00:10.34 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
00:12.07 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
00:15.44 | *** join/#asterisk phl4kx (i=phl4kx@190.42.218.119) |
00:15.49 | phl4kx | hi all |
00:16.14 | phl4kx | I have a AEX 800P with 1 FXO port but the card never anserw the call |
00:16.26 | phl4kx | who can help me please |
00:17.37 | *** join/#asterisk moua (i=moua@82.66.50.159) |
00:17.39 | moua | hi |
00:21.58 | *** join/#asterisk Maxous (n=Maxous@76.97.26.56) |
00:22.30 | *** part/#asterisk Maxous (n=Maxous@76.97.26.56) |
00:22.40 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
00:24.46 | *** join/#asterisk Levonk (n=lk@adsl-76-230-108-150.dsl.lsan03.sbcglobal.net) |
00:31.41 | *** join/#asterisk ujwal (n=ujwal@124.41.255.249) |
00:42.09 | *** join/#asterisk seanmh (n=johndoe@c-76-113-36-170.hsd1.nm.comcast.net) |
00:59.55 | *** join/#asterisk shag- (n=jsb@neptune.uqtr.ca) |
01:05.37 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
01:06.28 | TJNII | /dev/md1 2.1T 115G 2.0T 6% /mnt/mediaHD |
01:06.35 | TJNII | giggles gleefully |
01:07.32 | *** join/#asterisk jeev (n=email@unaffiliated/jeev) |
01:09.12 | moua | anyone ? |
01:11.04 | moua | i have a sip account, i wish to get a busy signal on my DID when someone call it, and it should trigger a webpage (i want to activate a php script by call) |
01:11.13 | moua | someone know how to do this ? |
14:57.25 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
14:57.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
14:57.48 | ManxPower | about time you got here, jbot! |
14:57.58 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
14:58.35 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
14:59.09 | *** join/#asterisk spokra (n=spokra@host093-179-144.sea0.speakeasy.net) |
14:59.26 | MrTelephone | manx |
14:59.29 | MrTelephone | help me with the digest :P |
14:59.41 | MrTelephone | have you ever seen clients calculate the digest retardedly? |
15:00.06 | jaytee | hahaha, jbot overslept! |
15:00.22 | MrTelephone | jaytard |
15:01.04 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
15:01.53 | *** join/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
15:02.05 | *** part/#asterisk codestr0m (n=codestr0@unaffiliated/codestr0m) |
15:03.48 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:10.06 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
15:10.06 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc4 (2008/09/03), 1.4.22-rc3 (2008/09/03), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 (2008/09/03), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
15:10.14 | MrTelephone | it's crazay |
15:10.29 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
15:10.37 | MrTelephone | I had to edit chan_sip.c to log notice asterisks calculation so I can compare it to the response |
15:10.44 | ManxPower | MrTelephone: or maybe the port changes and it's not reregistering. set the register or nat keepalive on the device to a low setting |
15:10.58 | MrTelephone | i'll give it a try |
15:11.05 | MrTelephone | its not behind a nat though |
15:11.30 | MrTelephone | does anyone have a little digest calculator :P |
15:15.30 | MrTelephone | just checking my logs now and there are some ACL issues |
15:15.31 | MrTelephone | caca |
15:15.48 | MrTelephone | 10.1.3.85 registers with permit acl 10.1.3.0/255.255.255.0 and it denied it |
15:17.43 | ManxPower | Did it register or did it get denied? |
15:19.15 | MrTelephone | nah i took the user out of sip.conf and it gives me that message.. wierd.. it usually says authenticated failure instead |
15:19.28 | MrTelephone | i should upgrade to 1.4 |
15:22.40 | ManxPower | so basically you are just trying random things rather than actually trying to troubleshoot it |
15:23.49 | bakermd | Are there windows apps that run in the system tray and comminucate with Asterisk such that I can see phone numbers on the web highlighted, and when I click one it places the call through asterisk to my desk phone? |
15:24.21 | [TK]D-Fender | bakermd: There are Firefox add-ins for this. Go browse for them |
15:24.27 | *** join/#asterisk synchris (n=synchris@athedsl-4418625.home.otenet.gr) |
15:24.32 | bakermd | [TK]D-Fender, Will Do - Thanks |
15:26.58 | fogo | I've got a SIP phone that will register, but when they try to make a call, it goes unreachable. Shouldn't it be able to stay reachable if it can register initially? |
15:27.56 | [TK]D-Fender | fogo: No, and your description could be a lot fuller |
15:28.24 | *** join/#asterisk dr_gogeta86 (n=dr@81-208-88-100.ip.fastwebnet.it) |
15:28.28 | dr_gogeta86 | hi to alle |
15:28.30 | dr_gogeta86 | *all |
15:28.55 | dr_gogeta86 | who here use asterisk to recive faxes; |
15:29.56 | fogo | [TK]D-Fender: alright, so I have a remote user that has x-lite; I don't know fully what his setup is; most likely NAT'd. * has a public IP, with many other outside users registered w/o problems |
15:30.35 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff-emeritus/bhaal) |
15:30.48 | fogo | [TK]D-Fender: the user in question can register, but when he goes to make a call, his phone changes state to unreachable |
15:31.08 | ManxPower | fogo: do you have qualify=yes ? |
15:31.25 | [TK]D-Fender | probably not... |
15:31.30 | [TK]D-Fender | fogo: ... |
15:31.40 | fogo | I believe so.. checking for sure |
15:31.41 | *** join/#asterisk yidiyuehan (n=yidiyueh@bb121-6-113-252.singnet.com.sg) |
15:32.02 | ManxPower | I assume you have nat=yes too |
15:32.15 | fogo | yes, qualify=yes, nat=yes |
15:32.24 | [TK]D-Fender | [~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 |
15:32.26 | yidiyuehan | hi, guys, any one knows whether it's possible to play a pre-recorded file to the called party before he can talk? |
15:32.44 | ManxPower | fogo: odd. |
15:32.46 | [TK]D-Fender | yidiyuehan: "core show application dial" <- |
15:33.02 | ManxPower | fogo: you are not portforwarding anything on the client router? |
15:33.21 | yidiyuehan | D-Fender, thanks for the guide, if you have some web reference, I will really appreciate it |
15:33.40 | fogo | ManxPower: I don't think they have any ports forwarded currently. |
15:33.59 | [TK]D-Fender | yidiyuehan: web reference? |
15:34.12 | [TK]D-Fender | yidiyuehan: Go to * CLi and read Dial's instructions. |
15:35.08 | yidiyuehan | sure, brother, |
15:35.15 | yidiyuehan | i am looking at it. |
15:35.38 | yidiyuehan | but before that, is it possible to implement what I want? |
15:36.04 | ManxPower | yidiyuehan: It is already a feature of the Dial command. That's why [TK]D-Fender told you to look there. |
15:36.36 | jameswf-home | ~book | yidiyuehan |
15:36.54 | yidiyuehan | ok, stupid I am, thanks to both of you man |
15:37.20 | ManxPower | yidiyuehan: how well it works depends on your interface to the telco. |
15:37.52 | The-Bat | Hie |
15:37.56 | The-Bat | How can I execute a shell script without using System() |
15:38.15 | ManxPower | The-Bat: you could write an AGI. |
15:38.29 | yidiyuehan | i will implement using simple analog telephone lines, once it's answered, one IP phone will pick up the call,and hear a pre-recorded audio file, after that it can start talking. |
15:38.30 | ManxPower | but the real question is "why don't you want to use System() |
15:38.43 | The-Bat | it gives me a APPERROR |
15:38.53 | ManxPower | yidiyuehan: Asterisk does not know when the far ends picks up on analog ports. |
15:38.54 | The-Bat | m unable to debug it |
15:39.03 | ManxPower | The-Bat: then you are doing something wrong. |
15:39.14 | ManxPower | The-Bat: try starting asterisk as "asterisk -cvvv" for testing. |
15:39.18 | ManxPower | then try it. |
15:39.22 | The-Bat | it executes properly wen i run it manually |
15:39.29 | The-Bat | ok let me chk that |
15:39.43 | The-Bat | ManxPower, thx |
15:39.46 | ManxPower | that runs asteirsk in the forground so you should not do that on a production system |
15:40.01 | yidiyuehan | manxpower, asterisk will know once somebody call in the fxo ports am i right? the other end will be the IP phone |
15:40.36 | [TK]D-Fender | yidiyuehan: * will process the call however you set it up to |
15:40.39 | ManxPower | yidiyuehan: yes. |
15:41.01 | ManxPower | it's just calls going OUT an FXO signalled port that has this issue. |
15:41.29 | ManxPower | yidiyuehan: if you sick to IP phones as the destination it should not be a problem. |
15:41.29 | yidiyuehan | manxpower, then it should be fine, as I just want to play to the IP Phone, one way only. |
15:41.54 | yidiyuehan | ManxPower,that's great, i will go and study the Dial application |
15:42.17 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
15:42.33 | *** part/#asterisk JenniferAkemi- (n=akemi@206-248-161-97.dsl.teksavvy.com) |
15:44.08 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
15:44.20 | *** join/#asterisk tengulre (n=tengulre@118.112.226.16) |
15:46.12 | dr_gogeta86 | Anyone here play asterisk to income faxes |
15:47.20 | fogo | ManxPower: back to the previous question - it looks like the soft phone in question has a publicly-addressable IP as well (bypassing their router (NAT)) and they still can register, but when they go to make a call, status changes to unreachable. Any guesses? |
15:47.56 | ManxPower | fogo: many guesses all involving a problem with the client router. |
15:48.17 | ManxPower | you're not doing something stupid like using bindip= or port= in sip.conf? |
15:49.06 | fogo | ManxPower: hmm... port=5060 |
15:49.35 | ManxPower | DONT'T DO THAT |
15:49.35 | Assid | ManxPower: you know any provider like voicepulse BUT white labelled so i can port incoming there.. without really disclosing who the end provider is? |
15:49.43 | ManxPower | Assid: I don't know of any provider that would be suitable for the requirements you have. |
15:49.43 | Assid | someone that handles number portabilityh and gives unlimited incoming but based on channels |
15:49.55 | Assid | hrmm crap |
15:50.05 | ManxPower | maybe you should start contacting carriers rather than asking on this channel every day |
15:50.22 | Assid | im trying |
15:51.09 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
15:51.40 | Assid | cant find a carrier with similar features |
15:52.18 | ManxPower | Assid: I doubt one exists. |
15:52.51 | Assid | thats not good :| |
15:53.20 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
15:54.24 | *** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com) |
15:54.29 | fogo | ManxPower: I nixed the port=; still having the same problem |
15:54.56 | VJFROMGT | i did a yum update and lost my crontab ,, anyone know how i can get it back (backup has new version) |
15:55.03 | ManxPower | fogo: put your sip.conf on pastbin.con masking ONLY the passwords |
15:56.38 | fogo | ManxPower: http://pastebin.ca/1197300 |
15:57.13 | [TK]D-Fender | lol |
