for RTP |
11:30.16 | oktay | you know how they block Youtube here ? |
11:30.24 | eject_ck | s/OSI 7/OSI 5/g |
11:30.33 | carrar | Just VPN out |
11:30.42 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-zpsnvzqpfpjftosz) |
11:30.44 | carrar | then you can pass whatever you want |
11:30.46 | oktay | yes. i am looking for a phone that supports it |
11:30.49 | eject_ck | oktay: I have no idea :) |
11:30.53 | oktay | eject_ck: DNS |
11:30.56 | carrar | setup a VPN |
11:31.01 | oktay | they point to IP to a page that says it's blocked |
11:31.04 | carrar | so everything on your nework is open |
11:31.12 | eject_ck | LOL :( |
11:31.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
11:31.35 | oktay | if you use another DNS server, you're good to go. |
11:31.42 | carrar | or dump your nazi blocking ISP |
11:32.05 | oktay | carrar: it's not nazi. it's dubai. it think it's a sheik |
11:32.18 | oktay | and that is where the client is. no blocking on my side. |
11:32.37 | oktay | i want to make it simple. so a hardware phone is probably the best choice. |
11:32.48 | carrar | if the client is worth it, setup a site to site vpn |
11:32.51 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
11:33.03 | oktay | carrar: he's my brother :) |
11:33.14 | oktay | we have vpn |
11:33.15 | carrar | defintely more trouble hen it's worth :) |
11:33.37 | eject_ck | Guys I'm playing around features.conf and can't understand why it not works for me :). I'm able to get tt-monkeys via Dynamic Feature (*6) in my case, but can't get any of builtin features like Attended Transfer, One Touch Monitor working :(. What I missed ? I see them on features show but my keypress ignored |
11:33.37 | carrar | If you have a VPN, why not use it? |
11:33.40 | oktay | we have openvpn and I'm looking for a phone that can use it out of the box |
11:34.03 | carrar | pass your SIP through your existing open VPN tunnel |
11:34.07 | carrar | SIP & RTP |
11:34.21 | oktay | what would the phone have to have for that to work? |
11:34.21 | eject_ck | oktay: add youtube hosts to u r local hosts file :) |
11:34.32 | carrar | phone wouldn't need to know |
11:34.33 | oktay | eject_ck: i use opendns. No problem. |
11:34.43 | oktay | carrar: i'm interested |
11:35.01 | eject_ck | oktay: I can recommend symbian (I have Nokia with VPN support) |
11:35.18 | carrar | Phone just things it's connecting to another IP address, just happens to go over your exsiting openvpn tunnel you already have established between your two sites |
11:35.24 | carrar | thinks |
11:35.34 | oktay | ah. need a server on the other end too |
11:35.40 | *** part/#asterisk Xt0f (~d54c8bc2@gateway/web/freenode/x-hioushtxffwxjpdo) |
11:35.46 | oktay | or.. perhaps and extra nic on his box |
11:35.57 | oktay | or just ocnfigure the gateway.. |
11:36.09 | oktay | but he's running windows.. forwarding would suck |
11:36.33 | carrar | build a cheap firewall/openvpn server |
11:36.41 | carrar | and set it to him |
11:36.42 | carrar | send |
11:37.14 | eject_ck | Can someone help me ? |
11:37.15 | oktay | that is not a bad idea. |
11:37.15 | carrar | this way all your lan to lan traffic is encrypted and secure |
11:37.48 | carrar | and do the same for anyone else in your family |
11:37.54 | carrar | linik them all together |
11:38.37 | carrar | You'll find other uses for this secure family VPN network I'm sure |
11:38.39 | oktay | i don't know if i want to have a remote server to look after.. i have to think about that. |
11:38.50 | carrar | then build it on unix |
11:39.03 | oktay | unix? |
11:39.33 | carrar | use small little atom board |
11:39.58 | oktay | hm.. just an extra pci.. |
11:42.27 | carrar | nice little Intel ZOTAC mobos work great |
11:43.15 | carrar | and can be used for other things |
11:44.19 | oktay | yup |
11:53.37 | *** join/#asterisk DennisG (DennisG@84.30.136.208) |
11:56.33 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
12:01.04 | *** join/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
12:01.27 | *** part/#asterisk rttrey (~trey@andc-office-fw.atlantic.net) |
12:01.58 | WIMPy | oktay: If I remember correctely, you can use the Snom 360 as router for the vpn. |
12:02.40 | oktay | 360? |
12:02.58 | WIMPy | Err, 370, sorry. |
12:03.01 | *** join/#asterisk iajrz (~irving@190.80.186.180) |
12:03.01 | oktay | ah. ok. |
12:03.06 | oktay | i am trying to locate one to buy |
12:03.09 | iajrz | hey everyone |
12:04.48 | iajrz | neeone here running asterisk on centos? |
12:05.07 | oktay | blasphemy |
12:05.24 | iajrz | ugh |
12:05.35 | iajrz | am I going to be burnt alive or something :P |
12:05.37 | oktay | what seems to be the problem specific to centos? |
12:06.12 | iajrz | been having issues for about a year, almost once a month. ofr some reason, the filesystem detects orphan inodes or some such |
12:06.17 | iajrz | and asks to do fsck manually. |
12:06.51 | oktay | during a reboot? |
12:06.55 | iajrz | the site this is set up, there's permanent electricity, and I've changed the hardware gradually... |
12:06.55 | WIMPy | iajrz: Doesn't belong here, but are you sure, your hardware is ok? |
12:07.18 | iajrz | after reboot... its working, then stops working, so needs reboot. |
12:07.39 | oktay | what stops working that needs a reboot? asterisk? |
12:07.41 | iajrz | WIMPy: you can never be too sure, but I've gradually changed the whole PC |
12:08.08 | iajrz | oktay: the PC dies. not only asterisk, but everything stops working. |
12:08.11 | oktay | my experience with linux says a reboot does NOT fix things :) |
12:08.20 | iajrz | I infer it might be a centos issue... |
12:08.37 | oktay | it's more probably a hardware issue |
12:08.42 | iajrz | reboot wont fix bug, alright. but when I reboot then fsck -y, it gets back to running |
12:09.06 | iajrz | I feared it might be so... just wanted to check if anyone had a simmilar issue |
12:09.45 | iajrz | so meanwhile, I've been trying to find out how to make the PC run fsck as if it was me when it goes into repair mode so I didn't have to come here once a month. |
12:10.02 | iajrz | just tell em "reboot the bitch!" and get her working |
12:10.19 | oktay | <PROTECTED> |
12:10.46 | WIMPy | Remember the Simpsons episode with the wooden parrot or whatever it was? |
12:11.10 | iajrz | oktay: it dies. Asterisk dies, at the very least. |
12:11.25 | oktay | when you restart it? |
12:11.29 | oktay | asterisk alone |
12:11.31 | iajrz | iirc, cuz lately the people here have been rebooting it by themselves |
12:11.46 | iajrz | by my recollection, everything from prompt to cron dies. |
12:12.02 | iajrz | so hard reset is needed : / |
12:13.32 | iajrz | anyways, just to get something to do, any pointers in what hardware might be faulty? any previous experience? |
12:14.15 | oktay | what have you swapped so far? board? memory? |
12:14.36 | iajrz | I swapped the whole box. |
12:14.44 | iajrz | from case to CPU, you name it, I changed it |
12:14.53 | iajrz | except... oh, god. Except for the T1 card. |
12:15.20 | oktay | you don't see anything int he logs? |
12:15.32 | *** join/#asterisk txwikinger (~quassel@sblug/member/txwikinger) |
12:16.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
12:17.59 | iajrz | uhm... I see a stop. |
12:18.00 | iajrz | literally |
12:18.14 | iajrz | all these entries per second, then silence. |
12:18.35 | oktay | what does 'last' show? |
12:19.00 | iajrz | uhm... anything from a call going in to a call going out or plain old regular stuff |
12:19.08 | oktay | no no |
12:19.11 | oktay | not asterisk |
12:19.13 | oktay | system logs |
12:19.28 | oktay | your system is hanging.. you should not try to debug asterisk |
12:19.46 | *** join/#asterisk jart (~jart@c-76-23-206-246.hsd1.ct.comcast.net) |
12:19.50 | iajrz | I checked asterisk just in case... the rest looks good, too. |
12:20.40 | dmast | g'morning all |
12:20.50 | oktay | g afternoon |
12:20.52 | iajrz | morning |
12:21.48 | oktay | iajrz: asterisk stops working. you look at the screen and you see? what? |
12:21.56 | iajrz | a prompt. |
12:22.07 | oktay | and you can log in? |
12:22.10 | iajrz | a dead prompt. |
12:22.11 | iajrz | nope |
12:22.20 | iajrz | keyboard wont answer either |
12:22.26 | *** join/#asterisk Victor_Yure (~c8a9dedb@gateway/web/freenode/x-arpynkmoktfwqgug) |
12:22.31 | oktay | dead cold |
12:22.32 | oktay | weird |
12:22.39 | oktay | it might still be the T1 card driver |
12:23.15 | iajrz | I had that shivering idea. It's a Rhino R2T2 card |
12:23.36 | iajrz | drivers were downloaded from the web. |
12:23.41 | oktay | fancy name. i've never used a T1 card |
12:24.35 | iajrz | ugh |
12:26.09 | iajrz | well, I'll just have to swap hardware. |
12:26.17 | iajrz | thanks a lot, oktay |
12:26.36 | iajrz | goes home, feeling noobish |
12:26.43 | oktay | hope it works. |
12:27.08 | iajrz | thx :) |
12:27.39 | *** part/#asterisk iajrz (~irving@190.80.186.180) |
12:31.24 | *** join/#asterisk BANSAL (~BANSAL@117.199.127.93) |
12:33.19 | *** join/#asterisk SaiSoma (~saisoma@client105.jdcc.edu) |
12:36.27 | oktay | snom 370 is quite expensive here.. |
12:37.05 | oktay | 250 euros |
12:41.02 | *** join/#asterisk smellynoser (~ashley@87-194-183-38.bethere.co.uk) |
12:41.09 | smellynoser | Has anybody ever used asterisk with a Swyx ISDN card? |
12:41.11 | smellynoser | Is it possible? |
12:41.47 | smellynoser | Also, is there a good SIP client that's recommended for Windows? |
12:41.56 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
12:42.51 | [TK]D-Fender | smellynoser: We don't typically recommend soft-phones period |
12:46.37 | oktay | is there a voip gateway product wiht openVPN (< $200) ? |
12:52.16 | *** join/#asterisk devyll (~paul@thpallady.net.hostway.ro) |
12:53.15 | oktay | which is which again ? FXS, FXO ? :) |
12:53.19 | devyll | hello. logfile: -- Registered SIP '<sipusr>' at <sipclient_ip> port 6202 . does 6202 represent the source port or the destination port ? |
12:53.28 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
12:53.30 | devyll | I want to secure from firewall all sip registration |
13:00.16 | oktay | i think the ports should be well documented |
13:00.18 | oktay | what do you need? |
13:01.13 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
13:01.38 | devyll | I need to restrict all connections to sip registration from all ips except two of them from iptables. and I wanted to make sure that udp:5060 is what I'm looking for |
13:04.10 | oktay | i have 5060 and 5061 |
13:04.24 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
13:04.41 | oktay | http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules |
13:06.22 | *** join/#asterisk centoslinux (~centoslin@1x-193-157-202-125.uio.no) |
13:10.05 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
13:10.32 | oktay | later guys |
13:10.36 | *** part/#asterisk oktay (~oktay@81.215.202.193) |
13:11.27 | [TK]D-Fender | devyll: You could just use the permit/deny masks per-host.... |
