| 00:00.17 | Katty | i have pictures. |
| 00:00.27 | WimpMan | Never tried parmesan with potatoes. But for mashed ones I NEED fried onions. |
| 00:01.07 | rob0 | mixes sour cream in 'shtaters |
| 00:01.10 | SwK | I just prefer brown gravy on mine... but fried onions go good on pretty much anything |
| 00:01.52 | Carlos_PHX | This is worst than going shopping when you're hungry. |
| 00:02.08 | Assimilate | lol |
| 00:02.40 | drmessano | Fried onions in mashed potatoes? |
| 00:02.41 | drmessano | How.... |
| 00:02.45 | drmessano | How can I ever thank you |
| 00:03.31 | Katty | it's going to get worse. i'm about to blog dinner. |
| 00:04.24 | Assimilate | Ok I'm stuck. I can make outbound calls to my sip provider but inbound I get the invite then I send a proxy auth then they ack then I send a notify with status of terminated and it goes to the sip providers voicemail. I am moving from 1.2 to 1.4 and its got me stumped. What am I missing? |
| 00:04.34 | Tuxguy | hardwire: so if i create a .call file in the outgoing folder, it will automatically know? |
| 00:04.56 | hardwire | like florz just said, it's every second |
| 00:04.59 | hardwire | in 1.4+ |
| 00:05.04 | hardwire | I was teh wrong |
| 00:05.43 | Tuxguy | oh ok |
| 00:06.05 | WimpMan | Tuxguy: DO NOT create them in the outgoing directory. Always mv them there. |
| 00:06.10 | Tuxguy | ok |
| 00:06.26 | Tuxguy | Create them in /tmp then move them... what is the reason? |
| 00:06.36 | WimpMan | Otherwise you risk * reading half a file. |
| 00:06.49 | Katty | Dinner: http://angela.sleekgeek.org/2008/11/11/pork-chops-yum-yum-crockpot-style/ with http://angela.sleekgeek.org/2008/11/11/homemade-mashed-potatoes-slacker-style/ |
| 00:06.54 | Tuxguy | Ah.. makes sense |
| 00:06.58 | WimpMan | Make sure it's on the same fs. |
| 00:07.13 | Tuxguy | Can it be a mounted network drive? |
| 00:08.15 | WimpMan | If both the temp and the outgoing directory are on the same mount that should be ok. |
| 00:09.18 | *** part/#asterisk jaytee (i=d8cff501@gateway/web/ajax/mibbit.com/x-292b263883f4d08e) |
| 00:10.12 | Carlos_PHX | Katty: If you blog dinner while I'm starving, I'll blog the AZ weather while you're in your snow/sleet/stuff. |
| 00:13.17 | *** join/#asterisk italorossi (n=italoros@201.76.152.226) |
| 00:14.14 | *** part/#asterisk zecrazytux (n=zecrazyt@boulz.org) |
| 00:17.51 | drmessano | I'm gonna blog.... |
| 00:17.58 | drmessano | Nothing.. I never blog |
| 00:20.26 | *** join/#asterisk ManxPower (n=manxpowe@241.sub-75-251-120.myvzw.com) |
| 00:20.50 | *** part/#asterisk korihor (n=korihor@201.210.239.172) |
| 00:23.00 | *** join/#asterisk kornelak (n=karl@199.33.79.4) |
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| 00:32.07 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:35.48 | kornelak | Hello, good evening! I hope you do not mind me barging in; does anyone know who I could talk to with a DUNDi question? |
| 00:36.36 | WimpMan | Try asking. |
| 00:36.41 | kornelak | OK! |
| 00:37.25 | kornelak | I'm wondering if there have been reports of DUNDi traffic not properly going from a 32-bit machine to a 64-bit machine. |
| 00:37.44 | drmessano | ..... |
| 00:37.51 | drmessano | nope |
| 00:38.08 | kornelak | For example, if a 64-bit machine sends a query to a 32-bit machine, and the 32-bit machine has a matching ext., 32-bit mach. response, and connection is made. |
| 00:38.25 | kornelak | But the reverse, 32-bit machine querying a 64-bit machine, doesn't work. |
| 00:38.37 | drmessano | Check your config |
| 00:38.46 | kornelak | That was my first guess. |
| 00:38.55 | kornelak | dundi show peers is OK. |
| 00:39.14 | kornelak | And for testing purposes, all machines have the same key, so inkey and outkey are the same on all configs. |
| 00:39.36 | WimpMan | Have you tried manual queries? |
| 00:39.59 | kornelak | WimpMan: Do you mean the dundi query command, or the dundi lookup command? |
| 00:41.00 | WimpMan | lookup |
| 00:41.30 | kornelak | OK |
| 00:41.36 | Katty | lukaciapa$$w0rd |
| 00:42.04 | Katty | oh. |
| 00:42.04 | kornelak | If I run `dundi lookup ext_on_32bit_mach@mindspeed` from the 64-bit machine, I get a result. |
| 00:42.08 | Katty | good thing that's not natted. |
| 00:42.16 | kornelak | (mindspeed is the DUNDi context name I'm using internally) |
| 00:42.45 | kornelak | If I run `dundi lookup ext_on_64bit_mach@mindspeed` from the 32-bit machine, I get "DUNDi Lookup returned no results." |
| 00:44.25 | kornelak | Also weird: If I query the 32-bit machine's entity ID from the 64-bit machine, I get a result (name, address, etc.). If I try to query the 64-bit machine from the 32-bit machine, I get "DUNDi Query EID returned no results.". |
| 00:46.35 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 00:46.50 | pcrane | hi guys |
| 00:46.53 | pcrane | got a problem |
| 00:47.21 | pcrane | is there a way to let agents in a queue what the next call is without presenting the call to them? |
| 00:47.42 | pcrane | (i.e. I've got the call limit set to 1, but they need to know what the next call is in the queue) |
| 00:48.06 | Katty | Qwell: wow is down :< |
| 00:55.01 | drmessano | ZOMG |
| 00:55.14 | drmessano | This could be the end of the world!...... of warcraft |
| 00:55.30 | drmessano | I loved that southpark episode |
| 00:57.24 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 00:59.42 | Katty | cries |
| 00:59.54 | Katty | pouts |
| 01:00.01 | seanbright | because of WoW? |
| 01:00.56 | Qwell | seanbright: of course! |
| 01:01.03 | Qwell | Katty: still down? |
| 01:01.24 | Katty | Qwell: )_= |
| 01:01.58 | seanbright | oh jeebus |
| 01:04.12 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:05.13 | *** join/#asterisk SwK (n=SwK@freeswitch/developer/swk) |
| 01:06.05 | Katty | haha |
| 01:06.27 | Katty | i just found old family photos |
| 01:12.08 | pcrane | any idea guys? |
| 01:14.52 | seanbright | pcrane: not sure that is possible with asterisk |
| 01:15.02 | pcrane | ok |
| 01:15.02 | seanbright | pcrane: would probably require modifying app_queue.c |
| 01:15.34 | pcrane | is there like a SIP message I can send out that just displays a message? |
| 01:15.39 | kfife | drmessano: how can you kill that...which has no life |
| 01:15.40 | pcrane | the phones all have LCD displays |
| 01:15.58 | seanbright | pcrane: not sure. but i don't believe so. |
| 01:16.14 | pcrane | if I can send a message, I'll send the message 'next call in queue' |
| 01:16.18 | Assimilate | pcrane, we use fop to show agents all the calls in our queues |
| 01:16.25 | kfife | Qwell: \caps annd\caps0 |
| 01:16.31 | pcrane | hmm |
| 01:16.31 | pcrane | ok |
| 01:16.36 | Qwell | kfife: eh? |
| 01:16.53 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 01:17.04 | kfife | Qwell: Sorry. Keyboard fart. |
| 01:17.05 | drmessano | Qwell: I demand SIP in AsteriskNOW |
| 01:17.08 | drmessano | Oh, wait |
| 01:17.18 | pcrane | I'd not thought of that Assimilate, thanks for the pointer |
| 01:17.19 | seanbright | Qwell: thanks for the tips re: centos x86_64 |
| 01:17.47 | kfife | Qwell: Any idea when they'll update the doc for installing/enabling HPEC in dahdi? |
| 01:21.45 | Assimilate | NOTICE[2588]: chan_sip.c:14383 handle_request_invite: Call from '509xxxxxxx' to extension '509xxxxxxx' rejected because extension not found. But I see it in ext.conf |
| 01:22.20 | kfife | Save, Reload ? |
| 01:22.34 | Assimilate | Yeah just rebooted the whole system heh |
| 01:22.45 | kfife | dialplan show. See it? |
| 01:23.50 | Assimilate | Yep second on the list it provides |
| 01:24.54 | kfife | 5095551212 != 15095551212. One of my ITSP's hands me DNIS without the leading 1, the other ITSP strips the 1. I have to handle the discrepency in my dialplan. Anything like that going on? |
| 01:26.04 | Assimilate | well I am moving from 1.2 to 1.4 and it works in 1.2. So I am handling the extention the same way. So I'd think the provider would hand off the same info |
| 01:26.34 | kfife | Assimilate: Hmmm. Same code verbatim? |
| 01:26.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7cbb67678c9c7d1c) |
| 01:26.55 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:27.03 | kfife | Is sip.conf pointing the inbound call to the right context? |
| 01:27.15 | kfife | I'm assuming it's a sip channel |
| 01:27.39 | Assimilate | kfife, Using the 2.0 branch of the gui to set this up. So I am going from a peer to what they define as a trunk |
| 01:27.56 | drmessano | ..... |
| 01:27.59 | Assimilate | but the context of the user is pointing at this context with the exten |
| 01:28.55 | Katty | bored. |
| 01:29.03 | Katty | throws paper airplanes at seanbright |
| 01:29.07 | kfife | I'm not familiar with the gui. |
| 01:29.26 | kfife | Assimilate: Trunks are provided by peers. |
| 01:29.36 | drmessano | slings starfish shaped potatoes at Katty |
| 01:30.07 | seanbright | is allergic to paper |
| 01:30.15 | drmessano | ROFL |
| 01:30.53 | drmessano | Im allergic to pink paper with the words "YOUR FIRED" written on it |
| 01:31.02 | seanbright | well then i have some bad news... |
| 01:31.13 | drmessano | Yes? |
| 01:31.14 | Assimilate | kfife, 1.4 is new to me... I am used to putting my peers in sip.conf, but the gui put them in users.conf. Is this new because of 1.4 or the gui? |
| 01:31.16 | kfife | Assimilate: Try a wildcard. exten => _X.,1,NoOp(matchy matchy) |
| 01:32.38 | Katty | :< |
| 01:32.53 | Katty | shoots marshmallows at Qwell |
| 01:32.54 | Assimilate | <PROTECTED> |
| 01:34.09 | kornelak | IIRC, putting entries in users.conf propagates changes to sip and the dialplan. Could it be a conflict between users.conf and what might still be in sip.conf? |
| 01:36.24 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 01:37.49 | kfife | I have to run. I wish I could be more help. |
| 01:38.38 | Assimilate | kfife, thanks for the suggestions |
| 01:38.47 | Assimilate | mand been at this too long can't type |
| 01:39.57 | *** join/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
| 01:40.29 | *** part/#asterisk kornelak (n=karl@199.33.79.4) |
| 01:40.31 | Spirits-Sight | wow, I have been reading and reading and still not sure of the answer that I ask eariler |
| 01:40.57 | *** join/#asterisk rcy (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 01:42.07 | *** join/#asterisk johann8384 (n=jonathan@75-120-52-110.dyn.centurytel.net) |
| 01:42.26 | Spirits-Sight | I was wondering, If I want to make a number of calls out on a system, do I need to have that many channels also, say from softphone 1 is making a call, I am making a call form a spa and another softphone 2 is making a call do I need to have three channels |
| 01:43.56 | Assimilate | Spirits-Sight, My sip provider provided call waiting, which gives me two channels so I can make 2 concurrent calls. I have 4 accounts with them and can get 8 calls in our out concurrently. |
| 01:44.18 | Carlos_PHX | Spirits-Sight: You only need provider channels for calls outside your network. |
| 01:44.36 | Spirits-Sight | correct, I know this at less :-) |
| 01:45.11 | Carlos_PHX | So you have three phones, and each phone has a call to/from the PSTN...three channels. |
| 01:45.30 | Carlos_PHX | You have three phones, all on calls to the PSTN, and one is on a conference call to a second person...four channels. |
| 01:45.42 | Carlos_PHX | Is that what you are asking? |
| 01:46.00 | lmadsen | Spirits-Sight: you have a channel between the phone and asterisk, and a channel from asterisk to the end point for each call |
| 01:46.01 | Spirits-Sight | Yes, so my understand is developing right then :-) |
| 01:46.57 | lmadsen | three phones all calling the PSTN is actually 6 channels |
| 01:47.04 | Spirits-Sight | lmadsen: this into the call is connected and then its a direct connection right, |
| 01:47.20 | lmadsen | phone ------ asterisk ------- pstn |
| 01:47.24 | lmadsen | each ------- is a channel |
| 01:47.47 | lmadsen | so with 3 phones you have this: |
| 01:47.49 | lmadsen | phone ------ asterisk ------- pstn |
| 01:47.49 | lmadsen | phone ------ asterisk ------- pstn |
| 01:48.07 | [TK]D-Fender | 6 channels, 3 calls |
| 01:48.28 | Spirits-Sight | I got that :-), but I only pay for the asterisk to pstn right? |
| 01:48.33 | lmadsen | yes |
| 01:48.51 | [TK]D-Fender | Spirits-Sight: Unless you feel like charging yourself |
| 01:48.52 | Spirits-Sight | so to me right now its only three I have to pay for :-) |
| 01:49.02 | lmadsen | Spirits-Sight: yes, but you have 6 channels active |
| 01:49.06 | Carlos_PHX | Spirits-Sight: Right, three channels for $$$ |
| 01:49.11 | Spirits-Sight | LOL, why not maybe I will get rich off myself LOL |
| 01:49.31 | Spirits-Sight | good at less I am understand something |
| 01:50.17 | Spirits-Sight | So, now I am going to try and understand the service end of this, I can have a number of companies providing the channels, right? |
| 01:50.40 | lmadsen | yes |
| 01:51.07 | Spirits-Sight | One for termnatiing the calls and another for incoming and I can have more then one for each if I really wanted to |
| 01:51.09 | Spirits-Sight | ? |
| 01:51.17 | lmadsen | yes |
| 01:51.25 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 01:51.39 | Carlos_PHX | Sure, we use quite a number, depending on location and service specifics. |
| 01:51.48 | Spirits-Sight | So the best thing for money wise is to shop around seperatly not together like they want you to think |
| 01:51.50 | Carlos_PHX | You control all that in your dialplan. |
| 01:52.15 | Carlos_PHX | Spirits-Sight: Probably not really a big savings to be had for a small account. |
| 01:52.22 | Spirits-Sight | I am starting to like this word "dailplan" :-) |
| 01:52.53 | Spirits-Sight | I need to understand for now and in future as things shape |
| 01:55.01 | Spirits-Sight | what do you think about http://www.ipcomms.net/html/FREEDID_Landing2.htm ? |
| 01:55.02 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 01:59.57 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 02:02.49 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 02:07.23 | d3wayne | O.O |
| 02:09.53 | *** join/#asterisk italorossi (n=italoros@201.76.152.227) |
| 02:11.51 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 02:13.01 | *** join/#asterisk edibrac (n=edibrac4@206.173.193.34.ptr.us.xo.net) |
| 02:13.13 | edibrac | can too much jitter cause a red alarm log entry? |
| 02:13.31 | edibrac | or lead to it? |
| 02:14.21 | Carlos_PHX | Jitter has nothing to do with, and can't happen on, your PRI |
| 02:14.54 | Katty | dances with Carlos_PHX |
| 02:15.13 | edibrac | ah it's at a different layer |