15:57.17 | ManxPower | fogo: if that is your entire sip.conf no wonder you are hving problems |
15:57.18 | [TK]D-Fender | ManxPower: run :) |
15:57.26 | fogo | oops.. entire |
15:57.38 | fogo | :) |
15:57.38 | ManxPower | [TK]D-Fender: if he can't follow simple instructions I doubt he can be helped. |
15:57.57 | [TK]D-Fender | ManxPower: You shouldn't need to ask for the full file. You should already know what you're looking at. |
15:58.25 | ManxPower | [TK]D-Fender: I'll go ask Cleo |
15:58.41 | [TK]D-Fender | ManxPower: Don't need to be psychic... |
15:58.44 | fogo | well, I can post the entire file, but it contains roughly 1000 sip entries in it |
15:59.10 | ManxPower | fogo: nevermind. You are using FreePBX or some other sort of spawn of Satan GUI, aren't you? |
15:59.15 | [TK]D-Fender | fogo: What he wants to see is [general] |
15:59.29 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
15:59.47 | fogo | ManxPower: yes, I am using a spawn of satan GUI |
15:59.54 | ManxPower | Therefore I will not help you. |
16:00.05 | ManxPower | Best of luck and maybe you should ask on the correct channel. |
16:00.14 | fogo | alright. thanks :) |
16:01.04 | *** join/#asterisk wiscados (n=mint@81.25.184.155) |
16:16.51 | *** join/#asterisk doolph (n=doolph@190.141.69.38) |
16:16.55 | doolph | x |
16:17.43 | *** join/#asterisk mighty-d (i=500@63.58.83.190.static.coldecon.com) |
16:17.45 | mighty-d | Hi |
16:17.50 | doolph | hi |
16:19.24 | mighty-d | im learning all-i-can from asterisk, and im not sure about something, it is codec-related, there are a lot of card vendors in the market, is there any special consideration with codecs like 711, GSM, etc.. with the cards, i mean is the codec applied by the card or is this done by the OS ? |
16:19.57 | [TK]D-Fender | mighty-d: Codecs have nothing to do with cards for your server |
16:20.15 | mighty-d | as i tought, thanks TK-Fender :) |
16:23.03 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
16:29.05 | mighty-d | What should i use to protect analog cards from electrical hazzards? |
16:29.25 | jaytee | line surge suppressors |
16:30.10 | jaytee | and if you have POTS lines from your telco they "should" be wired already with a PET (protected entry terminal) |
16:30.49 | jaytee | but alot of telcos are cheap and don't install them except for their larger customer accounts. |
16:31.01 | doolph | lol |
16:31.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
16:32.35 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
16:32.38 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.43) |
16:32.39 | jaytee | you can also get CAT5 suppressors too for ethernet connections. we use em here at the zoo because we get surges all the time when there are lightning strikes nearby. |
16:34.39 | drmessano | APC |
16:34.49 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
16:35.08 | jaytee | that's what we use |
16:35.15 | jaytee | I swear by APC |
16:35.33 | jaytee | Tripplite,.....not so much |
16:36.25 | *** join/#asterisk Xaviertoor (n=Xavierto@189-015-92-210.xd-dynamic.ctbcnetsuper.com.br) |
16:36.51 | *** join/#asterisk ManxPower (n=manxpowe@130.sub-75-249-91.myvzw.com) |
16:37.54 | jaytee | had a flood once in a basement server room due to a broken line in an upstairs dishwasher. Water was almost touching the motherboards of the 3 primary servers before we got in to kill the power. 4 computers plugged into an APC surge suppressor were still good because the APC gave up it's life to protect them. Called APC to tell em how good their stuff was and they said they'd send a replacement. I said, but it wasn't your products fault and it did |
16:37.54 | jaytee | it's job. |
16:38.19 | jaytee | they said it didn't matter, still covered under the warranty. |
16:42.22 | drmessano | nice |
16:42.58 | drmessano | I have had two APC's almost catch on fire, yet I still use them because the others have saved my bacon more often than not |
16:43.29 | *** join/#asterisk korihor (n=korihor@201.211.168.130) |
16:43.44 | denon | we run a ton of APC, but I trust Liebert a lot more |
16:47.35 | Juggie | our system @ work is leibert |
16:47.52 | Juggie | we have enough batteries to power an entire lab for 12hrs :) |
16:48.02 | Juggie | and thats only incase the generator doesnt start |
16:50.20 | *** join/#asterisk JG7634 (n=chatzill@69.62.244.244) |
16:50.33 | *** join/#asterisk Whisk (n=Whisk@82-44-94-242.cable.ubr04.croy.blueyonder.co.uk) |
16:51.24 | tvirus | I created 2 queues, http://rafb.net/p/KhcSQo42.html . My goal is to have it ring queue 901 if no one picks up on 900. This is inside extensions.conf http://rafb.net/p/MyRlX298.html . Queue 900 works fine but if I dial 901, I get a busy signal and if I dial 900 and wait it just rings and then hangs up after 30 seconds or so. |
16:53.16 | *** join/#asterisk mv2 (n=mv@83.240.229.38) |
16:53.28 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
16:53.50 | jaytee | all these doommongerers going on about the LHC firing up wednesday. "oh, no! we're all gonna die!!!". I swear cranial anal embedding has reached epidemic proportions. |
16:53.55 | *** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net) |
16:54.01 | mv2 | can someone explain AST-2008-003 ? |
16:54.22 | tvirus | I'd rather die from a black hole opening in the LHC than from a car accident. |
16:54.29 | *** join/#asterisk Levonk (n=lk@76.237.13.156) |
16:55.05 | jaytee | mv2, yeah it's 3 letters and a group of 4 numbers and then a group of three numbers all seperated by a dash. :-) |
16:55.51 | mv2 | jaytee:thanks |
16:56.24 | Kobaz | is there an easy way to play ringback (instead of music) when dialing into a queue, and periodicly play a message saying please hold blah blah |
16:56.42 | Kobaz | i can pass an r to the queue for ringback, but then how would i play a message |
16:59.07 | *** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net) |
17:00.29 | AndyML | hey there. I have a VoiceMailMain() channel stuck. any ideas on how to kill it short of restarting asterisk? |
17:00.54 | Qwell | AndyML: soft hangup <channel> from the CLI |
17:01.04 | x86 | omg I just a whole fleshlight |
17:02.43 | JG7634 | My company has two offices both with nortel norstar digital pbx and digital phones (52/16 phone stations). The two offices are trunked with Citel Gateway 2 and Citel Extender 7000. We have a 5Mb ethernet between the offices. We would like to replace the trunking Citel equipment with a Asterisk Trunking Solution and keep using the existing pbx phone systems with extension dialing still... |
17:02.45 | Qwell | ...what? |
17:02.45 | JG7634 | ...working. Is this possible with Asterisk? Thanks. |
17:03.05 | Qwell | JG7634: how are they connected? what protocol? |
17:03.11 | ManxPower | Kobaz: you can't have ringing and a message |
17:03.19 | Kobaz | ManxPower: that's what i thought |
17:03.33 | Kobaz | i can do a ring moh though |
17:03.50 | ManxPower | that's not ring, that moh that sounds like ringback |
17:03.55 | [TK]D-Fender | tvirus: It'd be real nice if your 901 exten had a priority 1 in there.... |
17:04.03 | JG7634 | qwell: rvp over ip |
17:04.12 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
17:04.19 | ManxPower | JG7634: Asterisk does not support RVP |
17:04.22 | Qwell | JG7634: then no, Asterisk doesn't support that |
17:04.47 | [TK]D-Fender | AndyML: "soft hangup [channelwithoutbraces]" |
17:04.56 | drmessano | Rob Van .. oh nevermind |
17:04.58 | Qwell | []=optional :p |
17:05.12 | *** join/#asterisk c4t3l (n=root@74.95.210.124) |
17:05.30 | x86 | http://forums.joerogan.net/showthread.php?t=74612 |
17:05.58 | JG7634 | qwell: if we are replacing the citel equipment we dont have to use the same protocol right? what would Asterisk use? |
17:05.59 | jaytee | the devs should build in a couple of settable parameters in queues.conf to allow for RAN announcements. Like ranmessage1= filename, i.e. "Your call is totally unimportant to us, please continue to be an idiot and hold" and a ranmessageinterval=30 for every thirty seconds. |
17:06.13 | Qwell | JG7634: presumably SIP, if the other box supports it |
17:06.24 | Qwell | JG7634: are you replacing both sides? maybe I misunderstood |
17:06.52 | jaytee | I can do first and second level RAN announcements, interflow, overflow and nite service treatments for Nortel ACD queues but * is kinda "weak" in that area. |
17:07.25 | JG7634 | qwell: keeping both offices digital pbx and phones. Just thinking to use Asterisk to create a link between the offices. |
17:07.46 | Qwell | so, 1 Asterisk box at each office, and a PRI link to the PBX or something? |
17:08.20 | JG7634 | qwell: yes i think on those lines |
17:08.43 | Qwell | SIP or IAX2 probably |
17:08.54 | ManxPower | most PBXs are very limited in their ability to integrate with 3rd party stuff like Asterisk |
17:09.05 | JG7634 | qwell: the citel equipment current links the two offices but the equipment fails often |
17:09.41 | JG7634 | qwell: over point to point ethernet |
17:09.42 | ManxPower | I thought Citel was a device to convert digital phones into SIP? |
17:09.51 | c4t3l | yes it is |
17:10.11 | JG7634 | manxPower: not the gateway 2 and extender 7000 they are propriatery I thought |
17:10.27 | tvirus | That fixed it from not working, thanks, TK. |
17:10.31 | ManxPower | JG7634: Don't think. Know or you will waste money |
17:11.28 | ManxPower | JG7634: how is the Citel connecting to your PBX. |
17:11.38 | c4t3l | i demo-ed the citel gateway equipment. It seemed pretty cool. But I've never seen it work in the real world |
17:12.23 | JG7634 | manx: pri t1 card |
17:12.41 | ManxPower | JG7634: and you are SURE it's a PRI, not a channelized T-1? |
17:12.56 | ManxPower | because nortel charges like $3,000 extra to enable PRI support on their PBXs |
17:14.01 | JG7634 | manx: its a large multipin plug about the size of a thumb, rj-21 cat 3 punchdown block |
17:14.14 | ManxPower | that would be analog |
17:14.26 | ManxPower | PRI uses exactly 4 wires (2-pair) |
17:14.26 | Qwell | eww... |
17:14.41 | *** part/#asterisk mv2 (n=mv@83.240.229.38) |
17:14.41 | ManxPower | It uses an RJ-45/RJ-48C |
17:15.34 | ManxPower | So you have PBX <- 25 pair AMP -> Citel? |
17:16.11 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
17:17.10 | ManxPower | JG7634: I don't see a "gateway 2" model from Citel |
17:17.33 | JG7634 | manx: PBX <- 25 pair AMP -> Citel Gateway II-> 5Mb Ethernet <- Citel Extender 7000 <- 25 pair AMP <- PBX |
17:17.51 | Qwell | that's nasty |
17:18.03 | ManxPower | JG7634: looks analog to me. |
17:18.08 | [TK]D-Fender | YUP |
17:18.14 | ManxPower | what EXACT model is the Citel Gateway II |
17:18.14 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
17:18.20 | AndyML | [TK]D-Fender: soft hangup doesn't seem to work for this channel. if i showed you a CLI output do you think you could offer anything else? |