13:14.48 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
13:15.04 | *** join/#asterisk dohd (~Xaa@nala.dohd.org) |
13:15.17 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
13:16.05 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
13:17.06 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
13:24.30 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
13:27.51 | *** join/#asterisk StuZZZs (~stuart@82.108.46.35) |
13:29.14 | *** join/#asterisk waKKu (~blah@unaffiliated/wakku) |
13:29.19 | waKKu | hi folks.. a quick doubt |
13:30.00 | waKKu | am I able to "cancel" a blind transfer in SIP ? .. I mean, if I press # to transfer, but want to get back to the call.. is it possible? |
13:30.48 | waKKu | of course, if i no longer type the exten to transfer to |
13:30.52 | WIMPy | I think, you can just press # again. |
13:31.14 | waKKu | hm... seems no work here, using xlite |
13:31.59 | ManxPower | real phones have a transfer button |
13:32.11 | WIMPy | Right. It's the hangup code. *0 for me, but I think I changed that. |
13:32.21 | WIMPy | Indeed. |
13:32.37 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
13:32.44 | waKKu | hm.. lemme try it |
13:32.49 | WIMPy | But it only works on IP phones :-( |
13:32.58 | ManxPower | huh? |
13:33.14 | *** join/#asterisk frk2 (~faraz@zivios/member/fkhan) |
13:33.28 | WIMPy | Have you ever tried a transfer on an ISDN phone? |
13:33.29 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
13:33.40 | WIMPy | Or any other feature except hold for that matter. |
13:33.44 | waKKu | nops.. i don't have one ;( |
13:36.30 | *** part/#asterisk MiserySoft (~lee@kalamazoo.dreamhost.com) |
13:36.33 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
13:37.12 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
13:38.42 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
13:40.12 | *** join/#asterisk [psy] (psy@lounge.datux.nl) |
13:40.44 | [psy] | is it possible to let asterisk call to its self by sip, without using any external servers? i want to do this to create a regression test for asterisk |
13:43.00 | waKKu | you mean, call files? |
13:44.31 | *** join/#asterisk hatoon (~zsggncb@187.111.253.180) |
13:44.40 | [TK]D-Fender | [09:29]<waKKu>am I able to "cancel" a blind transfer in SIP ? .. I mean, if I press # to transfer, but want to get back to the call.. is it possible? <- that is a DTMF Asterisk-based transfer, not a "SIP transfer", and no... blind is blind |
13:44.42 | pabelanger | [psy]: look into the Asterisk testsuite |
13:44.53 | pabelanger | ~testsuite |
13:45.10 | [psy] | pabelanger k i will |
13:45.13 | [TK]D-Fender | [psy]: Sure |
13:45.15 | pabelanger | [psy]: http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/ |
13:45.36 | [psy] | waKKu call files in combination with a good extensions.conf and sip.conf |
13:45.40 | waKKu | [TK]D-Fender, ok.. thanks :) |
13:45.43 | [psy] | but i'll look into the suite |
13:46.08 | *** part/#asterisk hatoon (~zsggncb@187.111.253.180) |
13:46.15 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
13:50.47 | [psy] | pabelanger what i wanna do is make lots of calls and playback a file, then record the file on the other end of the conversation, and after that compare the call quality |
13:51.04 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
13:51.10 | [psy] | you cant do that with sipp or pjsua, both of which we currently already use in our own asterisk test suie |
13:56.02 | pabelanger | [psy]: You can use Originate command |
13:56.07 | *** join/#asterisk ltd (~z@pat.transact.net.au) |
13:56.55 | *** join/#asterisk notjohn (~john@106.165.61.69.DED-DSL.fuse.net) |
13:57.26 | *** join/#asterisk bsaxon (~bsaxon@66.76.242.154) |
13:57.26 | notjohn | where can i find detailed info on reading/querying CDR logs? |
13:57.34 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
13:57.37 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
13:57.52 | [TK]D-Fender | notjohn: In the docs included in your source tarball |
13:58.35 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
13:58.40 | notjohn | [TK]D-Fender: Does it address the actual data or just the field descriptions for the cdr database table? |
13:59.07 | [TK]D-Fender | notjohn: Did you look? |
13:59.47 | notjohn | I've seen README that describes the cdr log table and it's lacking the details of what an "s" ... "h" etc means... how to determine pages/transfers from real calls |
14:00.19 | [TK]D-Fender | notjohn: those are extensions in your dialplan... if you don't know what those are... time to read Chapter 5 of THE BOOK |
14:00.21 | [TK]D-Fender | ~book |
14:00.22 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
14:00.50 | [TK]D-Fender | notjohn: And CDR isn't transactional per-call. Its a "lastapp" type record. |
14:00.59 | [TK]D-Fender | notjohn: Single per-call |
14:01.42 | notjohn | [TK]D-Fender: meaning each action is logged.... not just each call right? |
14:02.34 | [TK]D-Fender | notjohn: "isn't transactional" <- |
14:02.36 | [TK]D-Fender | NOT |
14:02.59 | [TK]D-Fender | [10:01]<notjohn>[TK]D-Fender: meaning each action is logged.... not just each call right? [10:00]<[TK]D-Fender>notjohn: Single per-call |
14:03.08 | [psy] | pabelanger thx, however, that does the same as using callfiles, correct? |
14:03.32 | *** join/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com) |
14:03.34 | kaarlo | hmmm... how did i get the asterisk dialtone when i pick up my soft phone? |
14:03.37 | kaarlo | like the real phone |
14:03.55 | *** part/#asterisk sjb_gt (~sachajber@71-15-84-164.dhcp.gnvl.sc.charter.com) |
14:03.59 | *** join/#asterisk Guest28029 (~GBove@208-104-67-26.dyn.fttp.comporium.net) |
14:04.01 | kaarlo | i want to make it people easier to change to voip |
14:04.04 | pabelanger | [psy]: yes |
14:04.20 | *** join/#asterisk BANSAL (~BANSAL@117.199.120.56) |
14:04.35 | [psy] | so can i create to sip peers and let 1 register to asterisk itself? |
14:05.18 | [TK]D-Fender | kaarlo: When you "pick up" your softphone... that isn't * providing a dialtone. |
14:05.46 | kaarlo | is it only possible over FXO? |
14:05.47 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
14:06.02 | [TK]D-Fender | kaarlo: Your device generates the tone. |
14:06.24 | kaarlo | okay... over FXO asterisk would generate the tone, right? |
14:06.39 | kaarlo | ehm, i mean FXS |
14:06.54 | [TK]D-Fender | kaarlo: FXS devices all generate dialtone. |
14:07.01 | [TK]D-Fender | kaarlo: That is implicit |
14:07.10 | kaarlo | okay |
14:07.13 | [psy] | so can i create two sip peers and let 1 register to asterisk itself? |
14:07.22 | kaarlo | thanks for informations |
14:08.09 | *** join/#asterisk devdvd (~myemail@173-31-160-214.client.mchsi.com) |
14:08.16 | devdvd | morning all |
14:10.20 | [TK]D-Fender | [psy]: You don't need to register for your test. |
14:10.31 | [TK]D-Fender | [psy]: You jsut need to place calls, and yes, you can. |
14:10.41 | [psy] | oki |
14:10.50 | [psy] | i'll fiddle around a bit more |
14:11.18 | *** join/#asterisk jtodd (uqtrdipom6@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
14:11.18 | *** mode/#asterisk [+o jtodd] by ChanServ |
14:11.59 | Faustov | nathalie? |
14:16.30 | devdvd | im trying to dynamically add a member to a queue with addqueuemember that comes in over a sip trunk. so the full line would be soemthing like SIP/trunk/1234567890. When I have AddQueueMember(queueName) in my dialplan and dial in with a locally connected extension (SIP/100) it works fine, adds SIP/100 to the queue and moves on. But when I call in off the sip trunk it only adds SIP/trunk. I have verified Add QueueMember can add the full |
14:16.40 | devdvd | or is it even possible |
14:17.27 | devdvd | using asterisk 1.6.2.7 |
14:17.33 | *** join/#asterisk Raden (~Raden@71.89.121.119) |
14:19.29 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ptzivwbihwuvpqse) |
14:19.34 | notjohn | [TK]D-Fender: I guess i'm confused as to what a "call" includes then |
14:19.54 | [TK]D-Fender | notjohn: "includes"? |
14:20.12 | pabelanger | [psy]: If you are using the testsuite, look at the iax-call-basic test. You can use it as a reference. |
14:20.36 | [TK]D-Fender | devdvd: Show us your actual attempt, and your actual failure |
14:20.44 | [psy] | pabelanger wont that just use the sipp program? |
14:20.50 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
14:21.22 | pabelanger | [psy]: no, the iax-call-basic test uses the originate command from AMI |
14:22.11 | notjohn | [TK]D-Fender: I see data that comes in say for DID call and that's pretty easy to determine.. but if someone comes in off a menu or a transfer from reception it isn't logged as the initial call. There will be the incoming call to the main number and then the transfer will show up as another row |
14:22.12 | pabelanger | [psy]: Actually, the originate command from the CLI |
14:22.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
14:22.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
14:22.41 | [TK]D-Fender | notjohn: a transfer is another call. |
14:22.53 | [TK]D-Fender | notjohn: * dialing another device in the dialplan is another call. |
14:23.04 | [TK]D-Fender | notjohn: Every call gets a CDR (answerwed ones that is) |
14:24.38 | notjohn | [TK]D-Fender: k... I understand that now |
14:27.48 | eject_ck | I still cannot make some bulitin features working :( One touch monitor for example |
14:27.56 | *** join/#asterisk jmkgreen (~chatzilla@fentech.gotadsl.co.uk) |
14:28.07 | devdvd | ok so i think i see the why. I just dont know how to fix it. |
14:28.09 | devdvd | <PROTECTED> |
14:28.09 | devdvd | <PROTECTED> |
14:28.10 | devdvd | <PROTECTED> |
14:28.12 | eject_ck | Can someone help how to make it working ? My features sho One Touch Monitor *1 |
14:28.30 | eject_ck | I make call and then press *1 and nothing happens |
14:28.45 | [TK]D-Fender | eject_ck: pastebin your complete failed attempt |
14:28.48 | jmkgreen | I'm getting 'File ... does not exist in any format' yet the file exists and appears to be the correct format. I'm in need of some inspiration of what to look at next |
14:28.59 | devdvd | looks like asterisk isnt attaching the number to the interface |
14:29.01 | [TK]D-Fender | jmkgreen: Show us the failure and your file. |
14:29.12 | [psy] | pabelanger thx i'll look into that :) |
14:29.13 | pabelanger | jmkgreen: load all format_ modules |
14:29.21 | jmkgreen | [2010-05-18 15:15:15] WARNING[6976] file.c: File voiceglue/tts/Please_confirm_that_this_is_you__Press_1_fJ5iYtBfsO-0E7LNnB56m_0 does not exist in any format |
14:29.24 | jmkgreen | that's the error |
14:29.25 | devdvd | so instead of SIP/callwithus/123456789 im just getting SIP/callwithus-XXXXX |
14:29.25 | [TK]D-Fender | devdvd: No, YOU aren't telling it what device to add |
14:29.29 | *** join/#asterisk Yon (~Yon@212.247.19.244) |
14:29.36 | [TK]D-Fender | jmkgreen: COMPLETE.... |