| 02:15.20 | tzanger | edibrac: if you slip a frame, you'll slip a frame... red alarm is the absense of a signal |
| 02:15.39 | tzanger | it should not be able to cause a red alarm with jitter |
| 02:15.51 | Carlos_PHX | wonders if Katty had a bit to drink with dinner. |
| 02:16.03 | Carlos_PHX | How were the potatoes? |
| 02:16.10 | edibrac | then ..something like bad/incorrect timing is one possible reason for a red alarm? |
| 02:16.19 | jaytee | usually |
| 02:16.26 | Katty | Carlos_PHX: just bored. |
| 02:16.43 | Carlos_PHX | I've never seen a red alarm on anything that isn't completely borked. This in the US? |
| 02:16.49 | Katty | Carlos_PHX: the potatoes were lovely. |
| 02:17.11 | Katty | edibrac: the only time i've seen a red alarm is when our PRI is not connected to the card. |
| 02:17.31 | edibrac | I get red alarms that last a few seconds - this happens everyday about 3-4 times |
| 02:17.48 | Carlos_PHX | 3-4 times...what... |
| 02:17.51 | edibrac | i have had it happen on two different boxes with different cards |
| 02:18.13 | edibrac | i mean, sometimes 3 sometimes 4 times a day |
| 02:18.22 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 02:18.42 | Carlos_PHX | You should call your service provider, that is not normal. Who is it from? It's PRI? |
| 02:18.51 | edibrac | no real pattern - morning, noon, night, weekdays or weekends |
| 02:18.59 | edibrac | a PRI from XO |
| 02:19.25 | edibrac | they ran tests and say it's fine .. I heard the last resort for something like this is to get them to run a test with a "tbird" |
| 02:19.33 | Carlos_PHX | Hmm. Well, I've seen flakiness with our XO PRIs. I blamed it on the NIU, but maybe I was wrong. |
| 02:19.54 | edibrac | yeah NIU is what they (claimed to have) tested |
| 02:20.01 | Carlos_PHX | A Thunderbird is a small box that does testing on premises instead of looping the NIU. |
| 02:20.13 | jaytee | T-Berd |
| 02:20.16 | Carlos_PHX | Right, but remote loops only test to the NIU, not from the NIU to your demarc. |
| 02:20.26 | Katty | jaytee: tbone? |
| 02:20.31 | Katty | gets fork |
| 02:20.34 | Carlos_PHX | Mmmm t-bone |
| 02:20.42 | edibrac | i have a third server made -- latest stable asterisk by source and zaptel.. should i try it? |
| 02:21.01 | edibrac | could red alarms be a result of a bad/old asterisk or zaptel build? |
| 02:21.05 | jaytee | no, a T-Berd for testing T1's. Bit error rate tests, frame slip tests, etc. |
| 02:21.25 | Carlos_PHX | T-bones are much tastier. |
| 02:21.44 | jaytee | they are but I prefer Filet Mignon |
| 02:21.51 | Katty | NY Strip |
| 02:22.13 | Katty | edibrac: would be worth a shot. |
| 02:22.15 | jaytee | charbroiled on the outside and barely dead on the inside |
| 02:22.20 | Katty | edibrac: you could prove or disprove some stuff |
| 02:22.51 | Katty | jaytee: moo |
| 02:22.53 | Katty | jbot: moo? |
| 02:22.53 | jbot | ACTION mooooooooo! I am cow, hear me moo, I weigh twice as much as you. I am cow, eating grass, methane gas comes out my ass |
| 02:23.03 | jaytee | if you have to use A-1 Steak sauce then what your eating shouldn't be called steak. |
| 02:23.04 | Katty | fascinating. |
| 02:23.17 | *** join/#asterisk ManxPower (n=manxpowe@65.sub-75-251-126.myvzw.com) |
| 02:23.20 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 02:23.24 | Katty | jaytee: ryan agrees with you whole-heartedly |
| 02:23.26 | jaytee | yay, jbot is back! |
| 02:23.47 | jaytee | Katty, Ryan is a guy with good taste. He's with you for one. |
| 02:23.58 | Carlos_PHX | jbot was gone? |
| 02:24.08 | jaytee | yeah, jbot was taking a half day today |
| 02:24.11 | Carlos_PHX | A1...ugh |
| 02:24.28 | Katty | jaytee: he's somehow endured two years of mood swings. it's amazing. |
| 02:24.38 | jaytee | if jbot doesn't use all his PTO time before the end of the year he loses it. |
| 02:24.38 | Carlos_PHX | I was flying at half mast this morning when... oops, wrong channel. |
| 02:25.11 | Carlos_PHX | So, is there a command I can use to get the jbot history? |
| 02:26.06 | Katty | history? |
| 02:26.07 | jaytee | mood swings? pffft, I'm a hot headed Taurus of Irish decent. I could be grumpy as hell in the morning and happy as hell in the afternoon. |
| 02:26.29 | Katty | jaytee: well grumpy and happy don't usually describe me. |
| 02:26.35 | Katty | jaytee: more like giddy and sobbing. |
| 02:26.46 | Katty | jaytee: but hey, i'm with ya on the irish bit (= |
| 02:27.08 | jaytee | I'm rarely giddy, usually I lean toward the goofy myself |
| 02:28.17 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 02:28.59 | Carlos_PHX | OMG, my house smells so bad. Wife & kid eating artichokes in the other room. |
| 02:29.05 | Carlos_PHX | seals office with duct tape. |
| 02:29.22 | Spirits-Sight | LOL |
| 02:30.43 | Katty | never had artichokes. |
| 02:30.59 | x86 | hmm, i've got asterisk on this brand new box and it's using 100% CPU |
| 02:31.16 | x86 | I've twiddled down modules.conf to only what I need, but no dice |
| 02:31.44 | Carlos_PHX | Katty: Consider yourself lucky |
| 02:31.50 | x86 | any ideas? |
| 02:31.51 | jaytee | x86, you're sure it's * using 100% |
| 02:32.20 | Carlos_PHX | They're about as nasty as broccoli. Or anything green and cooked really. Same stench. |
| 02:32.43 | Katty | mmm, broccoli |
| 02:32.44 | Carlos_PHX | x86: Pastebin your top output |
| 02:33.14 | jaytee | asparagus is gross, even long after eating it. |
| 02:34.40 | x86 | err |
| 02:34.45 | x86 | top shows it using 100%, yes |
| 02:34.48 | x86 | or 101% sometimes |
| 02:34.53 | x86 | (i've got 2 cores) |
| 02:37.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 02:37.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 02:38.09 | Carlos_PHX | x86: What version of Asterisk? What protocols (IAX, SIP)? |
| 02:38.18 | Carlos_PHX | Number of channels up? |
| 02:41.21 | Spirits-Sight | How do I install the web insterface for asterisk? (if covered in here) |
| 02:42.00 | x86 | NO channels up, 1.4.22 |
| 02:42.06 | lmadsen | likes asparagus and brocolli |
| 02:42.17 | x86 | no SIP peers, no IAX2 peers, no Zap channels configured... |
| 02:42.31 | x86 | it's more or less a vanilla 1.4.22 setup, with only a dialplan at this point |
| 02:43.02 | Carlos_PHX | x86: That's pretty screwy, can't think of anything that would do that. |
| 02:44.05 | x86 | I turned up logging all the way to debug and not seeing anything extra |
| 02:48.20 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 02:52.04 | x86 | so yeah this is crap.... |
| 02:52.15 | x86 | no way to tell why asterisk is using 100% cpu eh? |
| 02:54.40 | rob0 | strace (1) - trace system calls and signals |
| 02:55.18 | x86 | would work if something was failing |
| 02:55.27 | x86 | but it's not making the system unresponsive or anything |
| 02:55.59 | x86 | well, now it is |
| 02:56.02 | x86 | gah |
| 02:56.06 | x86 | keeps segfaulting too |
| 02:56.19 | x86 | perhaps I'll go back to 1.4.12.1, which works great |
| 02:56.38 | *** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com) |
| 02:59.07 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:59.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:03.35 | x86 | russellb: perhaps you can help me |
| 03:03.51 | x86 | russellb: I've got a mostly-vanilla 1.4.22 install that's using 100% CPU |
| 03:04.23 | x86 | russellb: all stock configs except extensions.conf and modules.conf (disabled a lot of modules to try and see if that was the problem, no dice) |
| 03:04.45 | russellb | with no calls up? |
| 03:05.04 | x86 | with full logging, not seeing any warnings, errors, or anything out of the ordinary |
| 03:05.09 | russellb | ok. |
| 03:05.20 | russellb | are there calls now? |
| 03:05.28 | russellb | because this debugging will lock it up for a bit ... |
| 03:05.28 | x86 | no calls up, no sip peers defined, no iax2 peers defined, no zap/dahdi channels defined |
| 03:05.31 | russellb | ok. |
| 03:05.34 | russellb | install gdb ... |
| 03:05.44 | russellb | then, as root .. # gdb asterisk `pidof asterisk` |
| 03:05.48 | russellb | (gdb) thread apply all bt |
| 03:05.50 | russellb | then pastebin that |
| 03:06.15 | x86 | trying to ssh in... taking forever |
| 03:06.18 | russellb | k. |
| 03:07.22 | x86 | cool, killed asteris, system responsive again ) |
| 03:07.33 | *** join/#asterisk CrazyTux (n=brandon@user-vcauh5h.dsl.mindspring.com) |
| 03:08.33 | x86 | this is ubuntu 8.10-server, intrepid ibex |
| 03:08.38 | x86 | if that matters to you or not |
| 03:09.13 | russellb | i'd rather have more info than I need than not enough |
| 03:09.14 | russellb | :) |
| 03:09.36 | russellb | if it would be trivial to give me ssh access, we could do that too ... |
| 03:09.41 | x86 | yeah yeah, working on it ;) |
| 03:09.48 | x86 | ah ok, lets do that :) |
| 03:10.18 | *** join/#asterisk BeeBuu (n=beebuu@218.13.97.202) |
| 03:11.13 | *** part/#asterisk italorossi (n=italoros@201.76.152.227) |
| 03:11.24 | russellb | just post your IP and root password right here in #asterisk ;-) |
| 03:11.34 | `Sean | lol |
| 03:11.36 | Carlos_PHX | Heh |
| 03:12.11 | russellb | asterisk eating up CPU will no longer be your biggest problem |
| 03:12.19 | BeeBuu | which option like the A(x) option in cmd Dial,but play file to the calling? |
| 03:13.08 | `Sean | lmao |
| 03:13.35 | [TK]D-Fender | BeeBuu: None. |
| 03:14.05 | BeeBuu | [TK]D-Fender: how can i do that? |
| 03:14.07 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 03:14.08 | russellb | yeah, i don't think there is an option for that ... |
| 03:14.16 | [TK]D-Fender | beeGet coding... |
| 03:14.31 | [TK]D-Fender | BeeBuu: get coding... |
| 03:14.33 | BeeBuu | [TK]D-Fender: i want to sound the agent's name to make call part |
| 03:14.58 | [TK]D-Fender | BeeBuu: Huh? |
| 03:15.31 | BeeBuu | let the calling man know which agent is pick up |
| 03:16.32 | russellb | there is no way to do that ... |
| 03:16.56 | BeeBuu | russellb: no way? |
| 03:17.03 | [TK]D-Fender | BeeBuu: He just said that |
| 03:17.10 | [TK]D-Fender | BeeBuu: unload chan_echo.so |
| 03:17.38 | BeeBuu | [TK]D-Fender: and ? |
| 03:17.44 | [TK]D-Fender | beeAnd what? |
| 03:18.06 | BeeBuu | are you sure what the russellb said? |
| 03:18.07 | x86 | russellb: check /msgs |
| 03:18.13 | stencil | Good evening, I'm getting this warning message "LookupBlacklist is depricated please use ${BLACKLIST()} instead" this what i've replaced it with "exten => _X.,1,GotoIf(${BLACKLIST()}?blacklisted,_X.,1)" is this proper method? |
| 03:18.26 | [TK]D-Fender | BeeBuu: Both of us have said it and russellb codes for *. |
| 03:18.42 | [TK]D-Fender | BeeBuu: How many more people have to confirm this for you? Would 10 be enough? |
| 03:19.09 | [TK]D-Fender | stencil: You tell us. What happens? |
| 03:19.15 | BeeBuu | [TK]D-Fender: thanks.i just know what is russellb job last second. |
| 03:19.23 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 03:19.53 | stencil | [TK]D-Fender: I just thought there might be an official method |
| 03:20.08 | *** join/#asterisk gones (n=gones@121.34.23.128) |
| 03:20.09 | Carlos_PHX | I believe BeeBuu is not a native English-speaker and doesn't easily understand what has been said. |
| 03:20.21 | [TK]D-Fender | stencil: unofficially I see no point to that app really... nothing more than 2 lines worth of dialpla to do it yourself.... |
| 03:20.32 | Carlos_PHX | Exactly. |
| 03:20.37 | [TK]D-Fender | stencil: I wouldn't want o to use AstDB for this anyways in most cases |
| 03:20.38 | Carlos_PHX | And then you can have more fun with them. |
| 03:21.00 | [TK]D-Fender | Carlos_PHX: There are a few other similarly wasteful apps out there.. this was a prime offender thought |
| 03:21.05 | [TK]D-Fender | though |
| 03:21.58 | BeeBuu | [TK]D-Fender: are all the OP codes for asterisk? |
| 03:22.02 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 03:22.39 | [TK]D-Fender | BeeBuu: No, not all. |
| 03:23.05 | [TK]D-Fender | BeeBuu: most are digium employees and larger contributors |
| 03:23.20 | [TK]D-Fender | BeeBuu: a few execption here and there. |
| 03:23.48 | BeeBuu | O |
| 03:23.56 | Spirits-Sight | Ok, if I want to be on a three way call and the next person thatdecided to call go to voice mail I would need four channels? |
| 03:24.26 | Spirits-Sight | payed channels |
| 03:24.29 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:24.30 | *** join/#asterisk jsolis (n=Jimmy@200.121.160.35) |
| 03:25.04 | Carlos_PHX | Spirits-Sight: If a call originates from PSTN and hits your server for any reason, it's a channel. |
| 03:25.35 | Katty | is gonna go to bed. |
| 03:25.38 | Katty | ninite |
| 03:26.13 | Spirits-Sight | Carlos_PHX: Is there providers for unlimitied incoming channels |
| 03:26.32 | [TK]D-Fender | Spirits-Sight: what are YOU calling in on? |
| 03:26.35 | Carlos_PHX | Yes, but now. |
| 03:26.38 | Carlos_PHX | Yes, but no. |
| 03:26.51 | jaytee | nite Katty |
| 03:26.53 | Carlos_PHX | We sell unlimited channels, but you pay by the minute then. |
| 03:27.12 | [TK]D-Fender | Katty: Mew. |
| 03:27.19 | Spirits-Sight | [TK]D-Fender: what do you mean, what re you calling on?" |
| 03:27.29 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 03:27.36 | [TK]D-Fender | Spiri on a 3-way call, this is YOU + 2 others. |
| 03:27.46 | [TK]D-Fender | Spirits-Sight: on a 3-way call, this is YOU + 2 others. |
| 03:28.04 | Spirits-Sight | correct |
| 03:28.22 | [TK]D-Fender | Spirits-Sight: If YOU are using a local SIP phone, that is not a channel with your ITSP meaning the 2 OThER people are calls. therefor the THIRD person in would be the one hitting VM |
| 03:28.28 | stencil | thanks [TK]D-Fender & Carlos_PHX |
| 03:28.51 | Spirits-Sight | correct, which would be a channel also right? |
| 03:29.28 | Carlos_PHX | Spirits-Sight: Every SIP connection to a server is a channel. The channels from your server to a PSTN provider are paid, others free. |
| 03:29.30 | pcrane | anyone ever had problems upgrading a snom 360 phone? |
| 03:29.50 | [TK]D-Fender | Spirits-Sight: yes, but not a channel through your ITSP |
| 03:30.22 | [TK]D-Fender | Spirits-Sight: You don't pay for your phone to call * if its a local device |
| 03:31.03 | jaytee | i ate too much |
| 03:31.05 | Spirits-Sight | Ok, so if I am on the phone with person A & B and person C calls, there going to get a busy signal if I don't have three channels that are paid or am I missing something |
| 03:31.19 | Carlos_PHX | jaytee: Where'd you go? How's the class going? |
| 03:31.44 | Carlos_PHX | Spirits-Sight: When you say person A that is meaningless. |
| 03:31.50 | Carlos_PHX | You are on the phone with a PSTN caller. |