17:18.24 | JG7634 | manx: ok yes it is Analog equipment with a digital line |
17:18.36 | ManxPower | JG7634: it will never ever work with Asterisk |
17:18.54 | [TK]D-Fender | AndyML: CLI of your attempt & the channel dump would be nice. |
17:19.00 | AndyML | will do |
17:19.14 | Qwell | sure it would.. TDM2400 does AMP |
17:19.26 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
17:20.08 | ManxPower | you don't want to use analog |
17:20.22 | ManxPower | Qwell: and since it's a Citel it's not analog. |
17:20.33 | Qwell | oh... |
17:20.35 | ManxPower | Looks like they use PBX station ports or something like that. |
17:20.39 | Qwell | yuck |
17:20.40 | [TK]D-Fender | ManxPower: makes you wonder WTF is going on actually... |
17:20.47 | AndyML | [TK]D-Fender: http://pastebin.ca/1197371 |
17:20.52 | [TK]D-Fender | ManxPower: Could indeed be 2-wire station ports |
17:21.07 | JG7634 | manx: http://www.citel.com/Products/EXTender.asp ill get you model number |
17:21.44 | tvirus | What's the proper syntax (1.4.x) for a queue to go to the next queue if no one picks up in 25 seconds? exten => 900,1,Queue({$EXTEN}|t|||25) |
17:21.47 | [TK]D-Fender | AndyML: Try an AMI redirect to a dead-end |
17:22.02 | [TK]D-Fender | tvirus: "core show application queue" |
17:22.08 | AndyML | [TK]D-Fender: not sure what you mean. |
17:22.10 | ManxPower | http://www.citel.com/Products/EXTender/Citel_PBX_Gateway_II.asp |
17:22.28 | [TK]D-Fender | tvirus: Whats the point of doing queue's for so short a period of time? |
17:22.38 | tvirus | Call center. |
17:22.38 | [TK]D-Fender | AndyML: Go read up on AMI |
17:22.39 | AndyML | [TK]D-Fender: can you just grab a channel in AMI and redirect it? |
17:22.46 | tvirus | To escalate a call if no one answers |
17:22.50 | [TK]D-Fender | AndyML: thats the idea |
17:22.55 | AndyML | i'm familiar with AMI. just not with every feature. ok - will do. |
17:23.09 | [TK]D-Fender | tvirus: first should just be a Dial then |
17:23.17 | linuxer_igor | help, I have a problema with DAHDI |
17:23.26 | *** join/#asterisk Ziaeon (n=Ziaeon@c-71-57-142-206.hsd1.fl.comcast.net) |
17:23.29 | linuxer_igor | update my server today for 1.0.6rc4 and the DAHDI don't work |
17:23.40 | linuxer_igor | stay up but do not register |
17:24.03 | Ziaeon | Why does random play not work for me? It's all set up properly, random = yes, it just plays the same song over and over. The strangest thing is that it's not even the first song, alphabetically. |
17:24.35 | Ziaeon | (For MOH) |
17:25.19 | ManxPower | http://metrocomminc.com/images/verso/extender_pbxgateway_ii.pdf |
17:25.36 | ManxPower | notice the weird "telephony interface" on the picture of the back. |
17:26.16 | JG7634 | Manx: yes |
17:27.50 | JG7634 | manx: thats where the pbx connects in |
17:29.01 | ManxPower | JG7634: Best of luck. |
17:29.24 | [TK]D-Fender | Oh yeah, thats a Norstar Digital amphenol trunk device all right... |
17:29.33 | JG7634 | manx: we want to get rid of that box though what are some equipment that can get us... analog pbx -> ? -> Asterisk -> 5mb ethernet |
17:29.52 | [TK]D-Fender | JG7634: If you don't have a real T1 port on that thing you're in for a real PITA job |
17:30.09 | JG7634 | T1 port on the pbx? |
17:30.13 | [TK]D-Fender | yes |
17:30.15 | ManxPower | [TK]D-Fender: no wonder it doesn't work well. |
17:30.23 | [TK]D-Fender | ManxPower: Not an excuse for it |
17:30.48 | ManxPower | [TK]D-Fender: We are talking about Citel here. |
17:30.55 | [TK]D-Fender | JG7634: How many lines / stations at each site? |
17:31.11 | JG7634 | fender: 56/16 |
17:31.24 | ManxPower | you have 56 lines and only 16 stations? |
17:31.43 | ManxPower | might want to get rid of some of those lines. |
17:31.48 | [TK]D-Fender | JG7634: You're almost better off ripping it all out... |
17:31.50 | JG7634 | woops, 52 stations 13 lines |
17:32.01 | ManxPower | JG7634: stop being inaccurate. |
17:32.05 | JG7634 | other office 16 stations, dont know lines |
17:32.45 | Qwell | JG7634: might as well rip everything out, and replace it all |
17:32.46 | JG7634 | manx: i dont know everything, good luck on accuracy |
17:32.46 | tvirus | [TK]D-Fender: It's a queue because they dial in and get the voice menu and push 1 or 2 for Support/Sales and get into that queue. |
17:33.04 | ManxPower | JG7634: then you are guaranteed to fail. |
17:33.16 | [TK]D-Fender | tvirus: Its a queues because YOU send them there. |
17:33.26 | JG7634 | manx: this just feasability, professional will do lol |
17:34.04 | ManxPower | JG7634: you can't find out feasibility if you don't know exactly how things are set up and what protocols are used. |
17:34.29 | JG7634 | manx: we have to start somewhere |
17:34.44 | ManxPower | Do you really want to find out it's not going to work when you blow up the asterisk card because you plugged it into a station line? |
17:35.04 | ManxPower | or blow the line on the PBX? |
17:35.04 | [TK]D-Fender | JG7634: Go see if you have a real PRI port available on it. |
17:35.30 | JG7634 | fender: ok checking for pri |
17:35.31 | ManxPower | This PRI port would look similar a standard Ethernet port |
17:35.38 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
17:36.52 | *** join/#asterisk luca`gervasi (n=ashura@host76-170-dynamic.21-87-r.retail.telecomitalia.it) |
17:36.55 | luca`gervasi | Hallo |
17:37.29 | luca`gervasi | i need to test my line...can someone tell me a simple function to say "All right" ? |
17:37.44 | [TK]D-Fender | luca`gervasi: test your line by USING it. |
17:37.52 | ManxPower | [TK]D-Fender: on a Nortel MICS you have to pay extra (thousands of dollars extra) for PRI and even MORE for support for "PRI Tie-line" |
17:38.10 | [TK]D-Fender | ManxPower: Life sucks but rarely swallows |
17:38.15 | luca`gervasi | well, when i try to dial a number, it says "no such extension found" :D |
17:38.52 | [TK]D-Fender | luca`gervasi: then you should go look at what context its looking for that exten in and what exten it is looking for. Then maybe you'll see why it isn't finding a match |
17:40.07 | JG7634 | fender: our pbx has a T1 connected into a PRI |
17:40.37 | Qwell | you could put Asterisk in front of it... |
17:41.04 | Qwell | Telco > Asterisk > WAN > Asterisk > Telco AND, Telco > Asterisk > existing PBX > Digital phones (on both sides) |
17:41.05 | JG7634 | qwell: so t1>Asterisk>PBX? |
17:41.07 | [TK]D-Fender | JG7634: I also said "available" <- |
17:41.20 | Qwell | [TK]D-Fender: depending on how he architects it, it would become "available" |
17:41.36 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
17:41.37 | JG7634 | fender: looks like the only pri and its in use |
17:41.46 | *** join/#asterisk bug2000 (n=bug@unaffiliated/bug2000) |
17:41.51 | Qwell | then you could eventually phase out the existing PBX/phones |
17:41.59 | Qwell | without having to rearchitect it |
17:42.06 | [TK]D-Fender | Qwell: Sure, he could spend a lot more for a dual port card and have to lose channels going to the other side. |
17:42.18 | [TK]D-Fender | JG7634: is this on BOTH sides? |
17:42.23 | Qwell | or he could spend a bajillion dollars on a T1 card for the other PBX |
17:42.39 | [TK]D-Fender | Qwell: That's likely necessary as well :) |
17:43.48 | JG7634 | fender: i dont know about the other site, i'll make a phone call |
17:45.07 | *** part/#asterisk bug2000 (n=bug@unaffiliated/bug2000) |
17:45.17 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
17:46.06 | JG7634 | fender: looks like no pri in small office |
17:46.25 | JG7634 | fender: I think boss would be willing to gut small office |
17:48.32 | JG7634 | looks like we would have to spend more money than just tow asterisk boxes anyway |
17:48.39 | JG7634 | *two |
17:49.11 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
17:49.25 | [TK]D-Fender | JG7634: no, but more than clsoe enough to consider a complete replacement. |
17:52.17 | JG7634 | Thanks for the help |
17:53.32 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
17:56.05 | jaytee | my system is setup like this from telco -> PRI -> Nortel Option 11C inbound and any 4 digit extension on * exists as 4 digit extension on the Nortel that is externally callforwarded to the same extension with a 555 prefix which gets matched with _555XXXX and gets the first 3 of ${EXTEN} stripped out in the incoming context and then matched against my internal * VOIP numbers from the outbound PRI from my Nortel pbx that connects to * and calls to |
17:56.05 | jaytee | <PROTECTED> |
17:57.17 | jaytee | works like a charm and allowed me to avoid adding an additional T1 card to the Nortel pbx. Of course you need to be sure you have excess channel capacity to accomodate it. |
18:01.20 | *** join/#asterisk arpu (n=arpu@chello084114025240.14.vie.surfer.at) |
18:06.52 | *** join/#asterisk Levonk (n=lk@adsl-76-238-249-188.dsl.lsan03.sbcglobal.net) |
18:09.52 | *** join/#asterisk trumee (n=trumee@cpc1-seve11-0-0-cust723.popl.cable.ntl.com) |
18:10.22 | trumee | guys, i have lot of adresses in my kde addressbook. How can i import these into asterisk? |
18:11.44 | ManxPower | trumee: Huh? |
18:12.01 | ManxPower | trumee: you don't use address books in Asterisk |
18:12.11 | trumee | ManxPower, i want a call notification on the computer. |
18:12.16 | ManxPower | you might on thephone -- but that's not an Asterisk thin. |
18:12.41 | ManxPower | trumee: then build something yourself or look on the Wiki to see if there's something like that already built |
18:13.17 | ManxPower | Sounds to me like you want "click to dial" type of thine. |
18:13.17 | ManxPower | thing |
18:13.26 | trumee | ManxPower, ok lemme check |
18:13.46 | ManxPower | trumee: it is NOT a feature that comes with Asterisk |
18:14.03 | [TK]D-Fender | Asterisk is not a phonebook and is not a phone. It does not store any of this stuff... |
18:14.38 | [TK]D-Fender | ManxPower: How do I set my bicycle on "puree"? |
18:15.20 | jaytee | [TK]D-Fender, when I press the Enter key, exactly what happens next? :-) |
18:15.33 | [TK]D-Fender | jaytee: Where is the "Any" key? |
18:15.43 | jaytee | remember those Time-Life computer book commercials? |
18:15.44 | WimpMan | Oh, I've been doing it all the time. Lookup the number and broadcast number and name on the LAN to get some popups. |
18:16.07 | *** join/#asterisk geoff2010 (n=geoffrey@75.150.14.50) |
18:16.15 | *** join/#asterisk Bananaskin (n=Banana@93-97-226-229.zone5.bethere.co.uk) |
18:16.19 | jaytee | Wimpman, what are you using to facilitate that? |
18:19.41 | ManxPower | [TK]D-Fender: TAPI is what I suspect he needs |
18:20.07 | jaytee | but TAPI only runs on Windows unless he runs it in (yuck) WINE |
18:22.03 | [TK]D-Fender | Which clearly matches his description of "KDE".... |