14:29.55 | [TK]D-Fender | [10:29]<devdvd>so instead of SIP/callwithus/123456789 im just getting SIP/callwithus-XXXXX <- NO |
14:30.22 | jmkgreen | [TK]D-Fender: what bits were you expecting, sorry? |
14:30.29 | devdvd | ok? please educate me master |
14:30.31 | [TK]D-Fender | [10:28]<devdvd> -- Executing [s@agentlogin:1] AddQueueMember("SIP/callwithus-00000000", "support") in new stack <-- this is saying that this channel is the one CALLING the app.. not a PARAMETER passed to it |
14:30.40 | devdvd | ah |
14:31.11 | [TK]D-Fender | [10:28]<devdvd> -- Executing [901@default:1] Goto("SIP/callwithus-00000000", "agentlogin,s,1") in new stack <- guess you aren't noticiing that channel is passed with EVERY app executed. |
14:31.19 | devdvd | yea |
14:31.22 | devdvd | actually i did notice that |
14:31.30 | devdvd | which is where i got my earlier statement from |
14:31.31 | [TK]D-Fender | jmkgreen: COMPLETE failed call and ls dump of the file |
14:31.49 | [TK]D-Fender | devWell again it comes down to "you aren't telling it what device to add" <- |
14:32.16 | *** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca) |
14:32.35 | devdvd | i think the issue is that i expect it to work a way that it doesn't |
14:33.14 | [psy] | ah pabelanger you're of the testteam :) |
14:33.59 | *** join/#asterisk d_preston215 (~chatzilla@static-72-86-159-138.phlapa.east.verizon.net) |
14:34.03 | [TK]D-Fender | devdvd: "core show application AddQueueMember" <- start by reading the instructions. |
14:34.08 | d_preston215 | Morning |
14:34.16 | jmkgreen | [TK]D-Fender: http://pastebin.com/MBA5iKGh I've posted as much as I think is useful - it's bloody huge otherwise |
14:34.17 | devdvd | cuz on a trunk the interface actually is SIP/trunkname whereas with a direct attached extension (softphone, etc) SIP/exten is the interface name |
14:34.37 | *** part/#asterisk bsaxon (~bsaxon@66.76.242.154) |
14:34.42 | [TK]D-Fender | dedvYOU did not pass the device to use as a parameter when YOU called AQM |
14:35.20 | eject_ck | <PROTECTED> |
14:35.41 | devdvd | yea, i know i didn't. thats what im getting at. I was trying to make it dynamic. i know that i can do AddQueueMember(queuename) and that app will pick up the interface automatically |
14:35.50 | [TK]D-Fender | eject_ck: Never said you would see it normally. |
14:35.51 | devdvd | ive seen it done with internal extensions |
14:36.17 | devdvd | but i guess the way asterisk deals with trunks wont allow for that...which is fine, i just needed to know before i started trying to code around it |
14:36.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
14:36.26 | [TK]D-Fender | devdvd: Ther is no "automatic". YOU pass parameters to the app. |
14:36.53 | [TK]D-Fender | devdvd: "Deals with trunks? Pardon? You don't seem to understand the very basics of the dialplan. |
14:37.25 | devdvd | then how do you explain when I do AddQueueMember(support) and then dial in with a locally connected extension that it adds that extension to the queue? |
14:37.29 | [TK]D-Fender | devdvd: Exten => 100,1,AddQueueMember(PUT FUCKING PARAMATER IN BETWEEN THESE BRACKETS TO TELL IT WHAT THE FUCK TO DO!) |
14:37.41 | devdvd | and support is the parameter im passing |
14:37.44 | [TK]D-Fender | devdvd: The the INSTRUCTIOSN to find out what you have to PASS IT |
14:37.50 | pabelanger | [TK]D-Fender: time for a beer |
14:38.01 | [TK]D-Fender | devdvd: That ISN'T SPECIFYING THE device |
14:38.11 | jmkgreen | pabelanger: or a loud hailer |
14:38.55 | eject_ck | [TK]D-Fender: http://pastebin.org/248198 |
14:40.13 | [TK]D-Fender | eject_ck: Go prove that DTMF works from that deive with VoicemailMain or something |
14:40.44 | gelo | eject_ck: you don't have w or W options in your Dial... |
14:40.58 | [TK]D-Fender | eject_ck: And it helps wo read Dial's instructions |
14:41.10 | eject_ck | [TK]D-Fender: sec |
14:41.49 | jmkgreen | So can anyone spot the problem? |
14:42.09 | jmkgreen | I can't see it being a formatting issue as the wav files as been previously working |
14:42.17 | eject_ck | [TK]D-Fender: you are right! thank you a lot! |
14:42.35 | [TK]D-Fender | jmkgreen: pastebin your asterisk.conf |
14:42.43 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
14:43.10 | pabelanger | jmkgreen: And your modules.conf file too |
14:43.11 | ManxPower | Remember everyone, buying a NexTone MSX is something you'll be regretting for the rest of your life. |
14:43.35 | *** join/#asterisk jrz (~jrz@c63047.upc-c.chello.nl) |
14:43.58 | jmkgreen | ok |
14:44.02 | jmkgreen | reload that pastebin entry |
14:45.56 | [TK]D-Fender | jmkgreen: No, it gets a NEW number |
14:46.39 | jmkgreen | [TK]D-Fender: number? What number? |
14:46.43 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
14:46.50 | [TK]D-Fender | jmkgreen: PASTEBIN |
14:46.56 | ManxPower | jmkgreen, the one it gives you when you save a pastebin |
14:46.58 | [TK]D-Fender | jmkgreen: The old one doesn't get "updated" |
14:47.02 | jmkgreen | oh I'm sorry |
14:47.04 | jmkgreen | http://pastebin.com/fW1ttkp3 |
14:47.08 | jmkgreen | didn't spot that bit |
14:47.24 | ManxPower | <PROTECTED> |
14:47.40 | ManxPower | you are dangerously close to being labeled a fool. |
14:48.02 | jmkgreen | <PROTECTED> |
14:48.23 | pabelanger | jmacz: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
14:48.32 | pabelanger | convert your .wav file |
14:48.41 | [TK]D-Fender | jmkgreen: I saw the link... Ok, play another *-provided recording so we can verify the path |
14:49.48 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
14:50.53 | pabelanger | jmkgreen: sox foo-in.wav -r 8000 -c 1 -s -w foo-out.wav resample -ql |
14:51.03 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
14:51.45 | jmkgreen | pabelanger: The tts generator has not been changed - I'm struggling to understand how it might be spitting out the wrong format suddenly... |
14:52.48 | [TK]D-Fender | jmkgreen: Copy an * stock recording in there and try it |
14:53.09 | jmkgreen | that's easier said than done - this asterisk box is configured only to dial out using the tts recordings |
14:53.22 | jmkgreen | it doesn't actually have much inbound extensions :( |
14:53.32 | [TK]D-Fender | jmkgreen: Just shocve a manual playback into your AGI |
14:53.47 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
14:58.46 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
15:00.40 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
15:00.51 | jmkgreen | ok it played vm-INBOX.gsm fine |
15:02.32 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
15:05.27 | [TK]D-Fender | jmkgreen: PASTEBIN |
15:06.51 | Wimme | H |
15:06.55 | Wimme | woops :) |
15:07.18 | jmkgreen | [TK]D-Fender: http://pastebin.org/248339 |
15:08.21 | [TK]D-Fender | [10:52]<[TK]D-Fender>jmkgreen: Copy an * stock recording in there and try it |
15:08.26 | [TK]D-Fender | [10:53]<[TK]D-Fender>jmkgreen: Just shocve a manual playback into your AGI |
15:08.39 | jmkgreen | oh you mean into the voiceglue/tts/ dir? |
15:11.24 | [TK]D-Fender | THERE |
15:11.43 | jmkgreen | http://pastebin.org/248371 |
15:11.52 | jmkgreen | it can't find it within that sub dir |
15:12.20 | [TK]D-Fender | jmkgreen: Verify the fodlers owner as well |
15:12.24 | [TK]D-Fender | folder's |
15:12.30 | jmkgreen | yep it's all asterisk.asterisk |
15:12.33 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
15:12.50 | jmkgreen | drwxr-xr-x 6 asterisk asterisk 4096 2010-05-18 10:22 voiceglue |
15:13.04 | jmkgreen | within that: drwxrwxr-x 2 asterisk asterisk 4096 2010-05-18 16:09 tts |
15:13.16 | [TK]D-Fender | jmkgreen: dot eh call again with debug 10 |
15:15.13 | jmkgreen | http://pastebin.org/248383 |
15:17.04 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
15:19.01 | pabelanger | jmkgreen: Convert your .wav files. PCM Wav is not supported under Asterisk |
15:19.01 | [TK]D-Fender | jmkgreen: go verify that another folder isn't being used like /usr/share |
15:19.07 | [TK]D-Fender | pabelanger: thatr isn't it |
15:19.15 | [TK]D-Fender | pabelanger: he copied a stock recording there totest |
15:21.06 | *** join/#asterisk slima (slima@unaffiliated/slima) |
15:23.08 | pabelanger | jmkgreen: does vm-INBOX.ulaw exist? |
15:23.59 | *** join/#asterisk jtodd (wvnvjije0w@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
15:23.59 | *** mode/#asterisk [+o jtodd] by ChanServ |
15:24.08 | slima | Hi, I have some problem with dialplan, I would like to do this: When callerid isn't 100,101,102... go to: s,4: Gotoif($["${CALLERID(num)}" != "10x"]?s,4) isn't working, whats wrong? |
15:24.34 | slima | x is a problem a gues... |
15:24.34 | paulc | slima: I don't think you can use x as a wildcard character in that scenario |
15:25.14 | paulc | Look at the REGEX function |
15:25.45 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
15:27.51 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
15:28.06 | Corydon76-dig | Also try ${CALLERID(num):0:2} |
15:32.55 | *** join/#asterisk puckett_jw (~puckett_j@70.166.66.172) |
15:33.28 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
15:33.28 | *** join/#asterisk n3hxs (~HAMming@static-151-196-93-200.balt.east.verizon.net) |
15:34.27 | *** join/#asterisk KNERD (~KNERD@200.33.240.253) |
15:35.20 | *** join/#asterisk skymeyer (~skymeyer@91.183.54.9) |
15:36.06 | *** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33) |
15:36.36 | *** join/#asterisk roe_ (~roe___@unaffiliated/roe) |
15:37.04 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
15:37.04 | *** mode/#asterisk [+o file] by ChanServ |
15:37.53 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
15:38.20 | *** join/#asterisk voxter (~voxter@S010600090f53ea17.vc.shawcable.net) |
15:38.50 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
15:39.04 | *** join/#asterisk Warp4 (~Robert@firewall-a.buf.ny.i-evolve.net) |
15:39.17 | Warp4 | hi all |
15:39.32 | *** join/#asterisk notjohn (~john@106.165.61.69.DED-DSL.fuse.net) |
15:41.20 | *** join/#asterisk JT (~j@unaffiliated/jt) |
15:42.19 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
15:42.50 | KNERD | Warp4: Howdy |
15:46.48 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
15:47.40 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
15:47.40 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
15:55.08 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
15:56.04 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
16:01.50 | slima | what's wrong with it: Gotoif($["${CALLERID(num)}" != "^10[0-9]^$"]?s,4) |
16:01.52 | slima | ? |
16:02.08 | pabelanger | slima: not supported |
16:02.28 | slima | i give up |
16:02.29 | pabelanger | slima: look at the REGEX function |
16:02.50 | slima | hm |
16:03.36 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
16:04.35 | *** part/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
16:05.50 | jmkgreen | if you do 'core show settings' and look under Directories - do you see your sounds dir? |