| 03:31.51 | jaytee | it's going good. I just ordered pizza when I got back to the hotel. |
| 03:32.01 | Carlos_PHX | Another PSTN caller calls in, that's now two channels. |
| 03:32.14 | Spirits-Sight | [TK]D-Fender: I understand that one, I understand with then my network or any phone / softphone that is setup to connect to the * is channels but they don't cost me any thing |
| 03:32.16 | Carlos_PHX | You try to conference in a PSTN number, that's three. |
| 03:33.05 | [TK]D-Fender | Spirits-Sight: if you max out your channels, the next one gets rejected by your provider. maybe they offer you a VM service. maybe they get a recording, maybe they just get a busy signal |
| 03:33.32 | Spirits-Sight | Carlos_PHX: now a thrid person calls in from a PSTN, thats a channel, which would go to voice mail if thats the way it was setup |
| 03:33.36 | Carlos_PHX | hangs self rather than look at one more Gantt chart |
| 03:33.39 | jaytee | I wished I'd packed my backscratcher. I think I need to go find a tree with rough bark :-) |
| 03:34.05 | Carlos_PHX | Yeah, sometimes I really want to groom my back hair on a tree too. |
| 03:34.20 | Spirits-Sight | LOL |
| 03:34.42 | jaytee | getting old sucks. I got hair growing in all the places I don't want and hair failing out in all the places I want to keep it. |
| 03:34.42 | Carlos_PHX | Spirits-Sight: The voicemail part is confusing. Are you assuming it would go to VM if all channels are busy? Because it won't. |
| 03:35.18 | Carlos_PHX | I stood next to Robin Williams once and he said "Damn!" |
| 03:35.41 | jaytee | hahaha |
| 03:35.45 | Carlos_PHX | Chia Pets cower in my furry presence |
| 03:35.47 | [TK]D-Fender | jaytee: Sounds like a volunteer migration setup :) |
| 03:35.52 | [TK]D-Fender | jaytee: Get grafting! |
| 03:35.54 | jaytee | he's got the hairiest forearms of any guy I've ever seen |
| 03:35.57 | drako | jaytee, i feel the same |
| 03:36.06 | [TK]D-Fender | hands jaytee some CrazyGule |
| 03:36.14 | [TK]D-Fender | Glue* |
| 03:36.20 | Spirits-Sight | I understand that, so if I wanted the call to go to voice mail and I had a three way call going that was all on the PSTN then I would need three paid channels which would allow me to send that last person to voice mail and pay maybe a msg saying on the phone right now please leave you msg :-) |
| 03:36.29 | jaytee | [TK]D-Fender, I was thinking of letting the hair growin in my ears grow out and do a combover. |
| 03:36.52 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 03:36.59 | [TK]D-Fender | Spirits-Sight: You are not getting it... |
| 03:37.27 | [TK]D-Fender | Spirits-Sight: You are allowed X channels. Fill it up and the PROVIDER deals with the ones they dont send to you in whatever way they feel like. |
| 03:37.28 | Carlos_PHX | Spirits-Sight: Sorry, we seem to be miscommunicating on something. But around here it's much simpler if we don't talk in terms of call routing when talking about channels, because the call routing just confuses things. You buy one channel for each concurrent PSTN call whether it's VM or whatever it is. |
| 03:38.01 | jaytee | russellb , still in the office? |
| 03:38.04 | Carlos_PHX | As an example, what we do is let customers go over at first and then talk to them about raising their channel limit. |
| 03:38.06 | *** join/#asterisk sasargen (n=chatzill@173.100.37.65) |
| 03:38.12 | Carlos_PHX | Some others do this and just charge per minute. |
| 03:38.16 | russellb | jaytee: nope ... |
| 03:38.21 | jaytee | good! |
| 03:38.22 | [TK]D-Fender | Spirits-Sight: When I mentioned VM, I'm not talking ASTERISK and YOU in control, I mean the provider taking a message FOR YOU. out of your control in a PITA box to retreive from no doubt |
| 03:38.46 | jaytee | russellb ,did you figure out that segfault issue you were looking at? |
| 03:39.08 | russellb | yup |
| 03:39.08 | Spirits-Sight | OK, I would need two paid channels for a three way call right? (one for person A & one for person B) then if I wanted another person NOT to get a busy signal I would need another channel to allow them to go to the VM of *? |
| 03:39.14 | jaytee | cool! |
| 03:39.42 | Carlos_PHX | Spirits-Sight: While I'd like to say yes, it's important to make the PSTN distinction. Yes, if all the callers are PSTN. |
| 03:39.55 | jaytee | I was ready to slip into a coma by 5pm. The lasagna they served at lunch was making me drowsy all afternoon. |
| 03:40.07 | Spirits-Sight | then I am understanding right |
| 03:40.09 | jaytee | I'm probably gonna gain another 5 pounds this week. |
| 03:41.16 | Spirits-Sight | now to make things more interest for myself, what happens if its another SIP type phone calling me, thats still concidered PSTN right or less its calling my * directly |
| 03:41.27 | Spirits-Sight | not using the phone number |
| 03:41.59 | [TK]D-Fender | Spirits-Sight: Every call is just a bloody call |
| 03:42.26 | [TK]D-Fender | Spirits-Sight: By PSTN we're referring to calls coming in on DID's you PAY FOR from an ITSP |
| 03:42.38 | [TK]D-Fender | PSTN = real world phone number |
| 03:42.56 | [TK]D-Fender | (or more properly the actual phone network) |
| 03:42.58 | [TK]D-Fender | ~pstn |
| 03:42.59 | jbot | pstn is, like, Public Switched Telephone Network, or "please stop the nonsense" |
| 03:43.00 | [TK]D-Fender | ~did |
| 03:43.01 | jbot | hmm... did is Direct Inward Dialing, or just a phone number |
| 03:43.32 | [TK]D-Fender | Spirits-Sight: You pay an ITSP for a DID when people on the PSTN can call and they deliver to you via a VoIP protocol. |
| 03:43.39 | Carlos_PHX | If it's a SIP call that directly addresses your server by IP, then it is free. If it's dialing through an ITSP, then it goes to your ITSP and is paid. |
| 03:44.15 | Spirits-Sight | Got it, I wanted to make sure I understand this stuff, I want to make sure I get ponty of channels to cover a three way call and be able to have another person call in and get Voice Mail on the system, not on the provides system |
| 03:44.45 | Spirits-Sight | great I now undestand this aspect of things as much as I think I should :-) |
| 03:45.14 | [TK]D-Fender | Spirits-Sight: Which, if you don't understand as well as you should is just a s wrong :) |
| 03:46.16 | Spirits-Sight | well the brain does not work as good as I would wish, a number of years ago I would of pick this stuff much better but for somereason its not as easy any more |
| 03:46.50 | jblack | <PROTECTED> |
| 03:47.05 | Spirits-Sight | So please don't get borthed by my silly but siresue inquiryies as I am truly thing to understand |
| 03:47.19 | [TK]D-Fender | jblack: You haven't found an incoming channel limit yet? |
| 03:47.37 | jblack | There's probably one somewhere, but I've never seen it. |
| 03:47.38 | jaytee | has a hemorrhage trying to unravel the convoluted spelling |
| 03:47.45 | jblack | then again, when do I ever need more than 3 or 4 calls at once? |
| 03:47.50 | Spirits-Sight | thing = trying |
| 03:47.51 | jblack | hell. When do I need _1_ ? |
| 03:47.56 | [TK]D-Fender | jaytee: Fortunately I'm fluent in gibberish :) |
| 03:48.03 | jaytee | lol |
| 03:48.11 | *** join/#asterisk sergey (n=Sergey@sergey.iks.ru) |
| 03:48.15 | Carlos_PHX | bets he can max out ipkall... |
| 03:48.18 | [TK]D-Fender | jblack: I use my IPKALL # as an overflow for my office |
| 03:48.26 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 03:48.30 | [TK]D-Fender | jblack: PRI failover. |
| 03:48.33 | jblack | [TK]D-Fender: I wouldn't be surprised if they have no limits. They make money off the calls, so more calls, more better. |
| 03:48.46 | jaytee | everytime I install linux and get to the language selection I have to wonder how many people out there actually choose Esperanto. |
| 03:49.34 | jblack | heh. Nothing like a language that was born dead. ;) |
| 03:49.56 | jblack | I bet they could have had more success if they didn't need all those squigglies and such. |
| 03:50.00 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 03:50.07 | jaytee | I've never met a single person that claims to be able to speak it, yet it's always there as a language option. |
| 03:50.50 | jaytee | I bet there's more people on the planet that speak fluent Klingon |
| 03:50.55 | [TK]D-Fender | jaytee: Has it been translated into snaskit though? |
| 03:51.01 | [TK]D-Fender | sanskrit* |
| 03:51.30 | Carlos_PHX | Ok, so how does ipkall make money? |
| 03:51.33 | jaytee | I've never seen sanskrit on the list when installing. at least not in CentOS. |
| 03:51.46 | jaytee | Carlos_PHX, merchandising! |
| 03:52.09 | Carlos_PHX | It's all in the moinchandizing! |
| 03:52.29 | sergey | Hi. Asterisk SVN-trunk-r156051, tcpenable=yes, transport=tcp, when called in debug have INVITE SIP/2.0/TCP but tcpdump have not any ip , and slence in phone |
| 03:52.39 | jaytee | May the Schwartz be with you! |
| 03:53.27 | sergey | and receive sip/tcp - fine |
| 03:54.14 | [TK]D-Fender | sergey: Voice is still rtp (UDP)... so TCP has nothing to offer in improving that... |
| 03:55.32 | jaytee | the only practical use of tcp with SIP is to talk to snobby, snooty VOIP systems like OCS and Exchange UM without using something like sipX as a proxy |
| 03:55.45 | orkid | unless tcp is given preference by the router? |
| 03:55.58 | jaytee | yeah, that's a thought |
| 03:56.32 | Carlos_PHX | I had heard that OCS added UDP, any truth? |
| 03:56.41 | luke-jr | SIP should have always been TCP |
| 03:56.43 | luke-jr | UDP isn't reliable |
| 03:56.45 | jaytee | Carlos_PHX, supposedly in R2 |
| 03:56.55 | Carlos_PHX | ROFL.... |
| 03:57.00 | Carlos_PHX | Yeah, UDP sucks. |
| 03:57.05 | sergey | it via satellite i-direct system and require sip/tcp :-( |
| 03:57.06 | Carlos_PHX | None of us use it. |
| 03:57.09 | luke-jr | UDP has a different purpose |
| 03:57.19 | luke-jr | like realtime audio |
| 03:57.26 | Spirits-Sight | [TK]D-Fender: I have checked all of the listings for ITSP and I am still trying to find one that has a package of say 2 or 3 channels inbound that is not that costly, also looking for one that has 2-4 channels for outbound or even unlimited as long as the rate is not very high, I think .01 or .015 would be good if better then great any ideas any one? I have been calling different place today and reading all day just about t |
| 03:57.34 | [TK]D-Fender | orkid: Who cares about give SIP preference? |
| 03:57.36 | luke-jr | but it sucks to be billed for 24+ hours of a call because someone dropped your "Hanging up" packet |
| 03:57.42 | Carlos_PHX | sergey: You're going to try voice over a satellite connection??? |
| 03:58.19 | Carlos_PHX | Spirits-Sight: There are lots of them that are not costly. You're just trying to be cheap. |
| 03:58.30 | Carlos_PHX | You're pushing into "crap service" territory. |
| 03:58.32 | [TK]D-Fender | Spirits-Sight: Go look at all the suggested providers and just pick one. |
| 03:58.45 | luke-jr | Spirits-Sight: 1.3 c/min outbound with Voipjet |
| 03:58.58 | Carlos_PHX | Is there a jbot response for cheaping out on service? |
| 03:59.18 | [TK]D-Fender | ~cheap |
| 03:59.19 | jbot | somebody said cheap was a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 03:59.23 | luke-jr | Spirits-Sight: SellVoip is cheap, but they are very unreliable |
| 03:59.30 | [TK]D-Fender | Carlos_PHX: Applies to products as well as service |
| 03:59.35 | sergey | Carlos_PHX: yes, we have some users (about 10) via sat. and no problem but no asterisk |
| 03:59.37 | Carlos_PHX | And women. |
| 03:59.41 | Carlos_PHX | Well, maybe not. |
| 03:59.58 | Carlos_PHX | sergey: What is your latency and jitter. |
| 04:00.03 | [TK]D-Fender | Carlos_PHX: Women are both products & service ;) |
| 04:00.14 | Carlos_PHX | And sometimes cheap is good. |
| 04:00.23 | sergey | lat 600ms |
| 04:00.28 | Carlos_PHX | When you're up for that sort of thing. |
| 04:00.44 | Carlos_PHX | Well, that's fast for satellite, but unusable for voice, no? |
| 04:00.52 | Carlos_PHX | And jitter? |
| 04:02.30 | jaytee | some ITSP's have more latency than Senator Larry Craig |
| 04:02.57 | Carlos_PHX | Who is he, and why is he latent? |
| 04:03.21 | jaytee | Carlos_PHX, scandal? foottapping in an airport restroom? |
| 04:03.36 | jblack | he was a gay republican senator that tried to elicit sec in public restrooms by tapping his foot. |
| 04:03.46 | *** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 04:04.06 | Carlos_PHX | Oh yeah. What an idiot. |
| 04:04.13 | Carlos_PHX | Everyone knows the new code is |
| 04:04.24 | Carlos_PHX | wonders if that was my outside voice |
| 04:04.34 | jblack | well, the idiot part was that he spearheaded the atttempt to get anti-homosexual laws on the books. |
| 04:04.37 | jaytee | 600ms latency on a sat link? how does regular voice over sat work then? |
| 04:05.07 | sergey | Carlos_PHX: no it no problem to phone (g729/gsm) |
| 04:05.07 | Carlos_PHX | Voice over satellite always sucks, but it's managed very differently from the data side. |
| 04:05.19 | Carlos_PHX | My satellite phone sounds reasonable, but the data is horrible. |
| 04:05.37 | Carlos_PHX | Huh, interesting to hear. |
| 04:05.42 | Carlos_PHX | Who is the service provider? |
| 04:06.29 | sergey | our company is service provider :-) |
| 04:07.19 | Carlos_PHX | What is your company? Name, site? |
| 04:07.28 | Carlos_PHX | Always interested in another option for our customers. |
| 04:07.49 | sergey | www.iks.ru |
| 04:08.29 | Assimilate | SHould dahdi show channels show anything if there is no one on the line? |
| 04:09.30 | jaytee | yeah, should still list channels |
| 04:09.52 | Assimilate | hrm its blank and I just got chan_dahdi.c:11696 setup_dahdi: Unable to load zapata.conf |
| 04:10.14 | jaytee | ? |
| 04:10.36 | jaytee | DAHDI doesn't use zapata.conf |
| 04:10.49 | Carlos_PHX | realizes it is o-Scotch-thirty, off for the night. |
| 04:10.57 | jaytee | nite Carlos_PHX |
| 04:11.06 | Assimilate | well I was using zap show status/show channels etc, but it said that its being replaced with ther other command |
| 04:11.24 | jaytee | Assimilate, what version of * ? |
| 04:11.51 | Assimilate | 1.4.22 |
| 04:12.29 | jaytee | Assimilate, you can use either zaptel or dahdi with 1.4.22 but if you run 1.6.x you have to use DAHDI |
| 04:12.55 | Assimilate | Ok, using a digium card with zaptel modules compiled. |
| 04:14.08 | jaytee | and zaptel modules loaded? |
| 04:15.03 | Assimilate | lsmod shows wcte12xp, which was the one I used on my 1.2.5 box for this card |
| 04:15.19 | Assimilate | but it looks like all the other modules are there loaded as well |
| 04:15.49 | jaytee | you can comment out the others in the zaptel init script |