18:22.09 | jaytee | I'm using a SIP TAPI plugin that let's me call people in my Outlook Contacts and in Internet Explorer for websites that support click to call. |
18:22.17 | tvirus | [TK]D-Fender: Is there a proper way to cascade queues? After the 15-25 second time out it just hangs up on the person. :\ |
18:22.45 | [TK]D-Fender | tvirus: It doesn't. Pastebin <- |
18:23.06 | *** join/#asterisk reno_rr (n=reno@maillrar.vestcom.com) |
18:24.17 | AndyML | [TK]D-Fender: problems... http://pastebin.ca/1197421 |
18:24.21 | *** join/#asterisk angryuser (i=c392f452@gateway/web/ajax/mibbit.com/x-e3857a274a2fe382) |
18:24.23 | jaytee | I was wondering about that because if I want to recreate the functionality of my Nortel ACD queues when I move them to * I'm going to have to cascade them with a Playback message in between. Can an agent be logged into multiple queues at the same time? |
18:25.12 | AndyML | [TK]D-Fender: you'll note there are 3 new locked up voicemail channels... |
18:25.39 | WimpMan | jaytee: I've been using that system for ages. Was built for I4L originally. I wrote a little stand alone sender I called vial System() from *. |
18:26.05 | tvirus | [TK]D-Fender: Extensions.conf http://rafb.net/p/wx5reo46.html | Queues.conf http://rafb.net/p/V6f4c941.html |
18:26.37 | Ziaeon | how do you specify music on hold for outbound calls per extension |
18:27.32 | ManxPower | zamba: in the Dial line |
18:27.38 | *** join/#asterisk Trekk3r (n=Stargaze@c-67-163-135-32.hsd1.pa.comcast.net) |
18:27.46 | ManxPower | sorry, you mean which class, right? |
18:27.57 | jaytee | WimpMan, sounds cool. I'm just trying to figure out how to send MWI to my Polycoms from Exchange Unified Messaging without using a third party app like what Geomant offers. |
18:27.58 | Ziaeon | yeah |
18:28.21 | ManxPower | zamba: "show applications like music" and channelvariables.txt |
18:28.30 | ManxPower | I don't recall if it's a variable or an app. |
18:29.15 | Ziaeon | alright |
18:29.48 | jaytee | there's the MUSICCLASS function and a sip parameter you can define per phone in sip.conf also |
18:30.13 | ManxPower | jaytee: that is for OUTGOING calls from the device |
18:30.33 | ManxPower | not outgoing from Asterisk |
18:30.39 | ManxPower | which would be incoming to the device. |
18:31.59 | [TK]D-Fender | tvirus: Wheres the failed call's CLI output? |
18:32.09 | Ziaeon | From outside -> asterisk -> myext, my MOH class works fine. From myext -> asterisk -> mycoworkersext, default MOH is used. From myext -> asterisk -> outside number, default MOH is used. |
18:32.56 | Ziaeon | my MOH class is defined in my inbound route and per exten |
18:33.43 | Ziaeon | but it seems it wants to use the outbound routes MOH, which is inappropriate because that would be global and not per who is calling out |
18:34.00 | *** join/#asterisk luckyaba (n=lucky@ip68-6-98-146.sb.sd.cox.net) |
18:34.33 | ManxPower | use the musicclass before the Dial |
18:34.51 | Ziaeon | im going to try that now |
18:35.01 | tvirus | One moment. |
18:36.39 | jaytee | and musicclass has been deprecated in favor of CHANNEL(musicclass) |
18:36.41 | tvirus | http://rafb.net/p/pa7HOn69.html |
18:38.07 | Ziaeon | alright, thanks, ill go from here |
18:40.26 | [TK]D-Fender | tvirus: Does exactly what its supposed to. Drops them in a queue and noone answered so it quit and the ran out of things to do and hung up. |
18:41.26 | tvirus | How can I make it go to the next queue :-\ |
18:41.43 | Trekk3r | Would anyone happen to have a thought on why I wouldn't hear anything when I call into my asterisk even though asterisk says it is playing a file (weasels )? |
18:41.58 | jaytee | your RTP configuration? |
18:42.11 | [TK]D-Fender | tvirus: By actually having more to do in that EXTENSION. |
18:42.11 | jaytee | or your firewall blocking the RTP ports |
18:42.28 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-161-97.dsl.teksavvy.com) |
18:42.29 | Trekk3r | all ports are open, no firewall. |
18:42.35 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
18:42.59 | jaytee | Trekk3r, any errors or warnings on the CLI? |
18:43.33 | Trekk3r | no, everything *looks* fine |
18:44.03 | tvirus | Do I need to use a GotoIf() in there or something? |
18:44.19 | [TK]D-Fender | tvirus: No, you need to have more to do in your extension. |
18:44.24 | jaytee | Trekk3r, calling from an IP phone and if so does 2 way audio work when calling another IP phone? |
18:44.49 | Trekk3r | I'm calling from a land line into a SIP |
18:45.36 | jaytee | you mean from a landline to an ITSP provided DID number that routes into your * server? |
18:46.05 | Trekk3r | sounds right? (i'm not the most knowledgeable on the lingo) |
18:46.32 | jaytee | do you have IP phones on the local net your * server is on? |
18:46.33 | Trekk3r | all I'm trying to get * to do is answer the phone and record an wav file, nothing else. |
18:46.59 | jaytee | you said you couldn't hear it play ttweasels. Hearing and recording are two different things. |
18:47.30 | Trekk3r | well it plays ttweasels before it records (which its not doing either). i figured i'd attack one problem at a time |
18:47.46 | jaytee | and for lingo what I'm trying to find out is HOW are calls getting to your * box from the outside? POTS or SIP? |
18:47.47 | tvirus | [TK]D-Fender: Something like exten = 900,n,Dial(SIP/901) ? |
18:47.52 | Trekk3r | SIP |
18:47.59 | Trekk3r | callcentric.com account |
18:48.00 | [TK]D-Fender | tvirus: SOMETHING |
18:48.13 | [TK]D-Fender | tiveright now you go into 1 queue for 15 seconds and thats it. |
18:48.47 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
18:49.01 | tvirus | [TK]D-Fender: I don't know what that SOMETHING is. I have no idea, which is why I'm here. Give me a hint. |
18:49.01 | Trekk3r | jay, could i PM you a few lines of whats going on here? that might make what i'm doing more clear |
18:49.43 | jaytee | [TK]D-Fender, so in that scenario would Dial(queue#) then Playback(some message) then Dial(newqueue#) work? |
18:49.55 | [TK]D-Fender | tvirus: What part of "you only have 1 priority in your exten and if you wanted it to do something following leaving that queue maybe you should add MORE priorities" are you having trouble with? |
18:50.01 | jaytee | Trekk3r, use a pastebin link |
18:50.15 | jaytee | ~pb |
18:50.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
18:51.18 | tvirus | You didn't say all of that... :| |
18:52.11 | Trekk3r | http://pastebin.com/d1649900e |
18:52.29 | jaytee | scrolls backwards.........yep, he did say that.......twice. |
18:53.06 | [TK]D-Fender | jaytee: nobody reads the BIG PRINT |
18:53.32 | tvirus | Yay, now it works. |
18:53.34 | Trekk3r | thats what color is for :P |
18:53.36 | tvirus | Thanks [TK]D-Fender :) |
18:53.47 | *** part/#asterisk MACscr (n=Mark@c-98-214-92-79.hsd1.il.comcast.net) |
18:53.48 | jaytee | [TK]D-Fender, Oh? so now we're supposed to ignore the BIG PRINT and the fine print? |
18:53.50 | tvirus | I didn't think about adding another priority for it. |
18:54.01 | tvirus | [TK]D-Fender: What's your address so I can send you some home made cookies? |
18:54.11 | [TK]D-Fender | jaytee: I'm going monospace-medium from now on |
18:54.16 | jaytee | hahaha |
18:54.33 | tvirus | Or do you prefer brownies. |
18:54.35 | tvirus | Your call. |
18:55.38 | jaytee | [TK]D-Fender, just use Dingbats and say "screw 'em!" like Anthony Hopkins in Legends of the Fall |
18:55.43 | [TK]D-Fender | tvirus: FDA wouldn't let it cross the border, and both are against my diet |
18:56.13 | jaytee | otherwise I'd be FedEx'ing him some fresh roasted coffee |
18:56.13 | *** join/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
18:56.37 | Alan_Hicks | Howdy. |
18:57.10 | tvirus | I could always use sendthemcookies.com :D |
18:57.32 | Trekk3r | jay, did that link i paste mean anything to ya? |
18:57.44 | Alan_Hicks | I upgraded from asterisk-1.4.10 (I think) to 1.4.21.2. Somewhere along the line my Polycom phones stopped doing paging correctly. |
18:58.06 | Alan_Hicks | They no longer auto-answer. sip.cfg and extensions.conf to follow. |
18:58.16 | *** join/#asterisk n3hxs (n=HAMming@151.196.87.132) |
18:58.56 | jaytee | Trekk3r, that pastebin is chopped off on the right margin and missing parts of the lines or your console window is too narrow. You're getting a Warning about maximum retries but it's chopped. |
19:00.02 | jaytee | Trekk3r, and if you don't mind me asking for a second time. Do you have 2 IP hard or softphones on the same local net as your * box you could test AUDIO with? |
19:00.16 | Trekk3r | sorry, http://pastebin.com/d80cc811 |
19:00.26 | Trekk3r | no i don't |
19:00.33 | Alan_Hicks | http://lizella.net/asterisk/sip.cfg http://lizella.net/asterisk/extensions.conf |
19:00.39 | Trekk3r | this is not a home machine either, its a hosted VPS |
19:00.50 | Alan_Hicks | ^^^^^ Pastebin was too much trouble. :-) |
19:01.36 | Alan_Hicks | When extension 899 is dialed, the phones ring normally rather than auto-answer as they should. Has something changed in asterisk between those two versions? |
19:01.49 | Alan_Hicks | Perhaps the "SIPAddHeader" has changed somehow? |
19:02.59 | jaytee | Trekk3r, something is messed up with your connection to your ITSP. read and follow this guide using whatever configuration parameters callcentric.com has provided you for your account. |
19:03.03 | jaytee | ~sipnat |
19:03.03 | jbot | [~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 |
19:04.04 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
19:04.19 | Trekk3r | i'll give it a shot, thanks |
19:05.25 | Kobaz | with a queue, how would i not play the "please hold.." message after 4 seconds of moh |
19:05.36 | Kobaz | i want the full periodic-announce-frequency to pass before playing any tracks |
19:05.41 | jaytee | Trekk3r, sorry, I'd missed the line about your * box being a hosted VPS. It's looking like audio is the least of your problems :-) |
19:07.04 | Trekk3r | well my home connection cannot handle the traffic i need to take so :P |
19:07.59 | [TK]D-Fender | Alan_Hicks: Short list : quotes areound Auto-Answer = bad, 2nd did you even set this up in your Polycom provisioning? 3rd Stop using priority jumping, its antiquated |
19:08.23 | x86 | Alan_Hicks: usually auto-answer is phone dependant |
19:08.32 | x86 | Alan_Hicks: check the settings on your phones |
19:08.39 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
19:08.49 | x86 | Alan_Hicks: if they require a special header, there is an asterisk application called sipaddheader, iirc |
19:08.51 | Alan_Hicks | The settings on the phones haven't changed. |
19:09.03 | x86 | show application SIPADDHEADER |
19:09.08 | [TK]D-Fender | Alan_Hicks: rest applies |
19:09.08 | Alan_Hicks | And it was working before the asterisk upgrade with the same config files. |