16:05.57 | jmkgreen | cause I don't see it here |
16:06.16 | bent_screwdriver | anyone know a good way to determine what area codes and prefixes are local for a given area? |
16:07.27 | pabelanger | jmkgreen: Because the logic does not exist in your version, it has been added to trunk |
16:07.49 | jmkgreen | [TK]D-Fender: interestingly the file works if I specify it with a full path. |
16:08.26 | jmkgreen | I'm trying to ascertain which path asterisk has for 'astsounddir' |
16:08.55 | Knightfal | jmkgreen: Vanilla install? |
16:09.16 | jmkgreen | Knightfal: close to |
16:09.25 | jmkgreen | it should be /var/lib/asterisk/sounds |
16:09.52 | jmkgreen | files in that dir play, files relative to that dir do not |
16:10.08 | jmkgreen | yet files within that dir, given as a full path, do play |
16:10.31 | jmkgreen | suggests the sounds dir is different, yet the "other" dir has the same files |
16:11.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
16:14.34 | *** join/#asterisk jhirley (~jhirley@adsl-4-164-143.mia.bellsouth.net) |
16:14.43 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
16:14.55 | jhirley | o/ |
16:15.31 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
16:16.02 | *** part/#asterisk gelo (~gelo@mx01.quobis.com) |
16:27.15 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
16:30.07 | eject_ck | how can I make delay before monitor() starts ? |
16:30.38 | eject_ck | There is my dialpan |
16:30.39 | eject_ck | exten => _067XXXXXXX,1,MixMonitor(${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${CALLERID(number)}_${UNIQUEID}.wav) |
16:30.39 | eject_ck | exten => _067XXXXXXX,n,Dial(SIP/121/${EXTEN},30,tTw) |
16:31.29 | eject_ck | I'm route call to SPA-3102 and it starts recording even if called station not answer |
16:31.35 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
16:31.39 | eject_ck | I need to set delay 10 seconds for example |
16:31.43 | eject_ck | is it possible ? |
16:34.00 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
16:34.30 | slima | paulc, pabelanger: I think it's good: GotoIf($["${REGEX("10[0-9]" ${CALLERID(num)})}" != "1"]?4) |
16:35.54 | paulc | slima: There you go! :-) |
16:37.06 | paulc | You could use CUT to examine just the first 2 digits if it's 10x (where X is anything). I thought earlier it was only 101 through 104, in which case REGEX is perfect |
16:37.49 | slima | anyway, thx for help |
16:38.43 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
16:38.55 | Knightfal | eject_ck: If I understand you right you may want to set the b option in mixmonitor |
16:39.41 | eject_ck | Knightfal: not sure :( |
16:40.03 | Knightfal | eject_ck: Try it out :) |
16:40.19 | eject_ck | In my scenario I'm calling to ext and it connects me with SPA-3102, then it calls to actual number |
16:40.20 | eject_ck | ok |
16:43.16 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
16:47.07 | ManxPower | I have a new theory about FreePBX. I believe that the ghost of Rube Goldberg possessed FreePBX developers! |
16:48.08 | *** join/#asterisk TimeRider (~steve@109.224.131.68) |
16:48.11 | jaytee | or the guy who invented the Mousetrap game |
16:49.26 | Corydon76-dig | jaytee: Um, same thing |
16:51.10 | Corydon76-dig | One of the main problems is that they sought to develop an interface around it without realizing that they could have influenced its development. They therefore created all sorts of workarounds for bugs... and when those bugs were fixed, their workarounds broke |
16:51.34 | Corydon76-dig | and then they complained vociferously that the bugfix broke their workaround |
16:53.15 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
16:53.58 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-jgasmfededfxwkrp) |
16:54.39 | *** join/#asterisk ccesario (~ccesario@200.178.219.194) |
16:56.05 | shader | does ConfBridge require a Zaptel timer like MeetMe does? |
16:57.10 | Corydon76-dig | No, it does not |
16:57.26 | Corydon76-dig | btw, they're DAHDI timers now |
16:57.43 | shader | oh? voip-info needs some updating then |
16:57.48 | Corydon76-dig | The owner of the Zaptel trademark would appreciate you not calling it Zaptel anymore |
16:57.57 | shader | ok |
16:58.04 | shader | makes sense |
16:58.43 | shader | do you use either MeetMe or ConfBridge? |
17:00.03 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
17:01.11 | Corydon76-dig | You can use both, if you like |
17:02.54 | shader | does ConfBridge have a means of finding the first available room? or does it work differently? |
17:03.55 | Corydon76-dig | ConfBridge does not have the set of features that MeetMe has. You'll need to build that into the dialplan |
17:04.05 | shader | ok |
17:04.13 | Corydon76-dig | I'd suggest using the groupcount feature |
17:04.39 | shader | ok. where can I find more documentation on it? I haven't seen anything about groupcount |
17:04.56 | Corydon76-dig | The GROUP() function is a core of groupcount |
17:05.07 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
17:05.41 | leifmadsen | waves to Corydon76-dig |
17:07.24 | shader | waves at leifmadsen |
17:07.32 | leifmadsen | zup yo? |
17:07.56 | shader | messin' with conference bridging |
17:09.40 | *** join/#asterisk luke-jr_ (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
17:11.54 | *** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
17:16.52 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
17:19.13 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
17:22.17 | *** join/#asterisk Z_God (~julius@wlan236196.mobiel.utwente.nl) |
17:25.00 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
17:26.47 | shader | how long do you have to wait before playing a sound file, so that the beginning doesn't get cut off? or is that just a side effect of the softphone I'm using? |
17:28.33 | paulc | I'm playing with ConfBridge too.. vs MeetMe.. vs Konference |
17:28.35 | paulc | it's a fun day so far |
17:28.52 | shader | paulc: how's that going for you? |
17:29.00 | paulc | MeetMe does everything I want, I just wonder about 100s of listen only peeps with dahdi_dummi as a timing source |
17:29.00 | shader | like any of them? |
17:29.18 | shader | how do you set up dahdi_dummi? |
17:29.29 | paulc | I like how MeetMe gives me recording and DTMF to exit the conference, as well as "everyone waits till the big cheese arrives" |
17:29.32 | shader | i.e. do you have any documentation on it I could see? |
17:29.40 | shader | ok |
17:29.44 | paulc | ConfBridge gives me the "wait for the big cheese" but no DTMF |
17:29.49 | shader | hmm |
17:30.02 | shader | I bet you could add it |
17:30.07 | paulc | Konference gives me "exit when the big cheese quits" and DTMF in a slightly "different" manner - work-around-able |
17:30.16 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
17:30.17 | paulc | so.. bit of a mish mash of options today :) |
17:30.22 | shader | yeah |
17:30.32 | shader | I'm still stuck on getting the timing device to work for MeetMe |
17:30.45 | shader | it seems that I've been following outdated wiki instructions |
17:30.53 | shader | that nobody's bothered to update |
17:30.55 | shader | :( |
17:31.14 | [TK]D-Fender | "instructions"... cute.. |
17:31.19 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
17:31.27 | [TK]D-Fender | Basic DADHI isntall works just fine. |
17:31.31 | paulc | Yeah - documentation is often the last bit to get consideration.. |
17:31.51 | [TK]D-Fender | And the WIKI is ancient shit you should reference only as a last resort |
17:31.57 | shader | ok then |
17:32.04 | shader | anything more recent I could look at? |
17:32.09 | paulc | [TK]D-Fender: Should I see dahdi_dummy in lsmod if it's loaded? I see a bunch of stuff but no specific "dummy" reference |
17:32.37 | *** part/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
17:32.41 | [TK]D-Fender | paulc: modprobe is run dahdi_cfg -vvv, then test |
17:34.21 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
17:36.37 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
17:37.49 | Knightfal | ~sip |
17:37.50 | infobot | sip is, like, Session Initiation Protocol, http://www.cs.columbia.edu/sip/ (see RFC 3261) It's HIP to be SIP! |
17:38.04 | Knightfal | ~providers |
17:38.05 | infobot | providers is probably http://www.voipreview.org/service.all2.aspx?Country=1&Area_Code=0&CallingArea=0&provider=0&serviceType=1&Adv=1&Features=43 |
17:38.54 | Knightfal | Hrmm whats the command for trusted providers |
17:39.08 | Qwell | none are trusted |
17:39.10 | Qwell | but |
17:39.12 | Qwell | ~itsplist-us |
17:39.13 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
17:39.13 | Knightfal | lol |
17:39.24 | *** join/#asterisk d00gster (~dt@87.109.238.2) |
17:40.52 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
17:41.06 | *** join/#asterisk Janos (~cramos@190.10.52.113) |
17:41.35 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.236.61.dsl.dyn.forthnet.gr) |
17:41.48 | *** join/#asterisk QaDeS (~mklaus@p54A1A0E2.dip0.t-ipconnect.de) |
17:42.19 | Ad-Hoc | hi ppl |
17:44.14 | SaiSoma | hi guys, i am trying to find an asterisk cmd that will count the number of digits in a dialed number. google and other searches are turning up null. I'm obviously not using the proper terms. Any pointers? |
17:44.52 | WIMPy | SaiSoma: len |
17:45.10 | *** join/#asterisk gandhijee (akp@ip67-152-15-148.z15-152-67.customer.algx.net) |
17:45.12 | SaiSoma | WIMPy, rgr. *sigh* so simple:). thank you! |
17:46.19 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
17:46.29 | paulc | [TK]D-Fender: I see "No hardware timing source found in /proc/dahdi, loading dahdi_dummy" but no dahdi_dummy specifically when I lsmod | grep dahdi - so.. I'm good? or it's not actually loaded? |
17:47.35 | leifmadsen | SayNumber(${LEN(${myVariable})}) |
17:47.36 | [TK]D-Fender | paulc: USE IT |
17:49.09 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
17:51.41 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
17:54.29 | ecrane | anyone know why asterisk sip invite has a line in it with 'a=silenceSupp:off - - - -'? What's with the ' - - - -'? |
17:54.50 | KNERD | After all this time I am just starting to use the sound files. Are all different langage file titles suppose to in English? |
17:55.04 | *** join/#asterisk moldy (~rene@unaffiliated/moldy) |
17:55.07 | moldy | hi |
17:55.14 | [TK]D-Fender | KNERD: the STOCK ones, yes |
17:55.25 | KNERD | [TK]D-Fender: okay..thanks for that info. |
17:55.35 | [TK]D-Fender | KNERD: KNERD Because the languague is used as a prefeix to finding them |
17:55.39 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
17:55.49 | leifmadsen | all the sound files names are in english yes |
17:55.53 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
17:56.05 | KNERD | [TK]D-Fender: yes it woul dmake it easier to to stay consistent |
17:56.09 | leifmadsen | different languages are in /var/lib/asterisk/sounds/XX/ where XX is the language code |