| 04:15.52 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 04:16.22 | jaytee | wcte12xp is for T1/E1 IIRC |
| 04:17.06 | Assimilate | Digium Wildcard TE120P is the card |
| 04:17.13 | jaytee | yep |
| 04:17.22 | jaytee | so you've got a T1? |
| 04:17.53 | Assimilate | Yeah, I am moving to a new box. I had this card running in a 1.2.5 box but the HDD is going out fast so we are moving and upgrading. |
| 04:18.04 | Assimilate | painful.... |
| 04:18.33 | x86 | wow, this audiocodes MP-114 4-port FXO ATA is the weirdest ATA config interface I've ever seen |
| 04:18.43 | x86 | seems like it has a lot of options... maybe too many |
| 04:19.21 | jaytee | Assimilate, did you run ztcfg ? |
| 04:19.31 | [TK]D-Fender | x86: Yeah... powerful but cryptic. The learning curve is a bit steep. |
| 04:20.07 | x86 | [TK]D-Fender: yeah it could be a little more straight-forward |
| 04:20.10 | Assimilate | jaytee, nope |
| 04:20.23 | x86 | [TK]D-Fender: took your advice and went with an ATA to handle the analog stuff ;) |
| 04:20.30 | x86 | for this new deployment |
| 04:20.31 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 04:20.39 | [TK]D-Fender | x86: Once you build the config you can export it making mass deployment incredibly easy and maintainable. |
| 04:20.54 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net) |
| 04:21.14 | x86 | is it possible to make it register all 4 FXO lines to asterisk as a single SIP peer? |
| 04:21.45 | x86 | so I can do a simple Dial(SIP/ac-mp-114/${EXTEN}), and make it hunt an unused FXO line? |
| 04:21.51 | Assimilate | jaytee, I ran it with -vv and it shows 23 channels and then 1 d-channel |
| 04:22.08 | [TK]D-Fender | x86: yup |
| 04:22.13 | jaytee | Assimilate, run genzaptelconfig and then ztcfg -vvvc |
| 04:23.04 | Assimilate | I get an error... grep: /etc/asterisk/zapata.conf: No such file or directory |
| 04:23.26 | x86 | [TK]D-Fender: can you help me with that? :P |
| 04:23.31 | jaytee | is there a zapata.conf file in /etc/asterisk ? |
| 04:23.50 | [TK]D-Fender | x86: Nope, its been like 2 years since I've touched them... |
| 04:23.55 | Assimilate | I'm restoring my backup file |
| 04:24.23 | x86 | [TK]D-Fender: grr |
| 04:24.25 | jaytee | Assimilate, did you compile or install from packages? |
| 04:24.30 | Assimilate | compile |
| 04:24.56 | x86 | [TK]D-Fender: can I make all inbound calls come to asterisk on a single trunk also? |
| 04:25.21 | [TK]D-Fender | x86: Quite likely |
| 04:25.36 | jaytee | check the upgrade.txt file in the asterisk tarball. Not certain if they dropped zaptel in 1.4.22 as the default. In AsteriskNOW it defaults to DAHDI but that's a whole nuther animal. |
| 04:26.21 | Assimilate | jaytee, it looks like it worked. |
| 04:26.35 | jaytee | do a zap show status or zap show channels |
| 04:26.39 | Assimilate | it has now reset the channels like it used to |
| 04:26.50 | jaytee | should show all your b channels with zap show channels |
| 04:26.51 | Assimilate | yep 23 channels listed. |
| 04:26.55 | jaytee | cool! |
| 04:27.01 | Assimilate | Ok now to get the calls answered |
| 04:27.04 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 04:27.14 | Assimilate | I guess we know zaptel is still in 1.4.22 at least :) |
| 04:27.18 | jaytee | yep |
| 04:27.47 | jaytee | That's what I thought I'd remembered reading but I'm running an earlier 1.4.x in production and testing 1.6 |
| 04:27.52 | jaytee | in class we're using 1.6 |
| 04:28.19 | Assimilate | I'm always a version behiend |
| 04:29.01 | jaytee | Assimilate, they call it "the bleeding edge" for a reason. Bleeding ulcers, bleeding hemorrhoids, bleeding gums, etc. etc. |
| 04:30.36 | Assimilate | jaytee, trying to use the GUI.... thats bleeding edge enough for me :S |
| 04:31.45 | jaytee | Assimilate, the asterisk-gui? or freepbx? |
| 04:31.52 | Assimilate | asterisk-gui |
| 04:31.55 | Assimilate | 2.0 branch |
| 04:32.16 | jaytee | haven't messed with the 2.0 version. heard it's got alot of improvements |
| 04:32.42 | jaytee | the beta of AsteriskNOW 1.5 gives you the option of freepbx or asterisk-gui 2.0 |
| 04:33.14 | Assimilate | its made life a little easier. At least I can configure some options I have never been able to get working like follow me etc. |
| 04:33.17 | *** part/#asterisk jsolis (n=Jimmy@200.121.160.35) |
| 04:34.55 | jaytee | Assimilate, I haven't messed with followme. I'm kinda paranoid to begin with. figure we're all under surveillance most of the time and I don't want anyone following me. |
| 04:34.56 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net) |
| 04:36.40 | Assimilate | lol |
| 04:36.52 | *** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com) |
| 04:41.06 | jaytee | time for sleep |
| 04:41.09 | jaytee | nite all |
| 04:41.37 | *** part/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
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| 04:44.44 | *** join/#asterisk prodyan (n=ian@124.104.71.66) |
| 04:44.53 | prodyan | hello all |
| 04:53.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 04:57.43 | x86 | anyone have the firmware needed for the polycom IP330 phones? |
| 04:57.57 | x86 | it's too late to get ahold of my distributor |
| 04:58.01 | [TK]D-Fender | x86: www.polycom.com |
| 04:58.07 | x86 | gotta install this new system first thing in the morning |
| 04:58.15 | x86 | [TK]D-Fender: they allow downloads now? |
| 04:58.20 | *** join/#asterisk pcrane (n=pcrane@202.20.97.154) |
| 04:58.27 | [TK]D-Fender | x86: Always did, just not the latest |
| 04:58.43 | x86 | oh, well I think I need the latest for the 330's to work right |
| 04:58.58 | x86 | (as with 550's, 650's, 660's, 670's, and 430's) |
| 04:59.46 | x86 | I've got firmware for the 300/301, 501, and 601, but not sure if that'll work correctly on my 330's |
| 05:00.15 | [TK]D-Fender | x86: thats why God invented "reading" |
| 05:01.47 | x86 | s/not sure/almost positive because I've tried it before/ |
| 05:02.08 | aiksa[LV] | hi everyone again :)) |
| 05:02.43 | aiksa[LV] | after few days of trying to get "loopback" dynamic zaptel chan i have given up |
| 05:02.45 | aiksa[LV] | :P |
| 05:03.00 | aiksa[LV] | back and forth but stumbling into the same tree. |
| 05:03.23 | x86 | [TK]D-Fender: so you don't have the new firmware eh? |
| 05:03.38 | [TK]D-Fender | x86: Depends whats "new" |
| 05:03.45 | aiksa[LV] | however i stumbled upon the fact that speex library has acoustic echo cancelation |
| 05:03.59 | x86 | [TK]D-Fender: what about a sample sip.cfg, wasn't there something different than the sip.cfg and/or phone1.cfg that worked with the 301/501/601, but not 330? |
| 05:04.27 | [TK]D-Fender | x86: You ralize that without mentioning precise versions that you are spinning in retarded circles? |
| 05:04.27 | aiksa[LV] | would it be "switched on" if I added speex as a codec for asterisk 2 asterisk communication? |
| 05:04.38 | [TK]D-Fender | x86: Jeebus |
| 05:04.51 | [TK]D-Fender | x86: Go download the changelogs and get a clue already! |
| 05:04.58 | [TK]D-Fender | ~cluebat x86 |
| 05:04.58 | jbot | ACTION pulls out a ClueBat (tm) and thwaps x86. |
| 05:05.04 | x86 | [TK]D-Fender: i'm not sure which version i need for the 330... thought i made that clear already |
| 05:05.29 | x86 | guess i can use the one existing on the 330's already |
| 05:05.50 | x86 | but still would like a sample sip.cfg and phone1.cfg that works with a 330, if you have one handy |
| 05:05.53 | [TK]D-Fender | x86: and I thought that if someone really wanted to solve their problems they'd exhaust the readily available source at the get-go. But thats prejudiced against the reading-challenged :p |
| 05:06.26 | x86 | i'm exhausting this readily-available resource ;) |
| 05:06.54 | [TK]D-Fender | x86: Yes.... I'm tired of this already. No go to their site and read up on the versions available |
| 05:07.32 | x86 | i'll just use existing firmware on 330's |
| 05:07.34 | x86 | stock one |
| 05:07.43 | x86 | what about sip.cfg though, can you hook me up? |
| 05:07.48 | *** join/#asterisk Ccomp5950 (n=Ccomp595@66.190.102.236) |
| 05:08.32 | [TK]D-Fender | x86: And am I to guess which firmware you have that wouldn't BREAK at some random config I might pass you? |
| 05:08.50 | [TK]D-Fender | x86: Are you not getting that you have to get SPECIFIC on your versions before you FUBAR yourself? |
| 05:09.11 | [TK]D-Fender | x86: x86 Stop bing generic about this. Go to the damn site and READ |
| 05:09.28 | [TK]D-Fender | ARGH |
| 05:09.29 | x86 | i dont have the phones to check the version that's on them |
| 05:09.36 | x86 | whatever is stock |
| 05:10.07 | [TK]D-Fender | x86: And now I'm supposed to know what "stock" is? You haven't even gotten off your ass to read what your phone has on it. |
| 05:10.09 | x86 | do you have a sip.cfg that would work with a stock 330? |
| 05:10.17 | [TK]D-Fender | x86: Ok, I'mm off of this. |
| 05:10.20 | x86 | shakes fist |
| 05:10.56 | [TK]D-Fender | shakes his head |
| 05:11.24 | x86 | i'll just configure the three phones manually |
| 05:11.30 | x86 | not worth the trouble ;) |
| 05:16.52 | [TK]D-Fender | x86: No... you're definitely not ;) |
| 05:20.18 | Assimilate | I am trying to use the GUI to make an incoming rule and it won't let me hit update. I know thats a little out of this channels realm but any ideas? #asterisk-gui seems dead |
| 05:35.44 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
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| 06:05.18 | Micc | Why is it that caller id doesn't show properly when calls come in from our Zap/PRI. |
| 06:05.45 | Assimilate | Micc, Examples? |
| 06:06.02 | Micc | It only shows the phone number. |
| 06:06.04 | Micc | never the name. |
| 06:07.19 | Micc | Does asterisk have to use the number to do a lookup for the name? |
| 06:09.54 | Assimilate | Micc, I don't know about that, but I know I had to call the provider of my PRI and have then enable features to get the ffull callerid on my end. |
| 06:10.11 | Micc | aha, thats probably it then. |
| 06:10.12 | Micc | thanks. |
| 06:13.44 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 06:18.24 | drmessano | CNAM lookup isn't always free or included |
| 06:23.50 | Assimilate | Is there a command from source that will compile blank config files for you? |
| 06:25.06 | drmessano | make samples |
| 06:25.14 | drmessano | but they're not "blank" |
| 06:26.33 | Assimilate | yeah I ran into problems with them. Mainly all calls going to the demo etc. |
| 06:35.55 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 06:43.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:46.17 | *** join/#asterisk CrazyTux (n=brandon@user-vcaup4j.dsl.mindspring.com) |
| 06:47.02 | drmessano | I'm beginning to think all VoIP providers are shitty |
| 06:47.59 | Assimilate | drmessano, we have been with ours for 4 years now. They have gotten a lot better. We now have our own PRI so we only use them for ourgoing LD |
| 06:48.14 | drmessano | Each one in a different way, of course, like redheads |
| 06:48.50 | *** join/#asterisk CrazyTux (n=brandon@user-vcaup4j.dsl.mindspring.com) |
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| 07:03.16 | baliktad | Can anyone spot the reason Asterisk is complaining here? http://pastebin.ca/1254092 |
| 07:04.35 | drmessano | PMS? |
| 07:05.00 | baliktad | Ahh, no I see it now |
| 07:05.02 | baliktad | I'm an idiot |
| 07:05.19 | drmessano | Glad I could help |
| 07:05.38 | drmessano | I dont ask for paypal donations for my sarcasm, but you know, I have to eat too... |
| 07:06.53 | baliktad | hmm, well I know what the problem is now, I just don't know how to solve it |
| 07:10.39 | baliktad | ok I would like to retract my previous statement |
| 07:12.17 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
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| 07:51.06 | nicox | anyone there who can help me to solve a problem? |
| 07:51.38 | Assimilate | No one knows... Since no one knows the problem |
| 07:51.56 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:53.47 | drmessano | fdisk |
| 07:54.19 | Assimilate | 42 |
| 08:00.52 | *** join/#asterisk pnlarsson (n=nick@83.218.94.2) |
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| 08:02.14 | pnlarsson | Any chan_skinny people here? I want two lines with the same number - is it possible? If i add two line=>103 i can't call the phone... |
| 08:03.12 | nicox | is someone there with big iax-experience |
| 08:04.13 | *** join/#asterisk jc_yyz2bkk (n=jc@ppp-58-8-60-233.revip2.asianet.co.th) |
| 08:04.18 | drmessano | nicox: Just ask the fucking question |
| 08:04.19 | Assimilate | pnlarsson, Why not add two peers then use one ext to calthem both? Or is this not what you are looking for? |
| 08:05.22 | jc_yyz2bkk | hi... i have a call file that goes to an extension that runs an agi script... in the call file i set var=1... will this var get passed on to the agi script? |
| 08:06.24 | Assimilate | jc_yyz2bkk, Is it a macro tht calls the agi? Also does it ever say something to effect of "New Stack" in your debug when the agi is called? |
| 08:08.06 | jc_yyz2bkk | the test.call file forwards to ext 777... ext 777 calls an agi script... is this what you mean... im check the new stack thing |
| 08:08.52 | drmessano | nicox: ask your question already |
| 08:09.12 | Assimilate | lol |
| 08:09.16 | nicox | i have a trunk between 2 asterisks and there are about 100 calls per minute (+-200) and every hour or so on i have 1 call which will be rejected |
| 08:09.23 | nicox | and i have no idea how to debug |
| 08:09.57 | nicox | chan_iax2.c: Call rejected by 10.x.y.z: No authority found |
| 08:10.12 | jc_yyz2bkk | it says test.agi starting in a new stack |
| 08:11.04 | Assimilate | jc_yyz2bkk, Now I am no expert in this stuff, but I have been told that when it executes a new stack that that stack has none of the previous variables in it. |
| 08:11.42 | jc_yyz2bkk | hmmm... |
| 08:12.06 | Assimilate | Like I was trying to set a specific ringtone for a special phone number then forward it to the queue. The queue executed in a new stack and didn't have my ringtone setting |
| 08:12.25 | Assimilate | if I bypassed the queue and made it ring a phone the ringtone would work |
| 08:12.40 | pnlarsson | Assimilate: I want two lines on the phone with the same number... In skinny as well |
| 08:13.17 | jc_yyz2bkk | assimilate; im gonna check up on passing vars to agi scripts... |
| 08:13.34 | Assimilate | pnlarsson, set them up as peers that can dial each other then make your dialplan dial them both |
| 08:13.43 | Assimilate | dial(SIP/100&SIP/101 |
| 08:13.55 | nicox | drmessano: any idea? |