19:09.30 | Alan_Hicks | However, I did add the quotes in SIPAddHeader hoping it would fix it and apparently forgot to revert that change. |
19:09.32 | implicit | anyone know what the acronym COB means in a telecom context? |
19:09.46 | DarKnesS_WolF | x86: go sleep ! |
19:10.29 | x86 | DarKnesS_WolF: in the middle of the work day? :) |
19:10.29 | DarKnesS_WolF | x86: so what :P |
19:10.29 | [TK]D-Fender | Alan_Hicks: fix that, then show us the failed call with SIP debug |
19:10.29 | x86 | DarKnesS_WolF: wish i could get paid for sleeping ;) |
19:10.29 | jaytee | implicit, usually Class Of Business |
19:10.35 | DarKnesS_WolF | x86: nop :P ur fired :D |
19:10.43 | x86 | ;) |
19:11.23 | implicit | jaytee: haha, people using too many acronyms |
19:11.26 | implicit | they meant close of business |
19:11.27 | implicit | hahahahaah |
19:11.31 | Alan_Hicks | Just a minute. Big pastebin. |
19:11.36 | jaytee | that too, |
19:11.42 | *** join/#asterisk logicwrath (n=no@68.41.24.98) |
19:12.10 | logicwrath | how do i restart the httpd daemon in asterisk now |
19:12.23 | logicwrath | its taking too long to respond and not showing me the web GUI |
19:12.38 | [TK]D-Fender | logicwrath: You ask in #asterisknow |
19:12.45 | Trekk3r | jay, i have a dedicated IP... this nat stuff still gonna apply for me? |
19:12.55 | jaytee | Trekk3r, nope |
19:12.55 | logicwrath | no one is ever in asterisknow |
19:13.19 | [TK]D-Fender | logicwrath: Still fails to be our problem. This is a distro question. |
19:13.19 | Alan_Hicks | http://lizella.net/asterisk/sip_debug.txt |
19:13.21 | jaytee | Trekk3r, don't know how your VSP provider has their stuff setup, you'd have to work with them |
19:13.53 | Trekk3r | fundamentally this is some sort of port problem you think then? |
19:15.01 | [TK]D-Fender | Alan_Hicks: And what happens on the actual phone? |
19:15.27 | Alan_Hicks | It rings normally and doesn't auto-answer. In that debug attempt I manually answered, then hung up. |
19:15.46 | ManxPower | Alan_Hicks: quotes are not part of anything |
19:15.47 | [TK]D-Fender | Alan_Hicks: The header is in there just fine. Must be your phone config. |
19:15.56 | x86 | logicwrath: unload http, load http? |
19:16.04 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
19:16.13 | Alan_Hicks | Hmm.... |
19:16.30 | Alan_Hicks | I don't know what could have changed... I didn't touch /home/polycom at all... |
19:16.34 | gaetronik | Hi, i've a question about the L option of dial |
19:16.42 | Alan_Hicks | Thanks for your help. I'll dig into the sip.cfg further and try to figure this out. |
19:17.22 | x86 | Alan_Hicks: it's gotta be a phone issue at this point |
19:17.33 | Alan_Hicks | nods. |
19:17.40 | jaytee | logicwrath, doubt if it works CoronaryWard or whatever that dogshit distro *NOW uses but you could try service httpd restart |
19:17.50 | gaetronik | if i make a dial(SIP/EXTEN@ipaddr) |
19:17.55 | logicwrath | http show status showing funny port, i got to it by using :8088 |
19:18.22 | logicwrath | for some reason it did not redirect me to 8088 from 80 so, I will research on my own. thanks |
19:18.22 | gaetronik | the time of the l start from the dial not from the time where the communication start really between two human |
19:18.50 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
19:20.56 | ManxPower | gaetronik: we have no idea what you are talking about. |
19:21.45 | gaetronik | ManxPower, i will try to xplain it more clearly |
19:22.01 | seanbright | he is saying the time used is from the start of the dial, not the answer time |
19:22.11 | gaetronik | i use sip to cellphone gateway |
19:22.16 | ManxPower | seanbright: but that's not what he said. 8-) |
19:22.32 | gaetronik | i want to limit call time to cell phone to 3 minutes |
19:22.36 | seanbright | it is what he said... just not how he said it :) |
19:22.37 | ManxPower | "cellphone gateway" changes everything |
19:22.43 | gaetronik | yes |
19:22.59 | seanbright | what about the S option? |
19:23.20 | ManxPower | cell phone gateways typically answer incoming SIP calls as soon as they arrive. You can see this in the CLI of Asterisk |
19:23.37 | gaetronik | so i'm screwed? |
19:23.54 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
19:24.29 | ManxPower | gaetronik: unless you can make the SIP gateway stop answering until the outbound leg answers |
19:24.49 | *** join/#asterisk hi365_m (n=hi365@213.151.62.113) |
19:25.14 | gaetronik | ok |
19:25.23 | gaetronik | i don't know if it's possible |
19:26.10 | gaetronik | planet things |
19:27.29 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
19:29.38 | *** join/#asterisk xacatecas (n=jkroon@41.25.174.193) |
19:29.55 | Alan_Hicks | Thanks guys. I don't know how that happened, but I've got it fixed now. |
19:30.06 | Alan_Hicks | I had to add a second alertInfo section to sip.cfg. |
19:30.12 | xacatecas | hi all, any ideas what could be wrong if the Monitor() application can record the individual streams perfectly, but MixMonitor() ends up with a totally garbled audio stream? |
19:30.16 | Alan_Hicks | <alertInfo voIpProt.SIP.alertInfo.2.value="Auto Answer" voIpProt.SIP.al |
19:30.17 | Alan_Hicks | ertInfo.2.class="3"/> |
19:32.09 | [TK]D-Fender | Alan_Hicks: Don't, you'll kill the first. |
19:32.17 | [TK]D-Fender | Alan_Hicks: you need to do both in the same tag |
19:33.07 | Alan_Hicks | OH! |
19:33.08 | Alan_Hicks | Thanks. |
19:33.26 | Alan_Hicks | I was bound to figure that out sooner or later. :-) |
19:36.50 | ManxPower | xacatecas: mixmonitor uses sox, IIRC to mix the audio..... |
19:37.02 | ManxPower | maybe your sox is bad? |
19:37.27 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
19:39.33 | *** join/#asterisk Meaty (n=patate@office.abi.ca) |
19:41.04 | gaetronik | ManxPower, fxo answer might be the setting? |
19:43.13 | ManxPower | gaetronik: Everything I know on the subject I have already said to you. |
19:45.01 | Alan_Hicks | Odd, it works fine both ways. |
19:45.19 | Alan_Hicks | I'm sure the single <alertInfo> line is correct though, so I'm using that. |
19:45.35 | gaetronik | thanks ManxPower |
19:45.41 | gaetronik | the better way is trying |
19:49.28 | *** join/#asterisk Great_Anta_Baka (i=c419fff6@gateway/web/ajax/mibbit.com/x-fff5b2528c0e7f3f) |
19:53.59 | Meaty | Hi Everybody. Hi have trouble with t38 configuration. I place t38pt_udptl=yes in general config section of sip.conf, and t38 is active for my sip device X, even if i place t38pt_udptl=no in my X config section. |
19:54.05 | Meaty | How its possible ? |
19:54.24 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
19:56.10 | linuxer_igor | please anybody help me! (DAHDI) |
19:56.24 | linuxer_igor | PRI span 1/0: Provisioned, Down, Active |
19:56.36 | linuxer_igor | don't work |
19:56.59 | linuxer_igor | today upgrade my asterisk 1.6.0rc4 + DAHDI |
19:57.21 | linuxer_igor | and dont register channels in my E1 |
19:57.31 | ManxPower | linuxer_igor: you understand this is not even release code, right? |
19:57.49 | [TK]D-Fender | RC! |
19:57.56 | linuxer_igor | yep |
19:58.01 | linuxer_igor | I know |
19:58.03 | *** join/#asterisk dieno (i=771e6e9b@gateway/web/ajax/mibbit.com/x-15a3356c7dba8d3a) |
19:58.11 | ManxPower | linuxer_igor: then your best bet is #asterisk-dev |
19:58.17 | linuxer_igor | but I need use this |
19:58.23 | *** join/#asterisk Phree_Beer (n=chatzill@fctnnbsc27w-142166248246.pppoe-dynamic.nb.aliant.net) |
19:58.43 | dieno | can any one tell me why my billsec goes 0 when i reset CDR from macro |
19:58.48 | ManxPower | linuxer_igor: most of us here run released codes. |
19:59.04 | linuxer_igor | they see mi "bugs.digium.com" |
19:59.09 | ManxPower | development code should stay on #asterisk-dev |
19:59.21 | ManxPower | linuxer_igor: then that is where you must go. |
19:59.58 | ManxPower | I suspect not a single person on this channel uses 1.6 prelease and DAHDI prerelease |
20:00.16 | *** part/#asterisk geoff2010 (n=geoffrey@75.150.14.50) |
20:00.24 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
20:00.32 | [TK]D-Fender | I suspect there is a single person and they are feeling very lonely. Won't yous tay a while? ;) |
20:00.46 | x86 | DAHDI? |
20:01.21 | ManxPower | x86: the new name for Zaptel (zaptel is (tm) someone not-digium). There have been several announcements on the mailing lists. |
20:03.16 | x86 | ah, ok |
20:03.23 | x86 | missed that one somehow |
20:07.13 | ManxPower | dieno: does it not do that when you reset CDR not in a macro? |
20:07.21 | *** part/#asterisk Alan_Hicks (n=alan@cardinal.lizella.net) |
20:09.10 | *** join/#asterisk talntid (n=t@66.208.251.170) |
20:09.13 | talntid | Boo |
20:09.45 | talntid | Anyone wanna buy a Rhino R1T1-EC card for $550 shipped? A few months old. No longer need it. bought from voipsupply. |
20:09.56 | seanbright | i'll give you $55 |
20:10.20 | russellb | I'll give you - $1000 |
20:10.20 | mvanbaak | ManxPower: |
20:10.20 | x86 | hahahaha nice, asterisk-users mailing list has an archive from 2016 ;) |
20:10.22 | mvanbaak | Asterisk SVN-branch-1.6.0-r139471-/trunk built by root @ threed002 on a i686 running Linux on 2008-08-22 17:28:21 UTC |
20:10.24 | x86 | http://lists.digium.com/pipermail/asterisk-users/2016-May/thread.html |
20:10.35 | *** join/#asterisk reno_rr (n=reno@maillrar.vestcom.com) |
20:11.06 | talntid | heh ; |
20:11.06 | talntid | ) |
20:11.11 | ManxPower | x86: You should get out more. It's had than for at least a yeart |
20:11.13 | seanbright | talntid: you drive a hard bargin... $60... not a penny more |
20:11.41 | ManxPower | Mr Randal Law had his date wrong |
20:11.53 | seanbright | Mr. Edward Nigma |
20:11.58 | seanbright | anyone? anyone? |
20:12.22 | gaetronik | i'm quite lost with bridgng and so on |
20:12.53 | ManxPower | mvanbaak: Thanks for volunteering to help linuxer_igore, but he's already gone. |
20:12.56 | gaetronik | when i make a call using a cellphone gateway, the bridging log message appaears in asterisk only when i hangup |
20:17.31 | x86 | ManxPower: I've not really played with 1.6 yet... |
20:17.43 | x86 | ManxPower: I did, however, stay at a holliday inn express last night |
20:17.45 | x86 | grins evilly |
20:18.01 | x86 | ManxPower: I just sub'd to 3 of the asterisk mailing lists |
20:18.11 | M1s3ry | ManxPower, it's odd that linuxer_igore was in here asking the same thing that he asked in #asterisknow, and I sent him several messages on... |
20:18.49 | x86 | hmm |
20:19.19 | x86 | I can see that asterisk-commits was not a good idea to subscribe to... been sub'd maybe 5 minutes now and I've got 6 emails already from it |
20:20.17 | gaetronik | then answer of a sip gateway is the time of the bridging, right? |
20:20.23 | seanbright | x86: you subscribed in the middle of an RC release |
20:20.33 | seanbright | x86: though it is pretty chatty in general |