17:56.17 | leifmadsen | en, es, and fr (english, spanish, french) |
17:59.18 | shader | paulc: did you get it to work? |
18:01.02 | paulc | shader: gimme a sec, got people at my desk |
18:03.35 | Slugs_ | at or under? |
18:04.59 | *** join/#asterisk QaDeS (~mklaus@p54A1A763.dip0.t-ipconnect.de) |
18:05.07 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
18:06.08 | moldy | when i want to move talks between different phones, do i need support on the phones for that? |
18:10.57 | paulc | LOL if only it was under ;) |
18:11.06 | paulc | moldy: Do you mean transferring calls? |
18:11.16 | *** join/#asterisk sprite-- (~sprite@c-98-251-108-29.hsd1.ga.comcast.net) |
18:12.38 | *** part/#asterisk d00gster (~dt@87.109.238.2) |
18:12.42 | sprite-- | Are AMI connections more reliable in 1.6 vs 1.4? I am building an Adhearsion app and it is losing AMI connection. Jason Goecke advised me that it is an Asterisk issue not an Adhearsion issue. That Asterisk sometimes drops the socket. |
18:13.16 | *** join/#asterisk d00gster (~dt@87.109.238.2) |
18:14.46 | moldy | paulc: yes, sorry |
18:16.45 | *** join/#asterisk mducharme-work1 (~nothing@4-121-188-206.rev.knet.ca) |
18:16.46 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
18:17.32 | KNERD | leifmadsen: leifmadsen: so when calling the different language (other than English) I write it as "/es/tt-allbusy" for example? |
18:17.52 | Qwell | KNERD: no |
18:17.52 | leifmadsen | KNERD: no, you set the LANGUAGE() in the dialplan |
18:18.29 | KNERD | So i have to make more than one dialplan |
18:18.35 | leifmadsen | no |
18:18.56 | leifmadsen | you just have to determine which language you want to use based on either input or whatever other factors you need |
18:19.25 | KNERD | That seems easy enough...thanks |
18:19.29 | leifmadsen | the filenames are the same, so if you specify the language to be used, it will automatically use the language files for the language specified |
18:19.47 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
18:19.56 | paulc | moldy: Hmm.. Transfer is usually a phone function, because it deals with renegotiating the connection between two end points. You can do a transfer within Asterisk (using the * key typically, after specifying a T or t option in your Dial() command) but it's generally better to use the native transfer function of the phone (off a dedicated button etc) |
18:20.49 | anonymouz666 | putnopvut: ping |
18:20.50 | moldy | paulc: hm, ok, thanks |
18:21.08 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
18:21.11 | putnopvut | anonymouz666: pong |
18:21.21 | moldy | paulc: i am currently thinking (again) about which hardware to buy. not sure if the phones i am looking at right now support such functions. |
18:22.04 | anonymouz666 | putnopvut: about that thread "utility of PLC" do you care to share how did you make the tests dropping approx 5% of the RTP streams? |
18:22.56 | putnopvut | anonymouz666: Sure, I patched main/rtp.c so that a random number is used to determine if we should send the packet or not. I can post the patch if you would like. |
18:23.15 | paulc | moldy: What phones are you looking at? Practically all phones support transferring calls. |
18:24.29 | moldy | paulc: gigaset a580 |
18:24.48 | *** join/#asterisk dzup (dzup@unaffiliated/dzup) |
18:26.42 | moldy | paulc: my idea is: buy 6 gigaset a580 phones and 2 gigaset a580 hybrid base stations, connect the base stations to asterisk via voip. |
18:27.28 | putnopvut | oh, he quit... |
18:27.42 | putnopvut | If anonymouz666 comes back, then http://pastebin.org/249285 |
18:28.21 | WIMPy | moldy: I wouldn't recommend the gigaset. I think it's identical to the (DTAG branded) Sinus 501V. |
18:28.28 | Qwell | putnopvut: cheeky |
18:28.33 | d_preston215 | Anyone used a redFONE before? |
18:28.48 | leifmadsen | putnopvut: I sent him a memo with MemoServ with your contents |
18:28.53 | [TK]D-Fender | d_preston215: According to most of their users : avoid. |
18:29.00 | leifmadsen | <PROTECTED> |
18:29.04 | leifmadsen | fyi :) |
18:29.09 | WIMPy | The later has some issues, like not ringing other phones any more if one of them has gone e.g. bue to empty batteries. |
18:29.30 | moldy | WIMPy: hm. is there anything you would recommend? |
18:29.57 | d_preston215 | Yeah....Don't really have a choice right now to avoid it.... |
18:30.16 | WIMPy | moldy: Maybe the Snom M3, otherwise I'd probably go for a ISDN DECT base. |
18:30.21 | Qwell | leifmadsen: oh burn |
18:30.26 | leifmadsen | oh burn indeed :) |
18:30.44 | Qwell | -MemoServ- Memo 2 - Sent by lmadsen, May 18 18:29:44 2010 |
18:30.44 | Qwell | -MemoServ- your face is a nub |
18:30.50 | d_preston215 | My boss swears by them.... |
18:30.50 | Qwell | you's trollin |
18:31.14 | leifmadsen | I don't disagree |
18:32.26 | *** join/#asterisk digiv (~mlhess@141.214.234.28) |
18:32.31 | moldy | WIMPy: the snom m3 is above my budget. i will take a closer look at the isdn/dect models :) |
18:34.44 | moldy | WIMPy: my reason for going for voip bases was to avoid the need for multiple isdn interfaces on the asterisk box |
18:35.55 | moldy | it seems most cheap dect bases can only take 6 phones, but i would need at least 6 cordless ones plus 1 non-cordless one |
18:36.02 | WIMPy | moldy: I had the same idea, but at least I found out that you don't want the cheap ones. |
18:36.04 | [TK]D-Fender | [14:30]<d_preston215>My boss swears by them.... <- most swear AT them |
18:36.15 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
18:36.30 | [TK]D-Fender | d_preston215: So if you don't have a choice... why aer you asking? |
18:36.36 | leifmadsen | I swear at my boss all the time |
18:36.38 | moldy | WIMPy: hehe :) |
18:37.21 | WIMPy | moldy: The Gigaset SIP also only takes six, ond only two simultaneous calls. (can be limited to one, yay) |
18:38.17 | d_preston215 | lol. |
18:38.41 | d_preston215 | I was wondering if anyone knew why this damn thing doesn't send any traffic. |
18:38.54 | d_preston215 | I can use fonulator and run configs and stuff. |
18:39.01 | d_preston215 | But no traffic. |
18:39.09 | moldy | WIMPy: yep, but at least i could connect those to the asterisk box using ethernet |
18:39.14 | WIMPy | I see the A580 IP also has POTS. So it's not identical to the 501V, that's IP only. But I'd fear the software is (at least) similar. |
18:40.13 | [TK]D-Fender | [14:36]<leifmadsen>I swear at my boss all the time <- enjoying being self-employed? ;) |
18:40.38 | leifmadsen | [TK]D-Fender: yep :) |
18:40.57 | sprite-- | Are AMI connections more reliable in 1.6 vs 1.4? I am building an Adhearsion app and it is losing AMI connection. Jason Goecke advised me that it is an Asterisk issue not an Adhearsion issue. That Asterisk sometimes drops the socket. |
18:41.40 | russellb | do you get errors at the asterisk console when it happens? |
18:41.41 | *** part/#asterisk waKKu (~blah@unaffiliated/wakku) |
18:42.00 | *** join/#asterisk datacompboy (~opera@l49-3-84.cn.ru) |
18:42.12 | WIMPy | sprite--: The only thing I can tell you is that I didn't have any trouble so far. Neither on 1.4 nor 1.6, but I used it less on 1.4. |
18:42.15 | sprite-- | I haven't seen any. It seems to happen overnight. So not sure how long the app actually runs before losing a connection. |
18:42.29 | datacompboy | Hi! I have peer with qualify=yes. Is there any function for GotoIf[], that i can use to jump if it UNREACHABLE ? |
18:42.50 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
18:42.50 | [TK]D-Fender | datacompboy: "core show function SIP_PEER" |
18:43.03 | [TK]D-Fender | datacompboy: Might not have the underscore.. I forget |
18:43.09 | sprite-- | I guess best way to find out will be to upgrade to 1.6 and see if I still have the issue. |
18:43.40 | leifmadsen | 1.6 is ambiguous |
18:43.48 | leifmadsen | you mean, "upgrade to 1.6.2" |
18:43.55 | datacompboy | [TK]D-Fender: yea! it without underscode, but that i needed. Lot of thanks! |
18:45.24 | d_preston215 | dahdi_tool gives me nothing but RED alarms for my spans. |
18:46.27 | datacompboy | Any way to test what function returns without change of dialplan and dial? |
18:48.21 | [TK]D-Fender | datacompboy: AFAIK... no |
18:48.31 | [TK]D-Fender | datacompboy: make a little dialplan and originate it... |
18:49.11 | *** join/#asterisk luke-jr (~luke-jr@2002:62b3:1d4c:0:20e:a6ff:fec4:4e5d) |
18:49.39 | datacompboy | [TK]D-Fender: yes... already doing so... thanks for help. now, looks, i can go to sleep... auto switching routes working. |
18:50.50 | *** part/#asterisk datacompboy (~opera@l49-3-84.cn.ru) |
18:56.55 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
18:57.41 | idespinner | anyone know offhand how hunting for a DAHDI group is done? e.g. Dial(DAHDI/g1/number) ? |
18:57.59 | dohd | you mean in what order a free line is looked for? |
18:58.04 | idespinner | yes |
18:58.15 | *** join/#asterisk TimeRider (steve@5ac7b347.bb.sky.com) |
18:58.27 | idespinner | or where in the code to look |
18:58.47 | dohd | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html |
18:59.19 | dohd | I've read it yesterday at voip-info.org, but this page has the same info |
18:59.31 | dohd | or at least it answers your question :-) |
18:59.35 | idespinner | thanks! thats exactly what i needed |
19:06.41 | WIMPy | It's also at the top of extensions.conf by default. |
19:08.17 | idespinner | thx WIMPy i knew i had seen it somewhere but couldnt remember |
19:09.32 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
19:10.12 | SaiSoma | Do * 1.6.x support native call parking on Polycom 331/560? (using feature, not softkeys)? |
19:12.04 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
19:12.51 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
19:14.42 | *** join/#asterisk stix (~stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
19:20.48 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
19:22.50 | TSM2 | i noticed that howler now has a g722 codec for asterisk but its marked as only for 1.4.26.2, any ideas where i can get later versions? |
19:23.24 | ManxPower | ask howler? |
19:23.44 | TSM2 | well i was going to do that just checking if anyone here knew more about it |
19:24.03 | ManxPower | SaiSoma, I strongly doubt it. |
19:24.36 | ManxPower | TSM, people that need G722 use 1.6.x.x |
19:24.48 | SaiSoma | ManxPower, *nod* actually, it does:), just not one button parking. i just figured that much out, thanks though. trying to do a softkey now for one button parking |
19:25.25 | ManxPower | SaiSoma, As I understand it the "server based" stuff in the polycom phones uses an XML protocol that Asterisk does not support. |
19:25.55 | SaiSoma | ManxPower, rgr. thanks |
19:26.55 | TSM2 | ManxPower: its built in on 1.6 then, ile have to see if my disto can migrate to 1.6 easly without breaking FPBX and the other bits |
19:27.16 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