| 08:13.58 | pnlarsson | Assimilate: It's not SIP, it's SCCP aka skinny |
| 08:14.07 | Assimilate | then use that tech |
| 08:14.18 | Assimilate | I use zaptel so |
| 08:14.34 | Assimilate | dial(zap/g2/100&zap/g2/101) |
| 08:14.55 | Assimilate | but you should be able to do show channeltypes and get the codes for it |
| 08:14.59 | pnlarsson | It's the phone that is the issue, with CCM i can set up two lines with the same linenumber |
| 08:15.02 | jc_yyz2bkk | is voip-info.org down for anyone else? |
| 08:15.23 | Assimilate | no its up here |
| 08:15.37 | pnlarsson | As with a Polycom SIP, you can have 2 lines with the same number |
| 08:15.44 | jc_yyz2bkk | dammit thailand |
| 08:17.09 | *** join/#asterisk mvanbaak (i=vanbaak@asterisk/contributor-and-bug-marshal/mvanbaak) |
| 08:18.55 | pnlarsson | mvanbaak: 2 lines with the same number on a 7960 with chan_skinny? |
| 08:20.04 | jjshoe | re |
| 08:21.23 | mvanbaak | nope |
| 08:21.39 | mvanbaak | line1 and line2 are different lines in skinny.conf |
| 08:26.32 | pnlarsson | mvanbaak: CCM can do this. Is there anyway to get the same result? |
| 08:27.48 | kaldemar | pnlarsson: why do you need this? |
| 08:28.09 | pnlarsson | To easy swap between two calls |
| 08:28.45 | mvanbaak | I have no idea |
| 08:29.10 | pnlarsson | The customer have this today, so it would be nice... |
| 08:29.13 | kaldemar | how is it making it hard if the lines configured in the phone don't have the same username? |
| 08:29.31 | mvanbaak | pnlarsson: I just moved into my new house. and my asterisk box is not powered yet. cant help you right now, sorry |
| 08:29.32 | pnlarsson | mvanbaak: thanks |
| 08:30.21 | pnlarsson | I tried some different ways, wanted to know if i was missing some bits |
| 08:30.46 | kaldemar | most phones support more than one line with just one configured user anyway. |
| 08:31.17 | mvanbaak | I think the cisco works so as well |
| 08:31.22 | mvanbaak | just add 1 line |
| 08:31.31 | mvanbaak | the second call should show up on the second button |
| 08:31.36 | mvanbaak | I think |
| 08:32.05 | kaldemar | or can it be that chan_skinny is just retarded in that way. |
| 08:32.21 | mvanbaak | lol |
| 08:32.28 | pnlarsson | I tried, with line => 103 and then line => 103 |
| 08:32.44 | mvanbaak | remove the second 'line => 103' |
| 08:33.04 | mvanbaak | and attach only line 103 to the device, no other lines and speeddials |
| 08:33.20 | mvanbaak | if that works we can see how to fix the situation where you want speeddials etc |
| 08:34.06 | pnlarsson | mvanbaak: with one line => 103 and no speeddails, i get 103 on the top linebutton and nothing else |
| 08:34.23 | pnlarsson | And i can't make two calls |
| 08:34.43 | kaldemar | does it matter what the line parameter says? |
| 08:35.17 | pnlarsson | With two lines in skinny.conf, i get 103 on the first and second, but i can't dial the phone... |
| 08:35.41 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:35.56 | pnlarsson | <PROTECTED> |
| 08:35.56 | pnlarsson | <PROTECTED> |
| 08:35.58 | kaldemar | you could have line => 203 as the other and set the caller id's the same. |
| 08:39.04 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-193-146.telkomadsl.co.za) |
| 08:41.17 | pnlarsson | That works but i then need to make the dial more complicated... |
| 08:41.38 | pnlarsson | If i make them known as Line1 and Line2 it could work |
| 08:42.25 | creativx | its kinda funny when googles captcha is so quirky not even a non-disabled person like myself cant read wtf it says |
| 08:42.38 | pnlarsson | Don't like the fact that it's ringing on both lines when only one caller, Dial(Skinny/Line1@103&Skinny/Line2@103) |
| 08:43.00 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:44.11 | kaldemar | pnlarsson: make it dial one at a time then. |
| 08:45.06 | kaldemar | you can do what ever you want with the call. |
| 08:46.46 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 08:47.18 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 08:47.33 | pnlarsson | Another skinny issue, i can't make a second call if i called out on the first |
| 08:47.54 | pnlarsson | But if the first call was inbound, i can make an other call... |
| 08:49.05 | pnlarsson | kaldemar: i'm trying to get this to work within freepbx, so i'm a little limited when it comes to dailing |
| 08:49.59 | kaldemar | maybe freepbx people could help you more then. |
| 08:50.48 | pnlarsson | Well it's more chan_skinny than freepbx at the moment |
| 08:57.45 | jc_yyz2bkk | hi, does anyone know of an example agi script that gets vars from the extensions.conf? |
| 08:58.51 | *** part/#asterisk pnlarsson (n=nick@83.218.94.2) |
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| 09:07.30 | *** join/#asterisk stix_ (n=stix@exchange2003.corporate.billetkontoret.dk) |
| 09:08.16 | stix_ | Guys, I am having a weird problem on my asterisk right now |
| 09:09.07 | stix_ | I am using 1.4.17 and I have two extensions which are not talking but they seem busy when I type show channels. |
| 09:09.55 | stix_ | I can type "show channels", the next time I do it nothing appears. Then I can type show cha<tab> and then the cli dies and I have to exit it with ctrl+c |
| 09:10.04 | stix_ | I can enter the cli again |
| 09:10.22 | stix_ | but it doesn't respond to the "soft hangup" command either |
| 09:10.30 | stix_ | what's wrong here? |
| 09:10.46 | stix_ | Ppl can still call on the system |
| 09:11.48 | WimpMan | That happened to me when I tried to use chan_sccp. |
| 09:12.19 | stix_ | I havn't touched the system, this suddenly appears |
| 09:12.40 | stix_ | and it is a production system with 70 users :) |
| 09:13.14 | stix_ | WimpMan, which asterisk version were you using? |
| 09:13.39 | WimpMan | 1.4.21 |
| 09:14.05 | WimpMan | 1.4.21.1 to be precise. |
| 09:14.14 | stix_ | okay |
| 09:15.05 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:17.59 | angryuser | stix_: check the logs if all modules are loaded fine |
| 09:18.18 | *** join/#asterisk h-idrisi (n=h-idrisi@212.100.196.195) |
| 09:20.02 | *** join/#asterisk lanning (n=lanning@c-98-210-178-28.hsd1.ca.comcast.net) |
| 09:20.09 | *** part/#asterisk lanning (n=lanning@c-98-210-178-28.hsd1.ca.comcast.net) |
| 09:21.10 | prodyan | The 'show channels' command is deprecated and will be removed in a future release. Please use 'core show channels' instead. |
| 09:22.13 | Assimilate | I'm so deprecated around here |
| 09:22.19 | *** join/#asterisk rcahilig (n=Administ@58.69.237.182) |
| 09:22.43 | jc_yyz2bkk | how do i get the vars passed from extensions in my agi script? |
| 09:22.57 | *** join/#asterisk bl4q (n=Bl@dslb-088-067-031-205.pools.arcor-ip.net) |
| 09:24.18 | prodyan | use fputs(STDOUT,"GET VARIABLE \"varnamehere\"\n") |
| 09:24.34 | prodyan | then just fgets after you use that command |
| 09:24.52 | rcahilig | hi, We have a DID number from DID exchange, how do I configure it to Asterisk server, we are using Asterisk 1.4 |
| 09:25.02 | rcahilig | we will use the DID for inbound |
| 09:25.06 | jc_yyz2bkk | but in extensions.conf i use agi(test.agi|var) ... no var name... |
| 09:26.37 | prodyan | hmm |
| 09:26.59 | prodyan | then set the var before passing it |
| 09:27.14 | *** join/#asterisk wry (n=wry@195.56.160.104) |
| 09:27.31 | jc_yyz2bkk | :) yep i could do that... and im tired of looking for this answer |
| 09:27.39 | prodyan | lolx |
| 09:27.47 | prodyan | all good |
| 09:28.11 | wry | hey. anyone has any idea what might be the cause of ExtenSpy starting properly, attaching to given chan but then only silence can be heared? |
| 09:28.57 | jc_yyz2bkk | wry... just a guess but NAT issue? |
| 09:29.16 | *** join/#asterisk kerx (n=prepro@adsl-68-126-112-153.dsl.irvnca.pacbell.net) |
| 09:29.33 | wry | NAT..? |
| 09:29.48 | wry | could you explain a bit further? :) |
| 09:30.00 | wry | im trying to listen to a Zap chan. |
| 09:32.04 | phpboy | can you run two calls through 1 physcial ISDN line? |
| 09:32.07 | jc_yyz2bkk | network... like the proper ports arent open... but like i said its just a guess |
| 09:32.07 | phpboy | BRI |
| 09:32.36 | kaldemar | phpboy: i BRI has two B-channels, so that would be a yes. |
| 09:32.53 | phpboy | I get that |
| 09:32.58 | WimpMan | phpboy: Two active ones to be precise. |
| 09:33.01 | *** join/#asterisk miloux (i=milu@213.88.194.123) |
| 09:33.02 | phpboy | but through 1 single physical cable? |
| 09:33.17 | phpboy | so, a single port card I can run two calls through one line? |
| 09:33.50 | prodyan | guys am i right in this assumption?, no one can call you through SIP if they don't register (as peer,user,friend) in your * server? |
| 09:33.55 | kaldemar | just like you can have n tcp-sessions over one cable. |
| 09:34.10 | WimpMan | prodyan: No |
| 09:34.11 | kaldemar | prodyan: no |
| 09:34.15 | prodyan | ohh oki |
| 09:34.54 | phpboy | WimpMan: The thing is, I'm using mISDN on one of my boxes and 1 call = 1 port |
| 09:34.57 | SwK | phpboy, ISDN lines are digital time domain multiplexed lines... so 1 cable but multiple channels.... |
| 09:35.02 | phpboy | 4 cables for my 2 ISDN lines |
| 09:35.22 | SwK | like a PRI is 23B channels and its just 1 cable |
| 09:35.39 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 09:35.58 | WimpMan | phpboy: No. A port in mISDN refers to an interface, not to a channel. |
| 09:36.17 | tzafrir_laptop | (like a port on the card) |
| 09:36.19 | WimpMan | SwK: And for the rest of the world it's even 30 channels :-) |
| 09:36.52 | SwK | tru date |
| 09:37.08 | SwK | 23b on a ulaw PRI, 30 on a alaw'ers pri |
| 09:37.10 | phpboy | WimpMan: but the trouble I'm running into is if I push a call through say port 1, it works. when I try to push another call through port 1... congestion |
| 09:37.19 | WimpMan | Actually you can have more calls than channels. |
| 09:37.23 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 09:37.37 | SwK | WimpMan, dont confuse the boy ;) |
| 09:37.38 | joobie | hey guys.. anyone know what ULL stands for? to do with ISDN |
| 09:37.54 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 09:38.05 | *** part/#asterisk BeeBuu (n=beebuu@218.13.97.202) |
| 09:38.07 | WimpMan | phpboy: In that case there's either something going wrong or something else is using the second channel. |
| 09:38.34 | SwK | joobie, unconditional local loop ? |
| 09:38.50 | SwK | joobie, not 100% on that tho depending on its context it could be something different hah |
| 09:39.01 | phpboy | WimpMan: I'm not calling using the group I'm calling using the specific port/channel |
| 09:39.10 | WimpMan | joobie: No. Never heard. Must be some NI speciality. |
| 09:39.26 | phpboy | WimpMan: So on a quad card, I can have 8 active calls to the PSTN? |
| 09:39.41 | WimpMan | phpboy: Good. Dialling on goups can case some trouble. |
| 09:39.50 | WimpMan | phpboy: Correct. |
| 09:39.57 | phpboy | ok |
| 09:40.09 | phpboy | WimpMan: It definitely does |
| 09:40.11 | joobie | SwK ya that's what it stands for.. duno what it means tho in relation to ISDN |
| 09:40.18 | phpboy | misdn/1 <---- 1 is for port or channel? |
| 09:40.31 | WimpMan | phpboy: Port |
| 09:40.37 | phpboy | perhaps this is where I got confused, I understand it as port |
| 09:40.53 | WimpMan | That's correct. |
| 09:41.08 | phpboy | but then why won't it let me push two calls through port 1? |
| 09:41.10 | phpboy | :( |
| 09:41.41 | *** join/#asterisk IsUp (n=nocturne@unaffiliated/isup) |
| 09:42.12 | WimpMan | What kind of line is it? ptp/ptmp? On a telco that knows about ISDN or not? |
| 09:42.32 | phpboy | it is ISDN |
| 09:43.02 | WimpMan | That comes in quite a number of flavours. |
| 09:43.29 | phpboy | hmmm, how can I tell? |
| 09:43.38 | WimpMan | And you can get really fancy configs in some places. |
| 09:43.53 | WimpMan | Look at what you ordered? |
| 09:44.09 | phpboy | ISDN2A |
| 09:44.20 | phpboy | it does have 2 channels, but two ports on the physical device? |
| 09:44.37 | IsUp | phpboy, did you fix the 'chennel' in your zapata.conf which you told me before? |
| 09:44.43 | WimpMan | In theory you could even have one channel restricted to inbound only. Hopefully noone sells something like that, but you never know. |
| 09:45.21 | WimpMan | zapata? We're on mISDN here. |
| 09:45.31 | phpboy | IsUp: this is a diff box, this is a BRI box |
| 09:45.33 | Assimilate | Ok I have been here for almost 10 hours after my normal 8 hour day at work. I have finally migrated from asterisk 1.2.5 to 1.4.22 with the 2.0 GUI. Time to drive 30 mins home and sleep for 4hrs so I can drive back to be here before 8am as there will be issues with the people loggin in to their phones. Thanks for all the help tonight! |
| 09:45.38 | IsUp | ah ok =) |
| 09:45.44 | phpboy | IsUp: I ended up figuring out what my problem on the PRI box was |
| 09:45.52 | phpboy | zaptel bug, PCI express cards |
| 09:45.53 | WimpMan | phpboy: A BRI has two communications channels on one interface. |
| 09:45.57 | phpboy | popped in normal PCI |
| 09:46.03 | phpboy | works like a dream |
| 09:46.09 | IsUp | good for you |
| 09:46.20 | phpboy | WimpMan: yes, two physical ports as well? |
| 09:46.36 | *** part/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 09:47.14 | WimpMan | phpboy: On a ptmp line you can connect up to 8 devices. It's a bus, so actually still only one port. |
| 09:48.57 | tzafrir_laptop | (kind of like you can connect many analog phones on the same line, but have only one concurrent call) |
| 09:49.00 | WimpMan | Most NTs will have two sockets for convenience plus a set of clamps to connect a cable w/o jack for fixed installation. But it's all the same port. |
| 09:50.39 | phpboy | WimpMan: PtP I'm assuming is otherwise |
| 09:51.55 | WimpMan | Correct. On ptp you can only connect one device. |
| 09:52.06 | phpboy | WimpMan: is there a way my country's Telco could've done this |
| 09:52.14 | WimpMan | But you won't get another NT. They're all the same. |
| 09:53.05 | WimpMan | Yes. As I pointed out above, each cahnnel can be configured for inbound only, outbound only or bidirectional. |
| 09:53.17 | phpboy | I see |
| 09:54.33 | WimpMan | Alt leas on a BRI you would assume both channels to be bidirectional, but you can configure really fancy stuff. Even worse if it happened unintentinally... |
| 09:55.32 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 09:55.57 | WimpMan | In the early days PRI were delivered with 10 channels inbound only, 10 channels outbound only and 10 channels bidirectional here. But that's quite some time ago. |
| 10:01.19 | phpboy | mISDN/1/$OUTNUM <--- 2 calls |
| 10:01.22 | phpboy | correct? |