20:21.00 | x86 | seanbright: yeah I seen them making the 1.4.22-RC tag |
20:21.21 | x86 | hmm |
20:21.59 | bipolar | Maybe someone here can help me.... I have my Asterisk box behind NAT, and I want to connect to my sipphone account to use it as a SIP trunk. It works when the box has a public IP, but not when behind nat. What do I need to forward for this to work? |
20:22.04 | x86 | I seem to remember someone saying 1.2.x is no longer supported, yet there are (2) 1.2.x releases on the announcements on the front page of asterisk.org... are they just bug fixes / security updates? |
20:22.14 | seanbright | ~sipnat |
20:22.15 | jbot | [~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 |
20:22.20 | seanbright | bipolar: ^^^ |
20:22.46 | seanbright | x86: 1.2.29? |
20:22.51 | seanbright | and .30? |
20:23.02 | bipolar | seanbright: thanks. I'll look at those pages. Everything I found so far has assumed connecting sip clients to asterisk behind nat, not the other way around.... :P |
20:23.09 | x86 | seanbright: *nod* |
20:23.16 | seanbright | yes, those were security releases |
20:23.31 | x86 | also, why is there no libpri-announce and zaptel-announce mailing lists? |
20:23.43 | seanbright | those get announced on the asterisk lists |
20:23.47 | x86 | ah ok |
20:23.51 | x86 | as does dahdi? |
20:23.59 | seanbright | yep |
20:24.03 | x86 | cool |
20:24.21 | x86 | when will 1.6 be ready for prime time? :) |
20:24.41 | x86 | russell was linking me on some very cool stuff he was planning for 1.6 a long time ago |
20:25.13 | x86 | like the ability to interrupt on a monitor session (via ChanSpy and/or ZapBarge) |
20:25.30 | seanbright | x86: that is already in trunk |
20:25.34 | gaetronik | no one to help me understand bridge logic |
20:25.39 | x86 | I'd really like that functionality with 1.4, as we're a call center |
20:25.41 | seanbright | x86: ChanSpy barge mode |
20:25.50 | x86 | that's probably the most requested feature by my managers |
20:25.59 | x86 | seanbright: will it get back-ported to 1.4? |
20:26.07 | seanbright | x86: no sir, not by the project anyway |
20:26.22 | x86 | :( |
20:26.26 | seanbright | x86: shouldn't be *that* hard though. 1.4 had audiohooks backported to it and chanspy uses those. |
20:26.39 | seanbright | x86: for the right price, i could do it. |
20:26.39 | seanbright | ;) |
20:26.47 | x86 | I'm a far cry from a C developer ;) |
20:26.51 | *** join/#asterisk kotique (n=picachu@host-static-89-41-72-147.moldtelecom.md) |
20:26.53 | kotique | yo. |
20:27.05 | x86 | nah, we've waiting this long, we can continue to wait until 1.6 is stable enough |
20:27.07 | kotique | Dial("SIP/11101-b5b04330", "IAX2/11100/${DID}||tr") - bad :/ |
20:27.15 | kotique | who's familiar with dialparties.cgi ? |
20:27.21 | x86 | wtf is that? |
20:27.28 | seanbright | console output |
20:27.29 | x86 | cgi? you mean agi? |
20:27.35 | x86 | seanbright: no kidding ;) |
20:27.40 | seanbright | well you asked. |
20:27.43 | x86 | seanbright: i was referring to the specific cgi ;) |
20:27.48 | kotique | what means dial= config option in iax2.conf / |
20:27.57 | seanbright | dialparties = AMP/FreePBX |
20:27.59 | kotique | searching all internets - no luck |
20:28.01 | seanbright | iirc |
20:28.11 | x86 | you searched ALL the internets? |
20:28.14 | kotique | yep |
20:28.17 | kotique | ALL |
20:28.20 | x86 | oh man, that's impressive |
20:28.27 | x86 | I just a whole fleshlight too |
20:28.38 | Kobaz | kotique: did you try teh interwebs too |
20:29.00 | seanbright | kotique: you're using freepbx? |
20:29.02 | x86 | seanbright: do you forsee a timeline as to when 1.6 will be production ready? |
20:29.18 | seanbright | x86: i'll probably run it in production right after release |
20:29.27 | Kobaz | seanbright: haha |
20:29.38 | x86 | seanbright: we've got 7 offices, each with a single Asterisk box and anywhere from 2-6 T1 ports per server |
20:29.39 | seanbright | isn't joking |
20:29.53 | xacatecas | ManxPower, so it records the two streams separately and then uses sox to mix afterwards? |
20:29.53 | x86 | seanbright: when will it be released? |
20:30.06 | seanbright | x86: once the RCs are fleshed out and such |
20:30.14 | seanbright | x86: another RC is being rolled right now |
20:30.38 | x86 | so the answer is, no there is not an approximate timeline? :) |
20:30.39 | Kobaz | is there an overview somewhere of all the new cool stuff in 1.6 |
20:31.01 | seanbright | x86: i'd say "soon" |
20:31.02 | kotique | hey guys. what does "dial" option mean ? |
20:31.09 | x86 | Kobaz: I know of at least a couple new badass features that have me salivating already |
20:31.11 | kotique | in iax2.conf 's peer configuration |
20:31.25 | Kobaz | x86: like what? |
20:31.29 | seanbright | x86: but i don't make those decisions, so... |
20:31.33 | seanbright | shrugs |
20:31.40 | x86 | Kobaz: like the new ChanSpy barge-in ability :) |
20:31.43 | Kobaz | heh |
20:32.03 | Kobaz | anything else? |
20:32.12 | seanbright | app_fax |
20:32.22 | x86 | there was something else russell was telling me about (or perhaps linked me to on his website)... can't remember what specifically, but I was all I CAN HAZ? |
20:32.43 | kotique | so russel is that development guy? |
20:32.56 | x86 | meh, I've had such bad luck with faxing and asterisk, I just outsourced my faxing to hylafax and an external 56k modem ;) |
20:33.05 | Kobaz | hehe |
20:33.19 | kotique | HAI GAIS. CAN I HAS VARIABLE dial in IAX conf explain PLIZZ ??? |
20:33.20 | seanbright | all the new stuff is in the UPGRADE.txt file |
20:33.34 | kotique | ok |
20:33.43 | seanbright | http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?revision=137631&view=markup |
20:33.44 | x86 | i've tried spandsp (app_rxfax, app_txfax), as well as iaxmodem + hylafax, and nothing was stable at all... |
20:34.01 | x86 | so I just put a modem on another box and have it setup with hylafax, works like a champ |
20:34.06 | kotique | i'm using iaxmodem + hyla, working great. |
20:34.27 | x86 | kotique: yeah i had it working, but it was far from reliable |
20:34.38 | x86 | I got a good 60% reliability out of it |
20:34.50 | kotique | well, go g711 and be lucky |
20:34.59 | seanbright | and CHANGES -> http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?revision=134126&view=markup |
20:35.03 | x86 | err, I would never even try anything else but g711 ;) |
20:35.13 | x86 | and actually, the call came in on a zap channel anyway ;) |
20:35.19 | kotique | why not ? t.38 pass-through |
20:35.23 | kotique | or even termination |
20:35.35 | x86 | telco --> POTS --> channel bank --> asterisk == iaxmodem fail |
20:35.59 | x86 | t.38 only applies when you're using IP somewhere, no? |
20:36.03 | kotique | I still don't get why people dont' use email |
20:36.06 | x86 | (which I am not) |
20:36.19 | kotique | yea, t.38 is ip thingie |
20:36.38 | x86 | yeah I had low reliability with a POTS line on a local zap channel |
20:36.44 | x86 | no IP involved at all |
20:37.08 | *** join/#asterisk StooJ (n=stooj@johnston37.plus.com) |
20:37.10 | x86 | of course we get about 50 faxes a day, I'd venture to guess |
20:41.04 | x86 | out of the 50, we'd have about 35 or so that would come through ok... the rest would be trauncated, garbled, or just complete failure negotiating the fax handshake |
20:41.28 | x86 | we've got a 100% success rate with hylafax and an external modem though |
20:43.02 | wiscados | QQ. can use asterisk to be a sip provider |
20:43.19 | *** join/#asterisk Ditegen (n=ditegen@mafet.ru) |
20:43.48 | drmessano | Asterisk provides SIP |
20:44.21 | Ditegen | yeah? |
20:45.35 | drmessano | Well, it is a server that makes SIP connections available client.. Like Apache makes HTTP connections available to clients, so yet. |
20:45.39 | drmessano | yes* |
20:46.22 | wiscados | ok, so I can set up a asterisk server at example.com and procide you@example.com and me@example.com |
20:53.14 | Ditegen | how about redundancy? have you been lucky with it? |
20:55.20 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
20:57.35 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
21:05.30 | *** join/#asterisk Paige (n=Paige@208.89.241.31) |
21:05.51 | jaytee | quittin time, back later |
21:05.51 | Paige | hi. which addons will work with 1.6.0-rc4? |
21:07.35 | x86 | ugh, DAHDI channels have to be invoked by DAHDI/channum, instead of Zap/channum... that's going to mess up my flash operator panel, and most of my AMI stuffs :( |
21:07.46 | [TK]D-Fender | Paige: Probably the one listed right along side it in the topic... |
21:10.09 | AndyML | [TK]D-Fender: did you happen to see my last message a couple hours ago? |
21:10.29 | [TK]D-Fender | AndyML: not that I recall |
21:11.58 | x86 | ugh, and Macro() is deprecated now? |
21:12.25 | mchou | what?? |
21:14.48 | Knightfal | <PROTECTED> |
21:16.26 | WimpMan | Well, never having used Macros befor I had the hope, they wouldn't change the context written to the cdrs, but as they do, I don't see any use anyway. |
21:16.39 | gaetronik | x86, are you kidding? |
21:18.07 | Kobaz | x86: what replaces Macro? |
21:18.10 | Paige | [TK]D-Fender, http://pastebin.com/d608752 |
21:18.22 | Paige | i get errors trying to build it |
21:18.35 | gaetronik | yes since it was very usefull for calling various Gateway |
21:18.43 | gaetronik | and do thing between it |
21:18.52 | WimpMan | Nothing. You have to Goto or Gosub. |
21:18.59 | Kobaz | that's kinda ghetto |
21:19.02 | *** part/#asterisk JenniferAkemi (n=akemi@206-248-161-97.dsl.teksavvy.com) |
21:19.03 | tzafrir_laptop | x86, what's the big deal? rewrite some config files |
21:19.35 | WimpMan | Hmm. Actually there IS a point in Macro: Passing arguments. |
21:20.02 | Kobaz | yeah, then you have to ghetto it up with using delimiters in the exten and then using CUT |
21:20.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
21:21.02 | Kobaz | GoSub(mysub,arg1^arg2^arg3^arg4,1) |
21:21.16 | Kobaz | or whatever |
21:21.18 | WimpMan | Or Set() before. That would be more elegant anyway, as the parameters have sensible variable names then, but that makes two lines (Set, Goto) instead of one. |
21:21.22 | Kobaz | i guess you could set a channel variable |
21:21.28 | Kobaz | yeah |
21:21.53 | *** join/#asterisk Superbaloo (i=FN@energeek.net) |
21:21.58 | Superbaloo | hi there |
21:22.07 | Kobaz | where can we petition to not have macro removed |
21:22.15 | x86 | Kobaz: GoSub / Return |
21:23.10 | x86 | Kobaz: it wont be removed yet, it's just deprecated |
21:23.20 | x86 | Kobaz: due to "performance reasons" |
21:23.26 | x86 | <PROTECTED> |
21:23.26 | Kobaz | deprecated = removed in the future |
21:23.31 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
21:23.43 | x86 | Kobaz: it explicitly states it will NOT be removed, but it is deprecated |
21:23.53 | Superbaloo | howto bind sip on two interfaces ? |