19:31.48 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
19:34.42 | shader | hey guys, I'm trying to create a dialplan where if an internal user dials any extension not in the dialplan, they get sent to an IVR. Unfortunately, when I try useing _X., it matches even for valid extensions |
19:35.07 | *** part/#asterisk Trixboxer (~praju@datacenter3.supportdepartment.net) |
19:35.30 | shader | any ideas on how to do that, that doesn't block extensions like 911? |
19:36.42 | bmoraca_work | asterisk should match based on the most exact pattern available, so "exten=>911" should be matched even when "exten=>_X." exists. additionally, it should also match based on the order they are written in extensions.conf |
19:37.14 | bmoraca_work | if it doesn't, i will of course be flogged summarily |
19:39.35 | shader | any idea why it wouldn't? |
19:39.46 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
19:39.46 | *** join/#asterisk FinboySlick (~shark@74.117.40.10) |
19:39.49 | shader | because that's definitely not happening |
19:40.01 | bmoraca_work | without logs and configs, i could only imagine. |
19:40.09 | shader | hmm |
19:40.36 | shader | is there a pastebin where you can upload files? |
19:40.50 | *** join/#asterisk kotp (~vgoff@96.2.187.66) |
19:40.50 | bmoraca_work | ~pb |
19:40.51 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
19:41.14 | bmoraca_work | just upload the relevant portions: the context in extensions.conf and the console log the failed call |
19:41.21 | shader | ok |
19:47.01 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
19:48.06 | *** join/#asterisk kerx (~kerx@72-57-179-244.pools.spcsdns.net) |
19:48.35 | shader | here it is: http://pastebin.com/BYb0Wkhb |
19:49.21 | shader | this is just a rought draft/demo dialplan |
19:49.25 | shader | *rough |
19:49.33 | shader | but it should still be able to dial 911 |
19:51.55 | WIMPy | What context is the phone you used in? |
19:52.05 | WIMPy | phones? |
19:52.08 | shader | yes |
19:52.36 | WIMPy | Then exchange the two includes. |
19:53.27 | shader | why should that help? |
19:54.46 | WIMPy | AFAIK order matters on includes, which would explain your observations. |
19:55.04 | shader | so, if I exchange the two includes, I can reach 911, but not the extensions in internal |
19:55.08 | *** part/#asterisk FinboySlick (~shark@74.117.40.10) |
19:55.08 | shader | hmm |
19:55.11 | bmoraca_work | you include "internal" in your context before "outgoing" which means it processes _X. before it even sees 911 |
19:55.25 | shader | ok |
19:55.32 | bmoraca_work | within a context, it will look for the most exact match, but you have multiple contexts here |
19:55.38 | *** join/#asterisk jmacz (~jmacz@200.85.225.62) |
19:55.39 | shader | ah |
19:55.45 | shader | so they aren't really "included"? |
19:56.00 | WIMPy | no |
19:56.08 | shader | hmm |
19:56.09 | shader | now what |
19:56.15 | shader | any better ideas? |
19:56.29 | kerx | hi all, i wanted to confirm my bandwidth and pps calculations are done correctly for g729 audio codec |
19:56.34 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
19:56.39 | WIMPy | You could create a new context for the catchall and include it last. |
19:56.39 | kerx | i have 30 calls running w/ g729 audio codec. |
19:56.45 | leifmadsen | include order does matter |
19:57.08 | leifmadsen | so make sure the more significant matches are listed prior to the pattern matches |
19:57.13 | shader | WIMPy: good idea |
19:57.46 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
19:57.55 | kerx | would this come out to 3,000 packets per second on 30 g729 calls in asterisk? |
19:58.01 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
20:02.12 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
20:02.55 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
20:04.44 | shader | what are the units of TIMEOUT(digit) and TIMEOUT(response)? |
20:04.49 | leifmadsen | seconds |
20:04.53 | shader | odd |
20:04.59 | leifmadsen | you can use something like 3.4 though |
20:05.06 | leifmadsen | why odd? |
20:05.12 | shader | does Set(TIMEOUT(response)=10) not do what I think it does? |
20:05.19 | shader | because it times out in 3 seconds |
20:05.20 | leifmadsen | I don't know what you think it does |
20:05.27 | [TK]D-Fender | kerx: No. |
20:05.30 | shader | time out in ten seconds if you don't do anything |
20:05.46 | leifmadsen | do anything where? |
20:05.53 | shader | i.e. push a button on your phone |
20:05.58 | leifmadsen | when? |
20:06.03 | [TK]D-Fender | SOON! |
20:06.04 | leifmadsen | in an auto-attendant? |
20:06.04 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
20:06.11 | shader | an IVR, yes |
20:06.15 | [TK]D-Fender | shader: SHOW US |
20:06.17 | leifmadsen | I think you mean auto-attendant |
20:06.22 | shader | though I might be doing it wrong |
20:06.29 | [TK]D-Fender | shader: Likely |
20:06.31 | shader | [TK]D-Fender: of course |
20:06.31 | leifmadsen | but I'd like to see the dialplan and the console output |
20:06.35 | leifmadsen | IVR != auto-attendant |
20:06.35 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
20:06.44 | shader | ok, which is which? |
20:07.24 | leifmadsen | http://en.wikipedia.org/wiki/Interactive_voice_response |
20:08.08 | shader | ok |
20:08.34 | shader | so IVRs involve reacting to the user's voice? |
20:09.08 | [TK]D-Fender | shader: PASTEBIN |
20:09.15 | shader | I'm getting to it |
20:09.19 | leifmadsen | that's not what it says |
20:09.37 | leifmadsen | IVR is something that interacts with an external resource |
20:10.11 | leifmadsen | accepts data from a user and returns data back to the user. A good example of this is something like when you call a pizza place and it says, "If you would like the exact same order as last time, press 1" |
20:10.26 | leifmadsen | you press 1, then it says, "your order will arrive in 40 minutes, thanks for calling Pizza Pizza!" |
20:10.43 | bmoraca_work | copyright infringement! |
20:10.52 | bmoraca_work | "Pizza Pizza" is trademarked by Little Ceasers! |
20:11.13 | leifmadsen | Pizza Pizza is the name of a pizza company in at least the Toronto area :) |
20:11.22 | bmoraca_work | interesting |
20:11.22 | *** join/#asterisk neurosys (~neurosys@69.199.183.150) |
20:11.36 | leifmadsen | http://pizzapizza.ca |
20:11.50 | bmoraca_work | you crazy canadians and your lack-of-rule-following ways |
20:13.01 | leifmadsen | we spit in the face of US patent and copyright laws! |
20:13.08 | leifmadsen | hides from the Feds |
20:13.15 | bmoraca_work | good, someone should. |
20:17.35 | Qwell | leifmadsen: Little Caesars was dumb. Their trademark is on "Pizza! Pizza!" |
20:17.44 | leifmadsen | haha nice |
20:18.04 | shader | lol |
20:18.12 | shader | so Pizza Pizza! is ok? |
20:18.18 | [TK]D-Fender | Qwell: The catch-phrase so nice they have to say it.... AGAIN.... |
20:18.25 | Qwell | leifmadsen: http://en.wikipedia.org/wiki/Little_Caesars#Trademark_in_Canada :) |
20:19.37 | leifmadsen | someone slipped up :) |
20:19.39 | shader | [TK]D-Fender: http://pastebin.com/bqteF6QZ |
20:20.00 | Corydon76-dig | leifmadsen: How long has the restaurant been around? |
20:20.06 | leifmadsen | years |
20:20.17 | bmoraca_work | lol |
20:20.19 | leifmadsen | it's not a restaurant though -- it's a pizza delivery service |
20:20.25 | Qwell | leifmadsen: actually, looks like Pizza Pizza came before Little Caesars starting using the catchphrase. |
20:20.28 | [TK]D-Fender | shader: exten => s,n,WaitExten(5) <-takes precedence |
20:20.37 | Corydon76-dig | I'd say Little Caesars has lost the trademark in canada, then |
20:20.38 | leifmadsen | weak sauce |
20:20.41 | leifmadsen | aye |
20:20.42 | [TK]D-Fender | shader: Here you explicitly overrode the channel timeouts |
20:20.43 | Qwell | so if they were to expand into the US, they probably *can* use it |
20:21.07 | shader | ah |
20:21.19 | shader | so it shortened to 5 from ten? |
20:21.31 | shader | btw, does it automatically wait for the end of the audio file anyway? |
20:21.54 | [TK]D-Fender | shader: 5s from end of audio |
20:22.12 | shader | so the timeout starts counting after the audio finishes |
20:22.16 | [TK]D-Fender | shader: Should |
20:22.20 | shader | good |
20:22.58 | shader | I tried fixing the syntax error mentioned in the log, but I couldn't |
20:23.36 | shader | it seemed like it was jumping into the invalid extension macro, even though no invalid extensions were entered |
20:23.49 | shader | am I supposed to terminate macros in a special fashion? |
20:24.00 | shader | or is something simpler going on? |
20:24.10 | [TK]D-Fender | shader: exten => s,n,GoToIf($[ISNULL(${NUMTIMEOUTS})]?set:main) <- wrong braces to reference a FUNCTION |
20:24.28 | leifmadsen | yep, missing ${ } |
20:24.44 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
20:24.45 | cusco | hello |
20:25.02 | leifmadsen | GotoIf($[${ISNULL(${NUMTIMEOUTS})}]?set:main) |
20:25.04 | cusco | can I have two different asterisks with the SAME queues? |
20:25.07 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
20:25.16 | cusco | we have two oficces, both with PRI |
20:25.17 | shader | so it's supposed to be ${ISNULL(${NUMTIMEOUTS})}? |
20:25.24 | [TK]D-Fender | shader: -- Timeout on SIP/201-0000006d, going to 't' <-- its went to "t" just like it should |
20:25.39 | [TK]D-Fender | shader: As one way, yes |
20:25.44 | cusco | so if WAN goes down, office1 cnnot come and queue in asterisk in office2 |
20:25.44 | leifmadsen | cusco: if you mean have two different boxes but expect the queues on those boxes to communicate the delivery order, then no, it will not work like that. |
20:26.20 | shader | [TK]D-Fender: there's another way? |
20:26.28 | kerx | [TK]D-Fender, what should 30 simultaneous calls in queue w/ g729 codec generate for packets per sec? |
20:26.29 | cusco | leifmadsen: so how is it done? |
20:26.36 | leifmadsen | cusco: how is what done? |
20:26.43 | kerx | I thought it was ~1,500 pps based on my calculation with http://www.bandcalc.com/ |
20:26.45 | [TK]D-Fender | shader: Yes. Go read up on Asterisk Expressions. |
20:26.54 | kerx | I select the top radio box that says Payload is |
20:26.56 | cusco | I mean.. BIG LARGE call centers have loads of asterisk boxes |
20:27.06 | cusco | do they use the same asterisk for queuing? |
20:27.12 | leifmadsen | cusco: very carefully -- you have different queues on each asterisk box |
20:27.14 | kerx | cusco, what do you consider BIG LARGE? how many agents on the floor? |
20:27.29 | cusco | right now we expect about 300 operators.. |
20:27.31 | leifmadsen | cusco: you cannot share caller position across multiple asterisk boxes |
20:27.37 | cusco | 100 in office1 and 200 in office2 |
20:27.41 | leifmadsen | cusco: they would be separate queues |
20:27.42 | kerx | ok that's definitely big large :) |
20:27.51 | cusco | nor queue weight |
20:27.58 | kerx | anything greater than 50 is considered big large in my opinion |