| 10:01.43 | WimpMan | correct |
| 10:10.12 | *** join/#asterisk xacatecas (n=jkroon@dsl-241-130-06.telkomadsl.co.za) |
| 10:12.28 | phpboy | WimpMan: turns out I was just being a dumbass |
| 10:13.02 | WimpMan | Bad luck :-) |
| 10:16.02 | phpboy | oh well, at least it's working the way I want it to :D |
| 10:16.22 | *** join/#asterisk af_ (n=getsmart@88-149-240-52.dynamic.ngi.it) |
| 10:16.59 | *** join/#asterisk xacatecas (n=jkroon@mail.interexcel.co.za) |
| 10:17.40 | xacatecas | hi, what are the implications of linking asterisk cdr modules against the C++ runtime? Mostly I'm looking for stl support ... |
| 10:17.48 | WimpMan | still ponders the idea of putting the interfaces into a seperate app and connect to * via iax or something. |
| 10:24.37 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 10:25.57 | *** join/#asterisk ifnotwhynot (n=davidh@196-209-94-84-tpr-esr-2.dynamic.isadsl.co.za) |
| 10:26.06 | ifnotwhynot | i am dialing from a sip extension(eyebeam) (picking up a zap trunk that is connected to a legacy pbx extension) another extension on the legacy pbx. i want to flash the fxo and send dtmf to that fxo port to transfer the call on the legacy side,, anyone is this possible? |
| 10:26.23 | *** join/#asterisk ltd (n=z@pat.transact.net.au) |
| 10:26.44 | ifnotwhynot | if that makes any sence |
| 10:26.50 | ifnotwhynot | hi ltd |
| 10:29.02 | *** join/#asterisk tuxx- (n=tuxx@nightshade.nl) |
| 10:29.04 | tuxx- | ello. |
| 10:29.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-126afc4455abefe8) |
| 10:29.39 | tuxx- | I'm trying to hook up asterisk 1.2.26.1 to an microsoft OCS server, anyone ever done this before? |
| 10:32.51 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:44.06 | ltd | hi ifnotwhynot |
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| 11:11.49 | wry | re. |
| 11:12.32 | wry | when using ExtenSpy, must one always pass the context to it or? |
| 11:25.44 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
| 11:31.52 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 11:34.47 | *** join/#asterisk ild2002 (n=ild2002@212.118.140.9) |
| 11:35.09 | ild2002 | hi all |
| 11:35.16 | ild2002 | have a good day |
| 11:36.01 | Vale-ICS | hello |
| 11:36.04 | *** join/#asterisk jareq (n=jarek@81.15.166.2) |
| 11:36.14 | ild2002 | may i ask if i can use normal voice modem with Asterisk server cuz we dont have any digium or Voip gateway in my country |
| 11:37.07 | *** join/#asterisk Shnootz (n=Hanan@bzq-219-113-98.static.bezeqint.net) |
| 11:37.09 | ild2002 | so how voice modem can worh with Asterisk to receive landline call |
| 11:37.17 | ild2002 | any help !!!!!!! |
| 11:37.25 | Vale-ICS | hmmm not sure about that one |
| 11:37.41 | ild2002 | thanks |
| 11:38.01 | Vale-ICS | what sort of modem are you referring to? |
| 11:38.15 | ild2002 | any one |
| 11:38.21 | ild2002 | us robotics |
| 11:38.24 | ild2002 | motorolla |
| 11:38.34 | ild2002 | realtech |
| 11:38.46 | ild2002 | i can use any modem |
| 11:39.03 | ild2002 | just tell me the prand name and i will buy it |
| 11:39.25 | jksM | why not buy an ATA in the first place then... |
| 11:39.49 | tzafrir_laptop | ild2002, it will work if someone iwll provide Zaptel/DAHDI drivers for it |
| 11:39.50 | ild2002 | no ata in my country |
| 11:39.56 | ild2002 | im in saudi arabia |
| 11:40.02 | tzafrir_laptop | or an other type of drivers for a different channel |
| 11:40.09 | jksM | ild2002, then import it |
| 11:40.52 | ild2002 | r there any software or driver for that ??? |
| 11:41.18 | ild2002 | one of my freind tell me there is a teleon driver for skype |
| 11:41.21 | tzafrir_laptop | there is for just one specific modem, which is not so common nowadays |
| 11:41.38 | tzafrir_laptop | why not just use SIP for VoIP? |
| 11:41.40 | jksM | ild2002, according to google, fvc distributes digium cards in saudi arabia... |
| 11:42.00 | ild2002 | it can allow me to use my land line to receive and send a call from my landline by internet |
| 11:43.27 | ild2002 | the sip is more cost for local call |
| 11:44.22 | ild2002 | i want a way or system to receive any call on my landline by internet also make a call |
| 11:44.35 | pputman | ild2002, see what jksM said: http://www.digium.com/en/ecosystem/distributors/locate.php?region=&country=SA&search=Search |
| 11:44.50 | ild2002 | my castomer call me localy on my company land line |
| 11:46.54 | *** join/#asterisk farkus___ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 11:46.56 | ild2002 | thanks i will see |
| 11:49.36 | *** join/#asterisk boch (n=fran@customer191-9.iplannetworks.net) |
| 11:50.37 | boch | could someone tellme how should i know the agents on pause trough AMI? im monitoring QueueMemberStatus for Paused: 1, but seems wrong |
| 11:56.48 | *** join/#asterisk andresmujica (n=andresmu@190.25.99.119) |
| 12:05.36 | gambler1 | Hi, is there any way we can have in enviroment variable a number of ms for the call duration? |
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| 12:17.24 | Vale-ICS | does anyone here have much experience using nvfax? |
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| 12:32.33 | jc_yyz2bkk | whats the yum package for asterisk-perl? |
| 12:40.08 | *** part/#asterisk Spirits-Sight (n=christop@c-24-218-128-159.hsd1.ma.comcast.net) |
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| 12:48.50 | Tuxguy | Are there asterisk packages for centos? I see asterisk-sound packages in yum, but not the core. |
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| 13:00.24 | synthetiq | anyone know a regex for extensions.conf that supports all ascii characters? |
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| 13:08.52 | *** join/#asterisk ZB2 (n=ZB@189.102.227.29) |
| 13:09.15 | ZB2 | good morning |
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| 13:11.04 | *** join/#asterisk tumisho (n=tumisho@196.41.8.89) |
| 13:11.20 | tumisho | hi all |
| 13:12.40 | tumisho | <PROTECTED> |
| 13:14.04 | ZB2 | if you use speech to read sms messages i think its possible |
| 13:15.12 | ZB2 | does anyone knows a good practice to prevent users of bridging ZAP channels ? |
| 13:15.37 | ZB2 | i am using this line on my extensions.conf: exten=>_[890].,1,GotoIf(${SIPPEER(${CALLERID(num)}:curcalls)}=0?block:dontblock) |
| 13:15.57 | ZB2 | where 890 are the numbers called to get externals lines |
| 13:16.29 | ZB2 | i want to prevent users of bridging external lines |
| 13:17.00 | ZB2 | caus when they do it i get two externals lines blocked and i have to restart asterisk... |
| 13:18.53 | ZB2 | The thing with the gotoif i am using is that it does only prevent two externals lines from bridging when the sip internal line receives one ext call and try to bridge with another. when it makes one ext and try to bridge with another ext i still have the problem. |
| 13:18.56 | *** part/#asterisk tumisho_ (n=tumisho@196.41.8.89) |
| 13:22.08 | ZB2 | is there a way to configure on extension.conf a hangup on all channels the sip extension is using before making a external call ? |
| 13:22.26 | ZB2 | this way i would resolve my problem |
| 13:22.42 | ZB2 | anyone alive ? |
| 13:27.10 | *** part/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
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| 13:32.13 | xacatecas | alive yes, know an answer, no. |
| 13:32.38 | *** join/#asterisk loompek (n=NoName@noname.rula.net) |
| 13:32.40 | loompek | morning |
| 13:33.19 | kaldemar | ZB2: you could take the output of show channels and use soft hangup from a script. |
| 13:34.12 | loompek | just a quick question... how to hide my number making an outgoing call via sip trunk |
| 13:34.24 | loompek | i tried with Dial(${EXTEN},,p) |
| 13:34.28 | loompek | but that didn't work |
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| 13:36.23 | guax | hello, im having a problem with MixMonitor and atended transfeers, only the first bridged channel is recorded the call record after transfer is lost, i had checked bugs.digium and it was not very helpfull |
| 13:36.26 | *** join/#asterisk redax (i=redax@r6.hu) |
| 13:36.29 | redax | hi, |
| 13:36.41 | kaldemar | loompek: you could try func CALLERPRES |
| 13:36.59 | loompek | googling |
| 13:37.01 | redax | which is better for a HFC based ISDN card and asterisk , mISDN or ZapHFC ? |
| 13:37.35 | kaldemar | loompek: don't google, enter "core show function CALLERPRES" in CLI. |
| 13:37.36 | loompek | SetCallerPres(prohib) |
| 13:37.37 | loompek | awsome :D |
| 13:38.09 | redax | with mISDN.git I got these messages: ECHOCAN: TXBUF Underrun:4096 txbuflen:64 rxcancellen:128 |
| 13:38.57 | loompek | Got SIP response 480 "No Routes Found" from... |
| 13:38.59 | loompek | bummer :S |
| 13:40.15 | *** join/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 13:40.39 | shazaum | ... |
| 13:41.51 | *** join/#asterisk SplasPood (i=jwb@jwb.sh) |
| 13:42.28 | shazaum | file ? |
| 13:43.03 | *** part/#asterisk SplasPood (i=jwb@jwb.sh) |
| 13:44.50 | *** join/#asterisk ber_ (i=brad@neu.cow.org) |
| 13:45.10 | ber_ | hey guys, i am running an asterisk box and see a lot of SIP channels (120) which appear to be zombie |
| 13:45.14 | ber_ | how would you clear them out? |
| 13:45.23 | ber_ | my other box which is a trixbox version doesnt seem to have the same problem |
| 13:45.39 | ber_ | i would ideally not have to restart the process |
| 13:45.49 | *** part/#asterisk boch (n=fran@customer191-9.iplannetworks.net) |
| 13:46.06 | guax | this is the bug: BUG 0013538: Recording stops after Transfer [status: feedback, reported by: mbit] (http://bugs.digium.com/view.php?id=13538) |
| 13:49.35 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:57.59 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 13:58.59 | Tuxguy | I am getting an error of Call from 'jimi' to extension '4000' rejected because extension not found when trying to place a call. |
| 14:03.24 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:03.41 | [TK]D-Fender | Tuxguy: Clearly you do not have a match for "4000" from the context that call is landing on in your dialplan. |
| 14:04.02 | etm124 | Tuxguy: copy and paste part of your extensions.conf where it says something like exten => 4000 |
| 14:04.12 | Tuxguy | [jimi] |
| 14:04.12 | Tuxguy | exten => 4000,3,Voicemail(44) |
| 14:05.34 | [TK]D-Fender | Tuxguy: Priority **3** |
| 14:05.54 | [TK]D-Fender | Tuxguy: Need to start at *1* |
| 14:05.58 | Tuxguy | oh |
| 14:06.32 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 14:06.37 | Tuxguy | Ok, I changed it to 1, and reloaded, and I still get the same error. |
| 14:07.02 | [TK]D-Fender | Tuxguy: Then perhaps that isn't the context that is being looked in. |
| 14:07.33 | [TK]D-Fender | Tuxguy: if thats a SIP phone calling, enable SIP DEBUG at CLI and verbose 10 and you'll see which context its looking for it in. |
| 14:08.27 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 14:08.50 | *** join/#asterisk Katty (n=asterisk@mail.copi-rite.com) |
| 14:09.43 | [TK]D-Fender | Katty: Mew. |
| 14:09.58 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:10.05 | Katty | [TK]D-Fender: mew. |
| 14:10.28 | *** join/#asterisk iamhrh (n=iamhrh@66-162-29-114.static.twtelecom.net) |
| 14:10.49 | feeds | Could someone please point me to the right direction of a guide to configure extensions.conf? |
| 14:11.02 | iamhrh | ~book |
| 14:11.03 | jbot | from memory, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:11.07 | Katty | hugs fskrotzki |
| 14:11.19 | Katty | [TK]D-Fender: talked to the doctor yet? |
| 14:11.26 | Tuxguy | I know how to enable CLI v10.. but what about SIP DEBUG? |
| 14:11.33 | fskrotzki | says morning darling |
| 14:12.00 | [TK]D-Fender | Katty: Nope... waiting for my Medicare situation to get cleared up. Trust me, as soon as I do this is #1 on my list followed up by all the dental work I put off. |
| 14:12.09 | [TK]D-Fender | feeds: ... |
| 14:12.11 | [TK]D-Fender | ~book |
| 14:12.12 | jbot | hmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:12.28 | feeds | thanks [TK]D-Fender |
| 14:12.32 | [TK]D-Fender | Tuxguy: "sip set debug". |
| 14:12.52 | Tuxguy | ah, i found it in the wiki, sorry. |
| 14:12.55 | [TK]D-Fender | Tuxguy: "sip set debug on". <- probably this in your version |
| 14:12.58 | pabelanger | Is there any information about asterisk and transcoding? IE: test cases. documentation, performace results? I'd like to get a better understanding how difference codecs and default sound files affect a system |
| 14:12.58 | hi365 | inbound call to a remote extension work fine. why would outbound calls have a one-way audio issue? |
| 14:13.36 | [TK]D-Fender | pabelanger: There are a few accounts of this on the WIKI, including one in this past 2 weeks on the Atom & G729 |
| 14:13.41 | [TK]D-Fender | pabelanger: Go look there. |
| 14:14.09 | [TK]D-Fender | hi365: todays magic word is "details" |
| 14:14.10 | Tuxguy | [TK]D-Fender: http://pastebin.ca/1254326 , here is the output from my SIP call |
| 14:14.23 | hi365 | wow, thats unusual :) |
| 14:14.40 | hi365 | sip -> server(port forwarded) -> pri |
| 14:14.42 | [TK]D-Fender | Tuxguy: Your USER is "jimi" but that is not the CONTEXT he points to : Looking for 4000 in default (domain 127.0.0.1) |
| 14:14.52 | [TK]D-Fender | Tuxguy: it is looking in [default] |
| 14:14.54 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 14:15.21 | iamhrh | I'm having some ugly issues with my T1 dying. Basically, upon startup it works fine for a period of time - then just quits completely inbound and outbound. here are my configs (zaptel & zapata) http://pastebin.ca/1254325. when I call dial, I'm using Dial(Zap/G1/1NxxXxxXxxx). |
| 14:15.33 | shazaum | <guax> this is the bug: BUG 0013538: Recording stops after Transfer [status: feedback, reported by: mbit] (http://bugs.digium.com/view.php?id=13538) |
| 14:15.40 | iamhrh | can someone point me to where I might find some more logs / info about what is going on here? |
| 14:15.47 | shazaum | guax, n00b, this is a problem of development |
| 14:17.14 | IsUp | iamhrh: 'pri debug span 1' |
| 14:17.19 | guax | shazaum: well, transfeer is a very basic feature, it should be working as well at this point =/ |
| 14:17.50 | iamhrh | getting a bunch of "sending set asynchronous balanced mode extended" messages on the cli |
| 14:17.54 | iamhrh | after that command |
| 14:18.21 | shazaum | guax, was what everyone expected |
| 14:18.23 | iamhrh | also here are the last 50 lines from /var/log/messages http://pastebin.ca/1254328 |
| 14:19.15 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 14:19.25 | fcois93 | hello all |