21:24.02 | Kobaz | why deprecate something if you're not planning on removing it |
21:24.25 | Superbaloo | what is the syntax of the sip.conf ? |
21:24.25 | WimpMan | Superbaloo: By not restrictiong it to only one? |
21:24.32 | Kobaz | i would think that there's a high liklyness that it will go away at some point if it's deprecated |
21:24.42 | WimpMan | Kobaz: Indeed. That doesn't make much sense to me, either. |
21:26.11 | Superbaloo | WimpMan, hum that's an interesting proposition but the 0.0.0.0 option is not really one for me (i'm running asterisk in vserver) and i need to have sip running on two interface (wifi and network) |
21:26.29 | x86 | oh man |
21:26.50 | x86 | asterisk 1.6 supports using X11 as a local video source for video over chan_oss |
21:26.56 | WimpMan | Superbaloo: And why would you need less than all then? |
21:27.31 | Superbaloo | WimpMan, cause on vserver it bind on all of my ip (including ip of others vservers) |
21:28.28 | WimpMan | Isn't vserver supposed to get that straight? |
21:28.42 | Superbaloo | it should yes |
21:29.12 | Superbaloo | ok so just forget that ... |
21:29.21 | Superbaloo | i would do it with iptable |
21:29.44 | WimpMan | Or fix vserver |
21:29.50 | Superbaloo | thanks anyway |
21:29.56 | Superbaloo | WimpMan, hum |
21:30.04 | Superbaloo | iptables would be fine :D |
21:31.25 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5 (2008/09/08), 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #a |
21:31.46 | Qwell | russellb: pwnt |
21:31.47 | WimpMan | was under the impression that this used to worl with vserver. But it's quite some time I looked into it. |
21:32.33 | russellb | \topic Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5, 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: irc://chat.freenode.net/#asterisknow irc://chat.freenode.net/#asterisk-gui irc://chat.freenode.net/#switchvox |
21:32.53 | russellb | major ownage |
21:33.04 | Qwell | O.o |
21:33.06 | russellb | lol |
21:33.25 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-rc5, 1.4.22-rc4 (2008/09/08), *-Addons 1.6.0-rc1 (2008/09/03), 1.4.7 (2008/06/04), dahdi-linux-complete 2.0.0-rc4+2.0.0-rc2 (2008/09/08), Zaptel 1.4.12 (2008/09/03), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-commits #asterisk-bugs #asterisk-dev. AstriCon! #astricon |
21:34.05 | Paige | anyone have any ideas why it enild? |
21:34.32 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:34.45 | *** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak) |
21:34.59 | WimpMan | AMI bridge sound cool. |
21:40.04 | *** join/#asterisk samad (n=samad@116.71.178.25) |
21:40.16 | samad | hi everyone |
21:40.36 | samad | anybody knows about nortal programming |
21:40.38 | samad | ? |
21:42.32 | [TK]D-Fender | samad: try #nortel , you're in the wrong channel |
21:43.37 | samad | ok |
21:44.26 | samad | is there any other room you can suggest |
21:46.30 | samad | ? |
21:47.28 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
21:47.36 | Titanous | is there any way to do a SIPAddHeader with a manager originate? |
21:48.55 | [TK]D-Fender | Titanous: Use a Local channel as your "Channel:" |
21:50.12 | *** join/#asterisk EI5GTB-macbook (n=EI5GTB@78.16.75.126) |
21:50.13 | Titanous | [TK]D-Fender: I'm not sure I follow, I am using a local "Channel:" |
21:50.27 | Titanous | I'd like to add a SIP header to the call to that channel |
21:50.59 | [TK]D-Fender | Titanous: Show us a sample of your "channel" line |
21:51.16 | Titanous | Channel: SIP/200 |
21:51.20 | WimpMan | Titanous: You want to put it in your dialplan and call that via local/exten. |
21:51.25 | [TK]D-Fender | Titanous: that is NOT a Local channel |
21:51.37 | [TK]D-Fender | Titanous: Go read up and Asterisk's channel types |
21:51.49 | Titanous | ah, ok, now I understand, thanks |
21:54.27 | outtolunc | he should also be able to use Variable: SIPADDHEADER=yadda |
21:55.56 | russellb | please don't do that ... |
21:56.15 | russellb | the name of the vars that store the headers is an internal implementation detail |
21:56.24 | russellb | however, you _should_ be able to use the dialplan function that way. |
21:56.56 | russellb | by adding a line like ... Set: SIP_HEADER(X-My-Header)=foo |
21:57.25 | outtolunc | he was wanting to do it from an originate |
21:57.30 | outtolunc | anyways |
21:57.38 | russellb | ah, same thing, in a Variable: header or whatever |
21:57.53 | lmadsen | the point is to use the dialplan function to set the header :) |
21:58.01 | russellb | right. |
21:58.28 | lmadsen | embrace the dialplan function; love the dialplan function |
21:59.24 | russellb | :) |
22:00.35 | [TK]D-Fender | goes to make some more Kool-Aid |
22:00.47 | lmadsen | I prefer Crystal Light |
22:01.02 | outtolunc | i was gonna say, isn't kool-aid deprecated |
22:01.09 | outtolunc | hides |
22:01.15 | lmadsen | heh |
22:01.20 | lmadsen | oh no |
22:01.22 | lmadsen | oh no |
22:01.26 | lmadsen | oh ya! |
22:01.31 | [TK]D-Fender | lmadsen: I prefer the "Presidents's Choice" alternative brand and the 50% it leaves in my bank account :) |
22:01.42 | lmadsen | [TK]D-Fender: I have not seen such a thing :) |
22:01.51 | lmadsen | or I would consider buying it |
22:01.59 | *** join/#asterisk jjshoe_ (n=jjshoe@72.37.252.50) |
22:02.22 | jjshoe_ | what's the best way to test for file existance like a voicemail greeting? readfile? |
22:02.39 | lmadsen | sure |
22:02.50 | outtolunc | stat? |
22:02.57 | [TK]D-Fender | lmadsen: it is 1/2 the price... goes by the title "Cool Delight" |
22:03.00 | russellb | there is a STAT function somewhere ... |
22:03.04 | russellb | might be 1.6 only though |
22:03.08 | lmadsen | I think only 1.6 |
22:03.10 | russellb | ah. |
22:03.13 | [TK]D-Fender | lmadsen: I buy them in droves :) |
22:03.15 | lmadsen | however, I did copy it to 1.4 and it "just worked" |
22:03.21 | russellb | nice. |
22:03.44 | lmadsen | [TK]D-Fender: crystal light certainly is tasty -- I like the idea of a cheaper version |
22:03.49 | outtolunc | STAT() is in 1.4.19 also |
22:04.13 | [TK]D-Fender | lmadsen: Same thing. Look around. We've got it at Maxi, Loblaws, etc.... |
22:04.16 | lmadsen | really? hrmmm... something I was using was 1.6 only... |
22:04.29 | lmadsen | [TK]D-Fender: hrmmm... I tend to shop at Dominion |
22:04.37 | [TK]D-Fender | lmadsen: Avoid aisle kiosk bis... thats where they keep pimping the brand name items |
22:04.56 | [TK]D-Fender | bins* |
22:07.03 | *** join/#asterisk bkruse (n=bkruse@nat/digium/x-e5c46459dd29b9a5) |
22:07.03 | *** mode/#asterisk [+o bkruse] by ChanServ |
22:08.47 | *** join/#asterisk Levonk (n=lk@adsl-76-238-249-188.dsl.lsan03.sbcglobal.net) |
22:09.12 | *** join/#asterisk tuxd00d (n=tuxd00d@128.187.132.25) |
22:11.19 | Yourname | Does anyone know of an online interface much like "ARI from Little John Consulting" that would let users check voicemail/call forward, etc? I can't seem to find the ARI thing either. |
22:16.48 | [TK]D-Fender | Yourname: what "ARI thing"? |
22:17.20 | Yourname | [TK]D-Fender: It's a web interface basically, the only way I know what it is is at the bottom of the page where it sas "ARI From little john consulting", so I thought someone would know |
22:17.26 | tzafrir_laptop | Yourname, the most up-to-date version is probably included in FreePBX |
22:17.50 | Yourname | tzafrir_laptop: Probably mixed in with almost everything in freepbx :S |
22:18.06 | tzafrir_laptop | I don't think that the original is being developed anymore |
22:18.33 | [TK]D-Fender | I see references to it, and links online... |
22:20.04 | outtolunc | he probably got tired of answering the thousands of 'trixbox users' questions and not seeing a damn dime <G> |
22:21.18 | Yourname | lol |
22:21.25 | Yourname | His name was Dan. |
22:21.28 | Yourname | or probably still is. |
22:21.48 | Yourname | Does anyone know of anything else that's like it? |
22:22.43 | *** join/#asterisk AndyML (n=alauppe@pool-71-185-74-86.phlapa.fios.verizon.net) |
22:24.33 | AndyML | I have an asterisk-gui system running in a small call center. I've configured it for dynamic login/logout via the dialplan but watching the Asterisk CLI makes me think I've done something incorrectly... |
22:25.59 | AndyML | examples - http://pastebin.ca/1197591 |
22:26.06 | jameswf-home | ~asterisk-gui |
22:26.07 | jbot | [~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0 |
22:26.38 | jameswf-home | shouldnt that say goto #asterisk-gui |
22:27.17 | AndyML | it shouldn't say goto asterisk-gui because asterisk-gui is part of a standard asterisk distribution. |
22:27.24 | *** join/#asterisk Whisk (n=Whisk@82-44-94-242.cable.ubr04.croy.blueyonder.co.uk) |
22:27.25 | Qwell | No it isn't |
22:27.35 | jameswf-home | ~asterisk-gui is also For support go to #asterisk-gui |
22:27.35 | jbot | jameswf-home: okay |
22:28.35 | AndyML | pretend for a second that I didn't say i was using asterisk-gui. I built most of the logic directly in extensions.conf and queues.conf. if anyone here knows anything about asterisk queue engine, they can probably help me. |
22:28.47 | jameswf-home | ~ask |
22:28.48 | jbot | ask is probably Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
22:29.20 | jameswf-home | pastebin your dial plan and give up the issues |
22:29.52 | AndyML | jameswf-home: see above - I have pastebin'd my issue already. |
22:30.27 | AndyML | i'll dig up the dialplan as you look at that. If you're willing to help, i would appreciate it greatly. |
22:30.51 | jameswf-home | well i cant be a condecending a-hole and read geesh... looking |
22:31.24 | AndyML | i'm confused - have I offended you? |
22:32.22 | jameswf-home | heh no not an easy task..... |
22:32.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
22:32.45 | outtolunc | takes jameswf-home's eggo |
22:32.49 | jameswf-home | my network doesnt like pastebin.ca hmmm |
22:32.59 | jameswf-home | leggo my eggo |
22:33.01 | AndyML | where else would you like me to put it? |
22:33.05 | jjshoe_ | Yourname ari? asterisk remote interface? |
22:33.17 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
22:33.21 | jameswf-home | ~pb |
22:33.22 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding 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://www.rafb.net/paste |
22:33.29 | AndyML | .com - ok. |
22:33.30 | jameswf-home | let me see what loads |
22:33.46 | jameswf-home | .com is good |
22:33.59 | [TK]D-Fender | AndyML>it shouldn't say goto asterisk-gui because asterisk-gui is part of a standard asterisk distribution. <- no such thing as an "Asterisk distribution". |
22:34.25 | AndyML | you're right - i used the wrong noun. "installation" |
22:34.41 | jameswf-home | isnt installation a verb |
22:34.47 | [TK]D-Fender | AndyML: And its not part of any of MY standard * installs |