20:28.11 | kerx | [TK]D-Fender, you there? Still trying to figure out packets per sec :) |
20:28.14 | leifmadsen | cusco: ok, lets make this even more general: you cannot share queue information across physical boxes |
20:28.27 | leifmadsen | cusco: they would be independent queues with their own non-shared information |
20:28.34 | shader | [TK]D-Fender: given that the wiki is a horrible reference, any other recommended place to look it up? |
20:28.46 | leifmadsen | shader: http://www.asteriskbook.org |
20:28.48 | leifmadsen | errr... |
20:28.52 | leifmadsen | shader: http://www.asteriskdocs.org |
20:28.52 | cusco | leifmadsen: ok ... |
20:29.00 | cusco | thanks for clearig my doubts |
20:29.57 | [TK]D-Fender | checkout time, BBIAB |
20:30.36 | kerx | anyone can help me figure this packet per second question |
20:30.41 | kerx | i'd really apprecite it |
20:31.46 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
20:33.45 | Qwell | kerframil: what is the question? |
20:33.47 | Qwell | kerx: |
20:34.24 | kerx | Qwell, I've got 30 simultaneous calls in queue w/ g729 codec |
20:34.36 | kerx | I want to know how many packets per second those 30 calls together are generating |
20:34.47 | Qwell | 20ms per packet |
20:34.49 | kerx | I'm using iptraf and I'm seeing something around 1,900 packets per sec |
20:35.08 | Qwell | (1000 * 30) / 20 |
20:35.12 | WIMPy | That depends on how much data you put into one packet. Codec doesn't matter. |
20:35.45 | kerx | how do i figure that out ? |
20:35.51 | kerx | in my sip.conf i dont have that specification |
20:35.53 | Qwell | <Qwell> 20ms per packet |
20:36.15 | WIMPy | (Unless you can't put as much data into a packet as you want due to network restrictions) |
20:36.35 | kerx | well i'm having major packet loss right now |
20:36.45 | kerx | because the carrier in my building i just moved to, doesnt have a good router |
20:36.47 | Qwell | changing the size of the packets isn't going to help that.. |
20:36.49 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
20:37.09 | Qwell | in fact, increasing the packetization will only make the problem more audible |
20:37.18 | kerx | for informational purposes, how do i change packet size in asterisk configuration? |
20:37.51 | WIMPy | Samller packets -> mor stress for the router, bigger packets -> more impact on the audio. |
20:37.52 | lanning | WIMPy: RTP packets are small, you don't stuff as much as you can. it's about keeping a steady flow. |
20:38.21 | bmoraca_work | i believe that asterisk uses the linux network stack, which means that your MTU is set there, and not within asterisk |
20:38.52 | Qwell | bmoraca_work: your MTU isn't going to be < 50 bytes... |
20:38.55 | WIMPy | Like lanning said, you will under no normal circumstances hit the MTU. |
20:39.15 | bmoraca_work | Qwell: i was just going to mention that it likely doesn't matter because your packets are much smaller |
20:44.06 | devmod | Is there any way to dynamically add/remove sip peers ? |
20:44.53 | pabelanger | devmod: Asterisk realtime |
20:46.41 | *** join/#asterisk italorossi (~italoross@201.76.154.130.intranet.digi.com.br) |
20:46.50 | devmod | Is this still true "There is no support for NAT keep-alives" |
20:50.35 | pabelanger | devmod: you can decrease your SIP registrations to 30 seconds |
20:54.40 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
20:55.12 | devmod | pabelanger, I guess I could do that, the problem is that i cannot always control the endpoints |
20:57.19 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
21:01.08 | *** part/#asterisk jplank (~GBove@208-104-67-26.dyn.fttp.comporium.net) |
21:02.18 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
21:04.22 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
21:08.23 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
21:13.47 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
21:13.52 | devmod | is there any way to filter the events I receive through AMI? like receive only a specific type of event, etc? |
21:16.37 | *** join/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com) |
21:17.35 | settntrenz | With the Cisco 7940 IP phone, missed calls from external #'s are correctly displayed as 10 digit (north american) numbers. The only problem with that is a "9" is required to dial that number back so trying to place the call from the missed call menu doesn't work. Any ideas for a workaround. I don't think adding a 9 in front of the CLID is the best way to solve. |
21:18.20 | p3nguin | Fix your stupid dialplan so you don't have to dial a 9 before the real number. |
21:18.30 | p3nguin | It's dumb and useless. |
21:18.31 | WIMPy | What else would you do, if not make the 9 unneccessary? |
21:19.35 | settntrenz | p3nguin: wish I could. Personally I agree, unfortunately the people who sign the checks don't. |
21:20.32 | p3nguin | If some people desire to dial a 9, leave that part of the dialplan to keep them happy, but fix it so it isn't required anymore. |
21:21.27 | p3nguin | Unless you have internal extension numbers of 7, 10, or 11 digits, it doesn't make any sense to require a special code to distinguish between internal and external phone numbers. |
21:22.48 | settntrenz | p3nguin: thanks for the suggestion. |
21:23.10 | [TK]D-Fender | settntrenz: change the callerid before calling your phones to add the 9 in front |
21:29.17 | shader | does anyone have a list of standard dialing patterns to cover international calls? |
21:29.22 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
21:29.37 | Qwell | . |
21:29.39 | Qwell | shader: ^^ |
21:30.13 | shader | yes, Qwell? |
21:30.23 | Qwell | . will cover international |
21:30.28 | shader | lol |
21:30.42 | jdoe | p3nguin: never got why people did that. |
21:31.03 | Qwell | shader: Qwellzakistan uses 1 digit phone numbers. |
21:31.15 | shader | awesome |
21:31.24 | shader | I bet you have less than 10 people living there |
21:31.29 | shader | do you have any spare numbers? |
21:31.30 | Qwell | There is no international regulatory body. |
21:31.32 | shader | how much do they cost? |
21:31.35 | Qwell | yes, 9 of them. |
21:31.39 | *** join/#asterisk zatriz (~asdf@static-98-117-149-122.sttlwa.fios.verizon.net) |
21:31.43 | Qwell | $1,000,000/min |
21:31.51 | shader | hmm |
21:32.02 | shader | I think I'll buy my DIDs from a less exotic location |
21:32.09 | *** join/#asterisk AdamTwelve (~Adam@adsl-75-15-191-192.dsl.bkfd14.sbcglobal.net) |
21:32.17 | shader | thanks for the offer though |
21:32.21 | Qwell | But you could have a prime DID like...7. |
21:32.28 | shader | very prime |
21:32.39 | shader | but with a price to match |
21:33.37 | shader | bill gates himself could only afford #7 for about a month |
21:33.42 | zatriz | Wondering if anyone has had any issues with a Polycom ip6000 connected to asterisk 1.4 not able to dial to pstn. Its able to recieve pstn calls and also able to call local peers |
21:34.02 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
21:34.31 | shader | zatriz: but the other peers can dial pstn? |
21:34.47 | zatriz | yes all the other peers are able to dial pstn no problem |
21:35.00 | shader | are they in the same dialing context? |
21:35.13 | zatriz | but the other peers are also cisco 7961 and yes all in the same context |
21:36.27 | shader | do you have a log of a call attempt? |
21:36.36 | [TK]D-Fender | zatriz: IP 6000 has nothing to do with PSTN <- |
21:36.38 | devmod | is there any way to filter the events I receive through AMI? like receive only a specific type of event, etc? |
21:36.43 | zatriz | yeah let me find some place to upload to |
21:36.44 | [TK]D-Fender | zatriz: it is a SIP phone. |
21:36.58 | [TK]D-Fender | ~pb |
21:36.59 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
21:37.05 | [TK]D-Fender | zatriz: ^^^^^^^^ |
21:37.47 | shader | too bad lisppaste doesn't support arbitrary irc channels |
21:37.55 | shader | it could have been so much more useful |
21:40.11 | zatriz | shader:http://pastebin.com/3AAU58ws |
21:40.20 | zatriz | its got pri debuging enabled as well |
21:40.59 | [TK]D-Fender | <PROTECTED> |
21:41.04 | [TK]D-Fender | zatriz: Calling an invalid # |
21:41.31 | *** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-32-49-180.mia.bellsouth.net) |
21:41.33 | zatriz | I have a capture of a cisco calling the exact same number and its going through |
21:41.41 | WIMPy | .. or incomplete |
21:41.45 | [TK]D-Fender | zatriz: Where? |
21:43.04 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
21:43.51 | [TK]D-Fender | < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event |
21:43.54 | [TK]D-Fender | Occurs twice |
21:43.58 | zatriz | http://pastebin.com/nFZBPRi7 with cisco calling out |
21:44.31 | [TK]D-Fender | Cisco = -- Executing [5642383@stations:1] Dial("SIP/desk-5878-00001117", "Zap/G1/5642383") in new stack |
21:44.40 | [TK]D-Fender | Polycom = -- Executing [8774936@stations:1] Dial("SIP/5839-0000111b", "Zap/G1/8774936") in new stack |
21:44.45 | [TK]D-Fender | NOT the same number |
21:45.21 | zatriz | wrong paste but with the polycom i can dial any number and it will do the same thing 911 even |
21:45.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
21:45.43 | [TK]D-Fender | zatriz: So far I see one number fail and a story that doesn't match |
21:46.22 | *** join/#asterisk timmyd (~timmyd@pool-173-79-13-149.washdc.fios.verizon.net) |
21:46.30 | zatriz | TK: What doesn't match |
21:46.50 | [TK]D-Fender | zatriz: the PHONE NUMBER |
21:47.50 | timmyd | are there any major security issues for asterisk 1.4.21.2? i'm getting scanned (e.g. [May 14 07:35:52] NOTICE[28107]: chan_sip.c:15236 handle_request_register: Registration from '"999"<sip:999@173.79.13.149>' failed for '74.115.162.15' - No matching peer found), then a little later [May 14 07:36:35] NOTICE[28107]: chan_sip.c:14035 handle_request_invite: Call from '103' to extension '0020121002828' rejected because extension not |
21:48.04 | Qwell | timmyd: asterisk.org/security |
21:48.13 | zatriz | would it make it any easy if i uploaded more logs that should it failing with other numbers because its failing |
21:48.29 | [TK]D-Fender | TimeRider: Do something stupid like use a purely numeric password or a dictionary word? |
21:48.32 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
21:48.47 | timmyd | no, using generated random passwords |
21:49.07 | shader | timmyd: you might want to restrict the ports on your firewall to only accept connections from your service provider, unless you're accepting connections from random other asterisk computers over the internet |
21:49.27 | [TK]D-Fender | zatriz: You tell me the PHONE is at fault and that the same number works on your Cisco. And you fail to show me that case |
21:50.03 | timmyd | shader: i thought contactpermit=10.120.0.0/255.255.0.0 would have fixed this issue but the line must be ignored? |