| 14:19.32 | Katty | herroes. |
| 14:19.42 | feeds | :D |
| 14:20.35 | fcois93 | how can I have the same CALL-ID in the 2 sides of the call ? |
| 14:21.36 | IsUp | iamhrh: i can suggest to you setup zaptel, libpri and asterisk over. and did you talk with your telco? |
| 14:21.43 | ZB2 | kaldemar Thanks for your answer, can you explain me better how could i do it ?? i should make a bash with gcron ?? |
| 14:22.01 | iamhrh | yes, i did - that's how i managed to even get it almost working |
| 14:22.17 | iamhrh | isup: here is the pri debug output when i try a call: http://pastebin.ca/1254330 |
| 14:22.33 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 14:22.33 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:22.44 | *** join/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 14:23.37 | Katty | hugs on russellb |
| 14:23.46 | fcois93 | how can I have the same CALL-ID in the 2 sides of the call ? |
| 14:23.52 | hi365 | get a room! |
| 14:23.57 | russellb | <3 |
| 14:24.03 | Katty | we have a room. |
| 14:24.07 | Katty | GET OUT OF OUR ROOM |
| 14:24.12 | hi365 | :) |
| 14:24.17 | Katty | hi365: <3 |
| 14:24.38 | hi365 | so the '3' stands for 3sum? |
| 14:24.50 | Katty | jbot: <3 |
| 14:24.51 | jbot | hmm... <3 is not >4, or not 3, or the emulation of a love symbol |
| 14:25.08 | IsUp | iamhrh: "No D-channels available!", it means your link is down or misconfigured zaptel. did you get details from your telco? |
| 14:25.15 | IsUp | framing, etc? |
| 14:25.21 | iamhrh | isup: yes the details are all from the telco |
| 14:25.35 | iamhrh | isup: and when i restart *, it all works fine :-/ |
| 14:26.16 | iamhrh | isup: i can try rebuilding zap, libpri, and * i guess |
| 14:26.33 | hi365 | russellb: care to have a look at a bug im dealing with? |
| 14:26.39 | hi365 | http://bugs.digium.com/view.php?id=12958 |
| 14:26.40 | russellb | can't right now sorry |
| 14:26.43 | hi365 | np |
| 14:26.55 | iamhrh | isup: what's the order that's supposed to go in again? libpri then zap? |
| 14:26.59 | IsUp | iamhrh: and empty your source dir. get the latest stuff. |
| 14:26.59 | hi365 | presumes russellb is 'busy' with Katty |
| 14:27.02 | IsUp | zaptel first and then libpri |
| 14:27.09 | feeds | hi365, lol |
| 14:29.22 | Katty | seanbright: WHAT ARE YOU DOING |
| 14:29.43 | Katty | hi365: psh. |
| 14:29.53 | russellb | Katty: video taping, leave him alone |
| 14:29.54 | Katty | hi365: i <3 indiscriminately. |
| 14:30.02 | Katty | russellb: :< |
| 14:30.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:30.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 14:30.04 | Tuxguy | [TK]D-Fender: ok i set context=jimi in my sip.conf for user jimi and it worked, sorta.. just getting a no dialplan error |
| 14:30.06 | Katty | lmadsen: GET OUT |
| 14:30.25 | hi365 | man, and to think drmessano was bad... |
| 14:30.49 | drmessano | GET THE EFF OUT |
| 14:30.53 | drmessano | That's proper |
| 14:30.54 | [TK]D-Fender | Tuxguy: If you want adive, pastebin is the best way... |
| 14:30.57 | Katty | EFF? |
| 14:31.01 | hi365 | there we go! |
| 14:31.10 | hi365 | eff = f (pronounced) |
| 14:31.22 | [TK]D-Fender | tugI tend to read all pastebin's regardless of the topic jsut in case there is something I might see to help a topic I'm not currently involved in. |
| 14:31.22 | Katty | ohah. |
| 14:31.41 | Tuxguy | [TK]D-Fender: No application 'dail' for extension (jimi, 4000, 1) . I guess that means I need to make a dial plan? |
| 14:31.45 | fcois93 | how can I have the same CALL-ID in the 2 sides of the call ? |
| 14:32.08 | [TK]D-Fender | Tuxguy: Have you considered spelling DIAL correctly? |
| 14:32.10 | drmessano | Suffering from schizo? |
| 14:32.21 | drmessano | You want to be you and you? |
| 14:32.42 | Tuxguy | I am dyslexic :( |
| 14:32.44 | *** join/#asterisk CrazyTux (n=brandon@user-vcaup4j.dsl.mindspring.com) |
| 14:32.44 | [TK]D-Fender | Tuxguy: unload chan_dyslexic.so |
| 14:33.10 | Tuxguy | :D |
| 14:33.14 | lmadsen | Katty: make me :D |
| 14:33.15 | shazaum | lol |
| 14:33.26 | Katty | lmadsen: now why would i go and do a crazy thing like that? |
| 14:33.30 | Katty | lmadsen: don't be riddicurus. |
| 14:33.31 | [TK]D-Fender | lmadsen: I tried, but I kept getting build errors.... |
| 14:33.41 | Katty | [TK]D-Fender: way to make me giggle. |
| 14:33.43 | [TK]D-Fender | waits for the lmadsen bug-fix to be released |
| 14:33.43 | lmadsen | Katty: I don't quite understand?! |
| 14:33.57 | lmadsen | rolls his eyes at [TK]D-Fender's nerdy joke :) |
| 14:34.00 | Katty | lmadsen: check your parser. |
| 14:34.03 | lmadsen | a build a joke?! I mean... SERIOUSLY?! |
| 14:34.12 | lmadsen | checks his parmesan |
| 14:34.27 | coppice | build error - credit unavailable |
| 14:34.28 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 14:34.35 | feeds | can't stop laughing |
| 14:34.55 | drmessano | ROFL |
| 14:34.56 | Katty | parmesan EXPIRED |
| 14:34.59 | drmessano | wow |
| 14:35.56 | drmessano | I tried "make me" here and got some error message about the target not being worth it |
| 14:36.10 | lmadsen | lol |
| 14:36.22 | lmadsen | [TK]D-Fender: look what you've started |
| 14:37.03 | |||Mad||| | Hi, all... I have a quick Asterisk problem, hopefully you can point me in the right direction |
| 14:37.16 | Katty | points left. |
| 14:37.21 | tuxx- | :-D |
| 14:37.22 | |||Mad||| | :) |
| 14:37.23 | coppice | the right direction is ----> that way |
| 14:37.39 | |||Mad||| | Good, that's where everybosy's listed |
| 14:37.41 | Katty | well aren't you punny |
| 14:38.04 | |||Mad||| | We've got an Asterisk box set up to act as a voicemail server, and it's working great for the most part |
| 14:38.13 | [TK]D-Fender | lmadsen: I'm like Michaelangelo... I jsut took the rough stone and now David is chiseling himself..... Hope you don't mind the creative license they are taking with your "privates" ;) |
| 14:38.28 | fcois93 | how can I have the same CALL-ID in the 2 sides of the call ? |
| 14:38.51 | |||Mad||| | Calls come in to the PBX and are handed over to the box over analog lines |
| 14:39.03 | *** join/#asterisk isup^ (n=nocturne@unaffiliated/isup) |
| 14:39.13 | lmadsen | fcois93: core show application dial <-- look for the 'o' flag |
| 14:39.30 | Katty | o, o, o, it's magic!!! ya know!!! |
| 14:39.49 | Katty | okay, maybe i've had a little bit too much caffeine this morning. |
| 14:39.50 | drmessano | make: *** No rule to make target `love'. Stop. |
| 14:39.52 | drmessano | :( |
| 14:39.55 | drmessano | I guess I don |
| 14:40.01 | drmessano | I guess I don't feel like makin love |
| 14:40.12 | drmessano | Paul Rodgers lied! |
| 14:40.17 | Katty | drmessano: /comfort |
| 14:40.30 | isup^ | drmessano: /ilovenattroubles |
| 14:40.33 | |||Mad||| | On occasion, when the caller dials an extension the phone rings and is answered, but a few minutes later the call goes to voicemail. |
| 14:40.47 | drmessano | I love the XKCD |
| 14:40.57 | fcois93 | lmadsen: please read my question! the same CALL-ID not the CALLER-ID : |
| 14:41.03 | drmessano | "Hey, make me a sandwich" |
| 14:41.06 | drmessano | "No way" |
| 14:41.13 | drmessano | "sudo make me a sandwich" |
| 14:41.16 | drmessano | "ok" |
| 14:41.27 | lmadsen | fcois93: it's early here |
| 14:41.31 | lmadsen | I just woke up |
| 14:41.40 | |||Mad||| | I asked our phone support people about it and they told me that the Asterisk box needs to "Release on Transfer" when it sends a call back to the PBX. |
| 14:42.14 | fcois93 | lmadsen: the -o flag qpeak about the CALLER-ID I need to have the same CALL-ID |
| 14:42.20 | |||Mad||| | So my question is, are calls normally handled that way? Or is there a setting I may need to enable to get things ot work properly? |
| 14:42.23 | lmadsen | fcois93: I get it |
| 14:42.48 | |||Mad||| | The trouble is, it's inconsistent... sometimes the call goes only a few seconds, sometimes over 20 minutes |
| 14:43.40 | fcois93 | lmadsen: when I do a tcpdump to look at a sip communication,I see that asterisk use 2 call-id. the first is for asterisk-user1 the second is for asterisk-user2. I want that asterisk use the same for the 2 users |
| 14:44.02 | lmadsen | I don't know that you can... they are two different channels |
| 14:45.00 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 14:45.02 | Assid | heya |
| 14:45.31 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 14:45.42 | fcois93 | lmadsen: I don't have that problem with openser... (I jknow it is a proxy...) |
| 14:45.51 | lmadsen | you said it |
| 14:46.06 | lmadsen | asterisk is a B2BUA... those channels are independent of each other |
| 14:46.19 | fcois93 | too bad |
| 14:47.04 | lmadsen | kinda what I was thinking |
| 14:47.04 | *** join/#asterisk mocker (i=ksexton@198.247.173.227) |
| 14:47.10 | mocker | kicks this channel bank. |
| 14:47.27 | mocker | Detect DTMF damnit! |
| 14:48.02 | coppice | are channel banks suffering a credit crunch? |
| 14:48.24 | mocker | coppice: This one is about to suffer some kind of crunch. |
| 14:48.54 | lmadsen | the jokes abound this morning |
| 14:48.55 | lmadsen | breakfast!! |
| 14:49.37 | coppice | are you having credit crunch for breakfast? |
| 14:50.59 | Katty | hehe |
| 14:51.05 | Katty | coppice = <3 |
| 14:51.10 | Assid | anyone here by chance worked with video conferencing.. even if its not asterisk based |
| 14:51.22 | Assid | must be high quality. and something for the corporate world |
| 14:51.55 | kfife | Dahdi question: zaptel.conf => system.conf? |
| 14:52.00 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 14:52.03 | kfife | zaptel.conf !=> dahdi.conf? |
| 14:52.04 | [TK]D-Fender | Assid: Polycom <- |
| 14:52.19 | Assid | [TK]D-Fender: perhaps i should describe what they want first |
| 14:52.23 | magronez | is away: nao esto |
| 14:52.41 | [TK]D-Fender | Assid: Chances are Polycom has it. |
| 14:52.50 | c4t3l | g'mornin guys |
| 14:52.51 | Assid | ok |
| 14:52.56 | Katty | Assid: indeed, polycom. |
| 14:53.14 | Katty | Assid: they don't use asterisk tho. |
| 14:53.20 | coppice | if its bugs you want, polycom has plenty of those |
| 14:53.26 | Assid | some closed source application? |
| 14:53.46 | c4t3l | oh man. the polycom XML parser used to be so crappy |
| 14:54.05 | c4t3l | made me want to smash the damn things |
| 14:54.09 | Assid | thing is.. |
| 14:54.14 | Assid | i dont think polycom is in india |
| 14:54.38 | c4t3l | i think they are based out of israel |
| 14:55.02 | c4t3l | or maybe the equipment is made there |
| 14:55.06 | *** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl) |
| 14:55.37 | coppice | polycoms are all made in china |
| 14:56.04 | pabelanger | Cool, I'm heading to China this weekend. I should pick some up. |
| 14:56.13 | WimpMan | has the impression, that most XML parsers are not XML parsers at all. |
| 14:56.14 | c4t3l | really? i have some first gen that were made in israel |
| 14:56.19 | kfife | DAHDI question: etc/zaptel.conf = etc/dahdi/system.conf? |
| 14:56.45 | xacatecas | yes |
| 14:56.57 | xacatecas | ok, now, how do I make a module in asterisk actually load?!? |
| 14:57.24 | kfife | xacatecas: Thanks. The doc has not been updated. I'm trying to do this based on infernece. kind of a PITA. |
| 14:57.38 | xacatecas | I've now racked my brain, but I still can't make my C++ module load, it's as if the __attribute__((constructor)) functions isn't being called to register the module. |
| 14:57.44 | coppice | pabelanger: I guess China's so small your bound to be near their factory |
| 14:57.52 | xacatecas | kfife, pleasure. been through that two weeks ago. |
| 14:58.41 | *** join/#asterisk feeds_ChZ (n=chatzill@85-135-244-202.adsl.slovanet.sk) |
| 14:58.55 | feeds_ChZ | ~book |
| 14:58.55 | jbot | book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 14:58.56 | *** join/#asterisk klictel (n=klictel@nat/digium/x-3f45286532560e00) |
| 14:59.02 | kfife | xacatecas: did you perchance get HPEC to work? I've got an open ticket with digium. All the doc is still 'Zapcentric' --such a shame they had to change the name. |
| 14:59.08 | klictel | good morning all |
| 14:59.20 | kfife | Morning bruce. WHeres's bruce> |
| 14:59.22 | xacatecas | kfife, HPEC? |
| 14:59.22 | kfife | ? |
| 14:59.28 | lmadsen | jbot: book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 14:59.29 | jbot | ...but book is already something else... |
| 14:59.34 | *** join/#asterisk SgtPepe (n=SgtPepe@host74-16-static.41-88-b.business.telecomitalia.it) |
| 14:59.39 | lmadsen | jbot: no, book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 14:59.40 | jbot | okay, lmadsen |
| 14:59.43 | SgtPepe | hi everyone... |
| 14:59.48 | kfife | Morning bruce |
| 15:00.56 | kfife | xacatecas: HPEC is digium's High Performance Echo Canceller - a licensed echo canceller from digium. Comes with most every new digium card now. Up to 128 taps. |
| 15:00.57 | SgtPepe | Is there anyone from Italy? |
| 15:01.19 | xacatecas | kfife, i've got a hardware echo canceller. |
| 15:01.52 | kfife | xacatecas: nice. Kind of spendy for my 8 loops |
| 15:02.12 | *** join/#asterisk johann8384 (n=johann83@intra.netlogic.net) |
| 15:02.29 | xacatecas | kfife, the software echo cancellers just didn't do the job when I started out, so I just forked out the cash and got rid of the problem. |
| 15:02.39 | Kobaz | anyone know what echo canceller audiocodes uses |
| 15:02.40 | xacatecas | cheaper than spending countless hours trying to trouble-shoot it. |
| 15:03.26 | kfife | That was a good call. My understanding is that HPEC is a good as the hardware echo canceller provided you have ample CPU to handle it. |
| 15:04.35 | kfife | xacatecas: I like the simplicity. I wish Digium had an 8 port hardware daughter board that was 1/3 the price of the regular 'up-to-24-port' board. |
| 15:05.24 | kfife | xacatecas: of course you have to screw around with getting licenses installed much like g.729. |
| 15:05.59 | *** join/#asterisk var1 (n=var1@92-236-96-224.cable.ubr20.edin.blueyonder.co.uk) |
| 15:06.37 | kfife | SgtPepe: my great, great, great, great, great, great, great, great, great, great, great, great, great, great, grandmother was 1/9th italian |
| 15:06.59 | SgtPepe | ok... so we're like brothers!!! |
| 15:07.03 | SgtPepe | ;) |
| 15:07.15 | kfife | SgtPepe:LOL |
| 15:07.58 | SgtPepe | ok... I'm sorry for the "stupid" question... but I'd like to know more about dCAP in my country... |
| 15:08.05 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:08.52 | SgtPepe | and, by the way, I always prefer hardware echo canceller than software ones... |
| 15:09.42 | SgtPepe | (and I'm sorry for my bad English) |