22:35.02 | [TK]D-Fender | AndyML: 0 for 2, care to make it a nice round 3? |
22:35.25 | AndyML | look - i love community supported software, but does ridicule have to be a part of it? |
22:35.45 | jameswf-home | AndyML: ummm yeah |
22:35.47 | AndyML | nobody likes being made a fool of. |
22:36.13 | jameswf-home | AndyML: its all free without ridicule of newbs we would see no return |
22:36.39 | [TK]D-Fender | AndyML: Believe me, I haven't actually started any kind of ridicule yet. That was a minor jab at best. |
22:36.47 | AndyML | I've been supporting Asterisk professionally for more than 3 years. Can you save it for the newbies at least? |
22:36.53 | jameswf-home | the only place the god complexes get worse is in -dev |
22:37.05 | [TK]D-Fender | AndyML: Now what you seem to show in your PB (what little there is to see), seems to indicate you keep calling a busy device. |
22:37.07 | jjshoe_ | AndyML it's more prevelant in certain channels |
22:37.19 | jjshoe_ | AndyML it's strange to see it so strong on freenode |
22:37.35 | [TK]D-Fender | AndyML: a better PB would have included SIP debug and the call from beginning to end. |
22:37.39 | jameswf-home | heh heh we are 1337 :)) |
22:37.41 | jameswf-home | sorry |
22:37.49 | AndyML | i'm not new to IRC either, or the god complex. I guess you just get sick of it after a while. |
22:38.01 | [TK]D-Fender | jameswf-home is teh shizit y0!!!! r3sepkt! |
22:38.34 | [TK]D-Fender | Best part about my God complex : no peer pressure :) |
22:39.00 | *** join/#asterisk rasterix (n=IceChat7@host86-154-172-124.range86-154.btcentralplus.com) |
22:39.04 | jameswf-home | hmmm thinktel.ca < get caught up at.... gueass thats the great firewall of canada |
22:39.36 | [TK]D-Fender | AndyML: So I've already given you 2 better ideas on providing more complete info to examine your problem... |
22:39.54 | AndyML | I can only process the shame and copy/paste so fast. |
22:40.55 | [TK]D-Fender | AndyML: And you fear you didn't set your dialplan up properly but showed us only the smallest amount of debug info, and not even the dialplan you are doubting. |
22:41.15 | [TK]D-Fender | sits back and sits his post-fondue broth |
22:41.22 | [TK]D-Fender | sips* |
22:41.23 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
22:41.23 | rasterix | ugh dont mention firewalls |
22:41.34 | AndyML | oh, even the great [TK]D-Fender mistypes sometimes. |
22:42.02 | Yourname | jjshoe_: Probably? I'm not sure.. but looks good, and want it lol |
22:42.03 | AndyML | i was pasting it all at pastebin.ca when jameswf-home said he couldn't get to pastebin.ca. |
22:42.21 | [TK]D-Fender | AndyML: I cut my thumb open with a sword by accident this weekend. Excuse enough :) |
22:42.29 | jameswf-home | canada rejected my passport |
22:42.40 | [TK]D-Fender | jameswf-home: TERRIST! |
22:43.10 | jjshoe_ | terrist? |
22:43.11 | jjshoe_ | sounds vicious |
22:44.03 | jameswf-home | heh heh terror www.youtube.com/watch?v=1uwOL4rB-go |
22:44.15 | rasterix | fender only mistypes to make himself appear human |
22:44.15 | rasterix | pfft canada rejected my country |
22:44.15 | rasterix | fender is impersonating a moron that cant type... dont be fooled its a cunning disguise |
22:45.48 | jameswf-home | http://www.100factsabout.com/ircuser/%5BTK%5DD-Fender |
22:47.06 | jameswf-home | heh # Give a man a fish, and you will feed him for a day. Give a man anything that is better than a fish, and Ircuser [TK]D-Fender will beat his ass and take it. |
22:47.40 | jameswf-home | better: # Ircuser [TK]D-Fender can eat a rubix cube and crap it out solved. |
22:47.46 | jameswf-home | neat |
22:48.21 | [TK]D-Fender | jameswf-home: Yeah I saved the link to that page you posted a few weeks ago :) |
22:48.33 | [TK]D-Fender | <- pornographic memory |
22:48.35 | rasterix | what worries me is how he peels all the labels off and sticks them back on b4 he craps it >< |
22:48.53 | scooby2 | Any idea why idle agents would not ring when there are calls waiting in the queue? Ring-all strategy. Or is this to force you to ditch agents with 1.4.x? |
22:49.28 | rasterix | scooby it cud be bcoz u have agents that do not appear busy when they are |
22:49.47 | AndyML | http://pastebin.com/m51964c1e |
22:50.05 | AndyML | excerpts from extensions.conf, queues.conf, the CLI and a SIP debug. |
22:50.43 | scooby2 | In what sense? We have zero wrapuptime. Asterisk will usually show a message if they turned DnD on |
22:50.53 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279394397.dsl.bell.ca) |
22:51.24 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
22:52.14 | rasterix | example a sip phone with multiple lines... the agent is on one but asterisk passes the call to the second line |
22:52.14 | rasterix | but that could have been down to logging devices rather than agents on to the queues |
22:52.47 | Yourname | jjshoe_: Any help on that? :) |
22:52.53 | [TK]D-Fender | AndyML: So far looks like a phone on DND... |
22:53.10 | scooby2 | rasterix: if the agent logs out and logs back in then they get a call. It is strange. |
22:53.12 | [TK]D-Fender | AndyML: And your extensions.conf does not play into this |
22:53.58 | AndyML | let me log that phone out of the queue. |
22:54.18 | rasterix | sounded similar to what u have |
22:54.18 | rasterix | what phones are you using scooby? |
22:54.41 | AndyML | actually - that phone is just taking a call. they're an active member of the queue |
22:55.01 | scooby2 | rasterix: SPA941-NA |
22:55.17 | scooby2 | call waiting turned off |
22:55.36 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
22:56.11 | AndyML | call waiting is off because the queue engine would send it more than one call at a time if it were on |
22:56.38 | rasterix | oooh |
22:56.38 | rasterix | linksys |
22:56.38 | rasterix | sounds even more familiar |
22:56.38 | rasterix | theres a setting |
22:56.38 | rasterix | i had to set on our spa942 |
22:56.38 | rasterix | to get queues working right |
22:56.58 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
22:57.45 | AndyML | these are polycom phones |
22:57.49 | [TK]D-Fender | AndyML: What ver of *? |
22:58.09 | AndyML | 1.4.19.1, as per pastebin entry ( <grin> ) |
22:58.25 | [TK]D-Fender | AndyML: Go follow UPGRADE.TXT and it won't send calls to busy members.. |
22:58.58 | rasterix | scooby u will prolly find it if u google spa942 queues and asterisk |
22:58.58 | rasterix | i forget what it was but basically there is a setting to disable multiple lines |
22:58.58 | rasterix | andy im talking to scooby |
22:59.29 | scooby2 | ok |
22:59.52 | AndyML | rasterix: its ok - i got that. |
23:00.36 | AndyML | I'll look at UPGRADE.TXT again. |
23:02.22 | jksM | anyone know a simple call generator for SIP/RTP that can be used to stress test asterisk? (preferable one that is free or cheap) |
23:02.43 | rasterix | its definitely solvable scooby |
23:02.43 | rasterix | i fixed it |
23:02.43 | rasterix | and im an idiot |
23:02.43 | rasterix | ask anyone here |
23:02.52 | scooby2 | lol |
23:02.56 | [TK]D-Fender | rasterix: with m4d l33t sk1llz like this : http://www.youtube.com/watch?v=eJEmtLxkEoI @ 4:06 |
23:05.22 | rasterix | scooby if u cant find it |
23:05.56 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
23:06.09 | rasterix | msg me |
23:06.10 | rasterix | ill log into work i have the solution logged in our "bible" somewhere |
23:06.24 | scooby2 | ok thanks. not finding it so far |
23:06.39 | scooby2 | but i'm looking still |
23:06.59 | rasterix | if i log into work ill get kicked off of here |
23:07.19 | rasterix | dumb vpn |
23:08.02 | scooby2 | lol |
23:10.03 | scooby2 | the only good thing about these phones is that they were cheap. Other than that, I wish we would have gotten Polycom |
23:10.03 | rasterix | no scooby |
23:10.20 | rasterix | they r great phones |
23:10.20 | rasterix | dont get mad over this |
23:11.07 | [TK]D-Fender | SPA's ar a decent runner up, but indeed behind in the race... |
23:11.08 | scooby2 | is it lines or queues that I should be searching for? |
23:11.46 | rasterix | disable call waiting |
23:12.09 | [TK]D-Fender | Fix your SIP configs <- |
23:12.33 | scooby2 | [TK]D-Fender: to send it only one call at a time? |
23:13.16 | rasterix | rawr i cant believe i cant remember this |
23:13.16 | [TK]D-Fender | scooby2: Set them up properly and you can leve CW on and * queues won't send calls to busy agents |
23:13.20 | rasterix | i might log into work |
23:13.39 | scooby2 | [TK]D-Fender: that would be a nice feature |
23:14.11 | [TK]D-Fender | scooby2: UPGRADE.TXT <- |
23:15.15 | rasterix | ah ok my fix is for 1.2 |
23:15.15 | rasterix | maybe there is a better solution for 1.4? |
23:15.51 | scooby2 | i just upgraded from 1.2.x to 1.4.21.2 last Wednesday |
23:17.27 | rasterix | and u had the problem since u upgraded? |
23:17.32 | scooby2 | yeah |
23:17.45 | scooby2 | in 1.2 we just turned off call waiting on the phones |
23:17.49 | scooby2 | hack but it worked |
23:18.40 | rasterix | thats what we did |
23:18.40 | rasterix | and no problems when we upgraded to 1.4 (about 2 months ago) |
23:21.59 | rasterix | scooby have u upgraded all ur spas to the latest firmware? |
23:21.59 | rasterix | coz id hate to struggle on this and find ur running 1980s firmware :P |
23:22.18 | AndyML | rasterix: as you've seen - queues in 1.4 isn't totally awesome out of the box. |
23:23.10 | AndyML | err scooby2 ... looks like rasterix has been able to make it work |
23:23.16 | scooby2 | not the latest latest but close. All the latest firmware does is allow four lines instead of the default two. Previously they charged for the upgrade. |
23:23.18 | rasterix | neither was queues in 1.2 |
23:23.18 | rasterix | i only have a month or 2 on 1.4 |
23:24.32 | AndyML | well i pretty much hate queues... |
23:25.29 | rasterix | scooby on the 942 the same time as the upgrade from two to four lines they added a new option for queues |
23:25.29 | rasterix | to fix problems like this |
23:25.29 | rasterix | ur really going to make me log into work |
23:25.30 | rasterix | *Sigh* |
23:25.30 | rasterix | ill be back |
23:25.30 | rasterix | bye |
23:26.28 | scooby2 | thanks |
23:26.33 | AndyML | thank you rasterix ... |
23:31.26 | *** join/#asterisk Alton2 (n=alton@69.45.115.198) |
23:37.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
23:38.58 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
23:42.50 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
23:52.09 | *** join/#asterisk coppice (n=chatzill@177.162.17.210.dyn.pacific.net.hk) |
23:53.30 | *** join/#asterisk mihinomenest (i=Mgah338m@66.255.220.17) |
23:56.36 | *** join/#asterisk hijacked (n=argh@c-71-205-135-177.hsd1.mi.comcast.net) |
23:58.32 | *** join/#asterisk Levonk (n=lk@75.62.142.43) |
23:58.54 | rue_mohr | ok |
23:59.24 | *** part/#asterisk korihor (n=korihor@201.211.168.130) |