21:50.06 | [TK]D-Fender | timmyd: Mixed alpha-numeric? |
21:50.24 | timmyd | [TK]D-Fender: yes |
21:50.34 | [TK]D-Fender | timmyd: that should be "permit", not "copntactpermit" |
21:51.13 | [TK]D-Fender | timmyd: And yes, there are plenty of security issues. Why do you think we're at 1.4.31? |
21:51.38 | timmyd | well ubuntu hasn't backported it so i guess i'll need to find an updated repostiory |
21:51.54 | p3nguin | Anyone know if the Cisco 7912G with SIP can do distinctive ring? |
21:53.03 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
21:53.19 | p3nguin | It doesn't respond to Alert-Info in a SIP header like the 7960/7940 does. |
21:53.21 | shader | how does the directory command make use of its context? where does it get the numbers it uses to call the mailbox owners? |
21:53.35 | zatriz | TK: i never said its the phone, but it could be the phone, or some asterisk configuration |
21:53.43 | [TK]D-Fender | timmyd: Funny... Mine says 1.6.2.5 |
21:54.09 | [TK]D-Fender | zatriz: You said "same number works from Cisco". Are you telling me different now? |
21:54.26 | timmyd | [TK]D-Fender: you're probably not on jaunty? |
21:54.30 | [TK]D-Fender | zatriz: Because right now all we have is your telco telling you "that number isn't valid" |
21:54.51 | zatriz | TK: http://pastebin.com/q2Lnc3KK |
21:54.56 | [TK]D-Fender | timmyd: Welcome to Antiquity... population : YOU |
21:55.23 | zatriz | Another number that fails from the polycom but successfully completes from the cisco |
21:55.47 | *** join/#asterisk jhirley_ (~jhirley@adsl-4-163-179.mia.bellsouth.net) |
21:55.56 | zatriz | I've have not been able to make even one successful call out to the pstn from the polycom from the 30 cisco phones i've had 1000's |
21:56.10 | *** join/#asterisk jtodd (anji00v6mn@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
21:56.11 | *** mode/#asterisk [+o jtodd] by ChanServ |
21:56.52 | [TK]D-Fender | zatriz: # -- Executing [8774936@stations:2] Dial("SIP/5839-0000111b", "Zap/G1/8018774936") in new stack |
21:57.14 | [TK]D-Fender | > Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '7178774936' ] |
21:57.21 | [TK]D-Fender | <PROTECTED> |
21:57.26 | [TK]D-Fender | Bull |
21:57.30 | [TK]D-Fender | ^^^^ |
21:58.21 | zatriz | because i didn't do a search replace all on the area code ? |
21:58.50 | [TK]D-Fender | zatriz: Don't waste our time fucking with the evidence |
21:59.54 | *** join/#asterisk pabelanger (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
22:00.17 | timmyd | [TK]D-Fender: oh well, time to upgrade to 9.10. thanks all |
22:00.21 | *** part/#asterisk timmyd (~timmyd@pool-173-79-13-149.washdc.fios.verizon.net) |
22:00.23 | hardwire | at this point I'm wondering if verizon made up their own phrases for all of their rate sheets. |
22:00.28 | zatriz | TK: if you can't help me then dont just because you dont understand how this could be happening. I've install enough asterisk systems to understand that this is whats happening |
22:01.17 | [TK]D-Fender | zatriz: You tell us "this is the situation". Then you preceed to NOT show us that this is the case. Then you forge a new BS story. How the hell is anyone supposed to figure out whats going on? |
22:01.41 | [TK]D-Fender | zatriz: So far you are altering the numbers, and your telco says "that number.... we don't like it"' |
22:01.46 | [TK]D-Fender | zatriz: There is nothing here we can trust |
22:01.51 | p3nguin | And tamper with the logs and expect you to solve the problem. |
22:01.54 | *** part/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com) |
22:01.58 | [TK]D-Fender | zatriz: And you've provided nothing else to go on |
22:02.36 | [TK]D-Fender | "Hi, here's a picture of me in a clown suit....... NOW TELL ME WHAT'S WRONG WITH MY DAMN CAR!" |
22:02.47 | [TK]D-Fender | And in other related news... |
22:02.55 | p3nguin | after he photoshopped his face on the picture. |
22:03.08 | [TK]D-Fender | p3nguin: No... Someone ELSES's face :) |
22:03.22 | *** join/#asterisk shader (~40846872@gateway/web/freenode/x-qkpoecyvbyycadbu) |
22:03.38 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
22:03.47 | zatriz | Damn i didn't think you could be that thick |
22:04.14 | TimeRider | fender been drinking too much peosi again? |
22:04.38 | p3nguin | I could use anice cold Pepsi right now. |
22:05.26 | [TK]D-Fender | zatriz: What have you given us? nothing consitent, no configs, no versions, we don't know even what telc you're using. The fact you seem to pass of 7 AND 10 digit numbers to them in the first place. |
22:05.35 | [TK]D-Fender | zatriz: What have YOU done to help us help you? |
22:07.09 | zatriz | Asterisk 1.4.27.1 Libpri 1.4.10.2 Zaptel 1.4.12.1 |
22:07.22 | zatriz | which configs do you want ? |
22:07.37 | zatriz | sip.conf extension.conf ? |
22:07.54 | [TK]D-Fender | zatriz: zatriz sip shouldn't matter. |
22:08.01 | [TK]D-Fender | zatriz: zatriz ZAPTEL <----------- |
22:08.07 | [TK]D-Fender | zatriz: Your PRI is rejecting you. |
22:08.15 | zatriz | Yes its not running DADHI |
22:08.24 | [TK]D-Fender | zatriz: Now * is placing that call.. and the signalling is there, and it is the other side telling you to GTFO. |
22:08.50 | [TK]D-Fender | zatriz: zapata.conf please. |
22:10.14 | zatriz | http://pastebin.com/hRGgNMzK |
22:10.53 | [TK]D-Fender | zatriz: That is not what I asked for. |
22:11.35 | zatriz | sorry hang on |
22:12.40 | zatriz | http://pastebin.com/PUnT1iRE |
22:12.47 | *** join/#asterisk Cresl1n (~matt@asterisk/libpri-and-libss7-expert/Cresl1n) |
22:12.47 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
22:13.26 | Qwell | zatriz: use a different site... surely you see the giant warning at the top of the page? |
22:13.47 | [TK]D-Fender | zatriz: pridialplan=national <- change to "unknown" |
22:14.58 | ManxPower | [TK]D-Fender, odd how following the documentation (that says you almost never need to set pridialplan) is totally ignored by so many people. |
22:15.16 | ManxPower | like zatriz, for example. |
22:17.03 | devmod | any recommendations on a reliable asterisk manager proxy ? |
22:18.06 | Qwell | asterisk |
22:18.11 | Qwell | no need for a proxy |
22:19.09 | devmod | Qwell, even if i will have multiple clients connecting to it? |
22:19.16 | Qwell | devmod: sure |
22:19.49 | p3nguin | Hahaha... like Asterisk is only capable of a single user agent connecting to it at a time, or something. |
22:21.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
22:21.31 | [TK]D-Fender | p3nguin: My sweater is multi-threaded, but only fits one user! HALP!!!!!!!!!!!! |
22:21.43 | p3nguin | lol |
22:21.55 | Qwell | [TK]D-Fender: little known fact - sweaters are single-threaded. it's just a really long thread. |
22:22.37 | [TK]D-Fender | Qwell: LIES |
22:22.50 | [TK]D-Fender | Qwell: I *know* the arms were separate! |
22:23.51 | *** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no) |
22:24.05 | *** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp) |
22:27.46 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
22:33.17 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
22:34.19 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
22:34.23 | ManxPower | Is devmod still asking silly questions? |
22:34.25 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
22:34.43 | devmod | ManxPower, yup |
22:39.37 | hardwire | what would you guess "switchless 1+ service" would mean? |
22:41.52 | hardwire | does it simply mean interstate trunk since you don't need to originate on the PSTN? |
22:42.23 | *** join/#asterisk blaines (~blaines@67.130.168.2) |
22:44.51 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
22:45.30 | drmessano | Did someone say something about arms? |
22:45.51 | drmessano | Because, you know, you can't hug with nuclear arms. |
22:46.24 | [TK]D-Fender | drmessano: You can... but it tends to make the other parties nervous :) |
22:46.33 | drmessano | lol |
22:47.47 | *** join/#asterisk gospch_ (~gospch@unaffiliated/gospch) |
22:47.51 | drmessano | I need a pet shock collar that works with X10.. |
22:48.04 | drmessano | Need to be able to shock my cat via Asterisk |
22:48.06 | ManxPower | I need one that work ON XO |
22:48.10 | p3nguin | Click a key on the computer, kill a dog. |
22:48.24 | [TK]D-Fender | DON'T TASE ME BRO! |
22:48.28 | drmessano | HAHAH |
22:48.40 | ManxPower | I'd love to ZZZZTTT! sales reps when they report a problem with a call but don't include any other information |
22:48.49 | drmessano | I just spit my drink.. brb |
22:49.31 | drmessano | Well timed "Don't tase me bro"..lol |
22:50.42 | ManxPower | Or verizon! Zap! What do you Zap! mean the customer Zap! was not there? Zap! They were Zap! there all day! Zap! |
22:51.51 | [TK]D-Fender | ManxPower: REMINDS ME OF THE OPENING SCENE TO gHOSTBUSTERS |
22:51.55 | [TK]D-Fender | darn caps |
22:55.24 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
22:56.24 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
22:59.05 | *** join/#asterisk gospch (~gospch@unaffiliated/gospch) |
23:03.01 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
23:06.15 | *** join/#asterisk jks (jks@193.189.93.254) |
23:07.54 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
23:08.07 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
23:15.55 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
23:21.22 | *** join/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
23:21.34 | *** part/#asterisk oldhack (~jfincher@cpe-24-27-56-221.austin.res.rr.com) |
23:21.39 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
23:33.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
23:34.47 | *** join/#asterisk boodu (~boodu@175.158.129.128) |
23:37.00 | *** join/#asterisk ketema (~ketema@2001:470:5:138:217:f2ff:fe05:1e70) |
23:37.28 | *** part/#asterisk ketema (~ketema@2001:470:5:138:217:f2ff:fe05:1e70) |
23:40.20 | *** join/#asterisk riddlebox (~james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:45.43 | *** join/#asterisk TSM2 (~the_softw@87-194-32-212.bethere.co.uk) |
23:46.22 | riddlebox | hello, if I have a sip endpoints as something like Tom, Brian,etc... then in my extensions.conf I set global variables as 110 => SIP/Tom, is there an easy way to say if someone dials a _1XX dial the global variable? or am I going about it wrong? |
23:49.47 | ChannelZ | http://dlisted.com/node/37318 |
23:49.53 | WIMPy | riddlebox: Interesting Idea. It's possible, yes. |
23:52.30 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
23:52.54 | riddlebox | WIMPy: I am sure its possible but now my dialplan is exten => _5XX,1,Dial(SIP/${EXTEN},20) |
23:53.17 | riddlebox | oops thats supposed to be a 1XX |
23:53.54 | bmoraca_work | you need dynamic variable names |
23:54.41 | WIMPy | ${${EXTEN}} |
23:54.45 | bmoraca_work | yep |
23:55.09 | [TK]D-Fender | Last I heard variables had to start with a letter |
23:55.15 | bmoraca_work | it does |
23:55.28 | bmoraca_work | ${PEER_${EXTEN}} |