| 15:10.15 | kfife | SgtPepe: even for 4 and 8 port interfaces? Is it worth the money? I don't mind buying hardware. I see it as hardware is cheap, time is expensive. |
| 15:11.03 | kfife | SgtPepe: you should hear my Zulu. It's horrible. I'm the only one in the whole tribe speaking zulu with a chicago-english accent |
| 15:11.12 | [TK]D-Fender | kfife: I especially liked the ODD NUMBERED fraction.... |
| 15:11.23 | kfife | [sic] |
| 15:11.54 | SgtPepe | ahahah... I will translate Zulu :) |
| 15:13.07 | kfife | or ubbi-dubbi, the Zoom language. That is my 'mother tongue'. I learned english only later when I learned to remove all the ub's preceeding the vowels. |
| 15:13.37 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:13.40 | SgtPepe | LOL |
| 15:13.56 | kfife | anyone remember ubbi-dubbi? |
| 15:14.01 | SgtPepe | is there anyone Asterisk certified? |
| 15:16.58 | SgtPepe | ok, next one.... |
| 15:17.38 | *** join/#asterisk EI5GTB (n=04s114@87-41-27-94.ptr.edu.ie) |
| 15:17.40 | SgtPepe | is there anybody who had experiences with Snom M3 and/or Siemens Gigaset? |
| 15:17.45 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:18.20 | kfife | I've done a fair amount of research on both of them. |
| 15:18.25 | EI5GTB | hi guys, to connect my asterisk box to the pstn i need an fxs interface? |
| 15:18.40 | kfife | EI5GTB: fxo |
| 15:18.45 | EI5GTB | ktnx |
| 15:19.10 | SgtPepe | and what do you think about?! what's better? |
| 15:19.47 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 15:19.52 | kfife | EI5GTB: this is kind of dumb but I always remember it like thsi: fxO is the interface of a phOne. |
| 15:20.17 | kfife | EI5GTB: and fxS provides 'S'ervice, as in Dial tone. |
| 15:20.52 | rob0 | dial tOne ? |
| 15:20.53 | SgtPepe | kfife: LOL! |
| 15:21.00 | kfife | EI5GTB: I know what they stand for, but that's how I always remembered it. |
| 15:22.06 | kfife | rob0: crimOny you need an analog port |
| 15:22.17 | kfife | rob0: and hell yeS, I should |
| 15:22.28 | rob0 | :) |
| 15:22.44 | rob0 | loves to wreck mnemonics |
| 15:24.03 | drmessano | I remember them this way |
| 15:24.31 | drmessano | "S" is for "Shit, bought the wrong one" |
| 15:24.36 | drmessano | and O is just O |
| 15:24.48 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-e97cece39ebd958e) |
| 15:24.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:25.19 | kfife | SgtPepe: I'd buy the m3, over the siemens, but I'm buying neither. |
| 15:25.32 | *** join/#asterisk RobertLaptop (n=rmiddle@mb90736d0.tmodns.net) |
| 15:26.40 | SgtPepe | kfife: thank's... I think I'll buy Siemens + repeater... |
| 15:26.58 | SgtPepe | M3 repeater is too young for me ;) |
| 15:27.09 | kfife | SgtPepe: I'd instead buy a Polycom 5020 or 5040. They are built better, and with the new cheaper dect server, quite affordable. |
| 15:27.58 | kfife | SgtPepe: my understanding is that the siemens 'hold' music is handset-generated cheesy-ass midi tones, ala electronic musical birthday card |
| 15:28.20 | SgtPepe | Since today I haven't seen a good polycom... |
| 15:28.36 | Katty | anyone know of a good mp4 to avi or mwv converter? |
| 15:28.40 | Katty | wmv. |
| 15:28.50 | Katty | that runs on windows. |
| 15:28.59 | kfife | SgtPepe: Supposedly it can not simply give the 'called party' the appropraite MOH class. How retarded is that. |
| 15:29.40 | [TK]D-Fender | kfife: Since when does the phone generate MoH? |
| 15:29.49 | SgtPepe | ah... I didn't know this... |
| 15:30.18 | jksM | kfife, which new cheaper dect server are you referring to? |
| 15:31.09 | doug | mmm, dect. |
| 15:31.27 | SgtPepe | doug: ?! |
| 15:31.29 | kfife | SgtPepe: I could be wrong. I don't own that phone. I'm recounting a complaint that I heard on the VUC. It was a siemens, but I don't know which model. IN any case check into it. Have you seen the polycom 5020/5040? I saw it at Astricon 2008. There's no comparison in build quality to the m3, which can be a little bit like holding a small bag of potato chips against your ear. |
| 15:32.46 | *** part/#asterisk iamhrh (n=iamhrh@66-162-29-114.static.twtelecom.net) |
| 15:33.30 | kfife | jksM: It's the smaller all-in-one device. Gimmie a sec and I'll try to look up the PN. It's a few hundred bucks, instead of a grand if I remember correctly. |
| 15:33.35 | seanbright | Katty: sleeping |
| 15:34.01 | jksM | kfife, the 300 you mean? |
| 15:34.42 | kfife | bingo. http://www.polycom.com/common/documents/support/sales_marketing/products/voice/kirk_wireless_server_300.pdf |
| 15:34.51 | jksM | kfife, have you used that one yourself? |
| 15:35.14 | *** join/#asterisk Tuxguy (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 15:35.27 | kfife | Not just yet. We'll buy it beginning of next year |
| 15:35.40 | Tuxguy | Can someone hel pme with this error? http://pastebin.ca/1254384 I have set up the users in sip.conf and extensions.conf, and included those sections in the paste |
| 15:36.00 | jksM | kfife, okay, I have found various problems with it, and were hoping to exchange experiences and work-arounds with someone |
| 15:36.15 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:36.16 | kfife | what sorts of issues have you had? |
| 15:36.31 | Katty | seanbright: oh? |
| 15:36.40 | jksM | kfife, all sorts really... everything from spontaneous reboots to one-way sound (and no, not NAT-related) |
| 15:36.44 | Katty | donates pillow to seanbright's cause. |
| 15:37.05 | kfife | That sucks. An excellent datapoint. |
| 15:37.11 | *** join/#asterisk Defraz (n=T0tal@24-117-156-21.cpe.cableone.net) |
| 15:37.27 | [TK]D-Fender | Tuxguy: context=default <- what part of "you don't have a dialplan extension called that" are you not getting? |
| 15:37.30 | SgtPepe | kfife: I don't see 5020/5040... but usually I don't like polycom style... |
| 15:37.33 | jksM | kfife, I'm getting some of it fixed in the next firmware though... testing a beta firmware for them right now |
| 15:37.52 | jksM | kfife, but still, would be nice to find work around for some of the problems they cannot find / do not acknowledge |
| 15:37.56 | seanbright | i don't need a pillow, i've got my laptop |
| 15:37.57 | SgtPepe | kfife: and often I had truble with them.. |
| 15:37.58 | Tuxguy | [TK]D-Fender: Where are the dialplans located? in extensions.conf? |
| 15:38.01 | seanbright | it's all warm and coozy |
| 15:38.06 | [TK]D-Fender | Tuxguy: dial = extensions.conf |
| 15:38.11 | [TK]D-Fender | Tuxguy: dialplan = extensions.conf |
| 15:38.40 | [TK]D-Fender | Tuxguy: Each of your phones is looking in a context that does not exist |
| 15:38.55 | *** join/#asterisk blinky42 (n=sbrown@67.200.59.43) |
| 15:39.40 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-aacf254ec3ff85b3) |
| 15:39.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:39.48 | kfife | jksM: SgtPepe: Excellent information. |
| 15:40.02 | Tuxguy | Oh, so I need to make a new one, and tell it to look for a local ext first before trying to dial out over lan line? |
| 15:40.16 | SgtPepe | anyway Kirk products are the best for dect system.. |
| 15:40.56 | [TK]D-Fender | Tuxguy: You need to pay attention to where YOU told your SIP devices to look for things they dial |
| 15:41.15 | kfife | SgtPepe: when you say Kirk, you mean the the oblong dect phones that were designed by kirk before the polycom acquisition? |
| 15:41.36 | [TK]D-Fender | Tuxguy: Maybe you should consider pointing them to a place that has extensions, and maybe even extensions you want them to be able to dial... |
| 15:42.20 | Tuxguy | [TK]D-Fender: That is what I am working on . I only want an ext->ext calling, not ext->outside world |
| 15:42.49 | [TK]D-Fender | Tuxguy: Well right now you are pointing them into dead space from what you PB'd |
| 15:43.05 | [TK]D-Fender | Tuxguy: Everything they dial will be rejected. |
| 15:43.12 | SgtPepe | no... I especially mean dect servers and dect kit for registration... about phones today I've experiences only with Siemens headsets.. |
| 15:43.34 | SgtPepe | siemens phones... |
| 15:43.41 | jksM | SgtPepe, which kirk dect servers for voip have you had good experiences with? |
| 15:43.48 | Tuxguy | oh, i wasnt sure, because 1000 was working for the demo |
| 15:44.42 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:45.04 | SgtPepe | wireless server 600 |
| 15:45.09 | SgtPepe | v3 |
| 15:45.22 | jksM | SgtPepe, I have tried that too, but had major problems with it together with asterisk |
| 15:45.31 | jksM | SgtPepe, like mysterious problems affect a smaller percentage of calls |
| 15:45.36 | *** join/#asterisk aliver (n=aliver@c-71-196-147-164.hsd1.co.comcast.net) |
| 15:46.07 | Tuxguy | So, I could do something like this, [default] |
| 15:46.07 | Tuxguy | exten => 4000,1,Dial(SIP/jimi); |
| 15:46.07 | Tuxguy | exten => 4001,1,Dial(SIP/ryan); |
| 15:46.22 | [TK]D-Fender | Tuxguy: This isn't a "demo", and it isn't "now". You have pointed your devices to a context that does not exist. This means all calls have no extens they can match. |
| 15:46.36 | [TK]D-Fender | Tuxguy: Well at least that would work. |
| 15:46.53 | aliver | Is there any way to cluster an * server in such a way that SIP clients on a conference (meetme) on box A could be seamlessly transitioned over to server B without dropping them? Ala Sun Cluster & Oracle Server. |
| 15:46.59 | [TK]D-Fender | Tuxguy: I'd avoid calling it [default] however. |
| 15:47.04 | SgtPepe | jksM... I don't know... I haven't any problem with that... |
| 15:47.20 | jksM | SgtPepe, what version of asterisk are you using them with? - any special settings? |
| 15:47.28 | SgtPepe | 1.4 |
| 15:47.28 | Tuxguy | I will change it to something more specific, i was just trying to get someting set up and able to make ext -> ext calls |
| 15:47.37 | jksM | SgtPepe, which subversion of 1.4? |
| 15:47.42 | SgtPepe | 2 |
| 15:47.43 | Carlos_PHX | aliver: Why is your server failing so often as to make that an issue? |
| 15:47.44 | [TK]D-Fender | Tuxguy: and never call SIP devices "extensions" |
| 15:48.01 | [TK]D-Fender | Tuxguy: an extension is a number you dial. |
| 15:48.03 | jksM | SgtPepe, okay, 1.4.2? - hmm.. I'm using a newer asterisk version |
| 15:48.07 | aliver | Carlos_PHX it's not. I'm just asking if that is possible. servers die. it happens. |
| 15:48.09 | SgtPepe | 1.6? |
| 15:48.18 | jksM | SgtPepe, nope, still 1.4 |
| 15:48.20 | SgtPepe | or other 1.4 |
| 15:48.51 | SgtPepe | ok... my dect phones are only for internal usage... |
| 15:48.56 | giovani | aliver: all of the asterisk "cluster" stuff I've seen is just HA failover, never seen a system that works with two hot, online nodes at once |
| 15:49.02 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 15:49.11 | giovani | but, I don't really know much about it -- try looking around on voip-info.org |
| 15:49.18 | Carlos_PHX | You can have a hot cluster, but I don't know about the conference. |
| 15:49.23 | pif | I just added some modules to asterisk, can I load them without stopping the main process? |
| 15:49.24 | jksM | SgtPepe, hmm, okay - we've only seen the problems at "high loads"... i.e. multiple simultaneous calls |
| 15:49.33 | SgtPepe | and only two of them can call out |
| 15:49.37 | Carlos_PHX | However we run Asterisk in VMware, so a hardware failure wouldn't drop a call. |
| 15:49.37 | aliver | giovani Cool. That's what I thought. So, as for now, there is nothing ala CARP-for-firewalls that can move clients over without dropping their state, right? |
| 15:49.53 | *** join/#asterisk CrazyTux (n=brandon@user-vcaup4j.dsl.mindspring.com) |
| 15:50.05 | Carlos_PHX | With VMware an Asterisk failure would still happen, just prevents hardware failure downtime. |
| 15:50.28 | giovani | Carlos_PHX: any info on the hot clustering? |
| 15:50.32 | SgtPepe | ah ok... we haven't this kind of problems with dect.. |
| 15:50.34 | giovani | a feature name I can google, etc |
| 15:50.37 | Carlos_PHX | aliver: You can keep state with Asterisk realtime, but I don't know if the call stays up on a meetme. |
| 15:50.44 | jksM | SgtPepe, weird stuff :-( |
| 15:50.55 | SgtPepe | jksM sorry... |
| 15:50.55 | jksM | SgtPepe, have you tried using the repeaters with the 600v3? |
| 15:51.00 | Carlos_PHX | giovani: Sorry, not much, my partner is doing it and it's not in production yet. |
| 15:51.04 | aliver | Carlos_PHX keeping state is cool, but can you actually transfer the state, client, and all to another box? |
| 15:51.04 | SgtPepe | no |
| 15:51.09 | Carlos_PHX | Mostly based on Asterisk realtime and SRV records. |
| 15:51.52 | SgtPepe | it cover all my 3 floor without repeater.. |
| 15:52.01 | jksM | SgtPepe, okay :-) ... too bad... really were hopings that others were using those products |
| 15:52.13 | jksM | SgtPepe, we're seing weird problems when they are stressed a bit |
| 15:53.25 | SgtPepe | jksM we are lucky but we never had problems caused by massive usage.. |
| 15:54.19 | jksM | SgtPepe, it's really weird... we've ditched the 600v3's and moving to the 6000 in order to solve |
| 15:54.23 | jksM | SgtPepe, really annoying :-( |
| 15:54.36 | Carlos_PHX | aliver: The state is kept, but I actually don't know if the call stays in progress. |
| 15:54.38 | jksM | the 600v3 is really great otherwise... when it works :-) |
| 15:55.08 | Carlos_PHX | So we can definitely process another call to the device, but I don't know if the call in progress switches stream destination. |
| 15:55.14 | Carlos_PHX | I think that would be up to the phone. |
| 15:55.22 | Carlos_PHX | You can certainly do it with SER or an SBC. |
| 15:55.32 | Carlos_PHX | But then that becomes the single point of failure. |
| 15:55.57 | SgtPepe | jksM I understood... but I'm sorry, i'm not expert in this field... I can't help you because in my experience never "fight" with that.. |
| 15:56.17 | Carlos_PHX | With realtime we can process a call from the server to the device from any of the servers and it accepts the call, so it load balances and does failover. |
| 15:57.47 | SgtPepe | now I'm going to smoke a cigarette before start a big monthly back-up... :( |
| 15:57.51 | Carlos_PHX | Ok, I can confirm that calls do not stay up. There is nothing in the phone that would redirect the media stream, even though the other server knows that the state is on a call. |
| 15:58.47 | *** part/#asterisk SgtPepe (n=SgtPepe@host74-16-static.41-88-b.business.telecomitalia.it) |
| 15:58.52 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 15:59.33 | Carlos_PHX | If you want to keep calls up, if you use a true cluster with a single IP address, that would work. |
| 15:59.36 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 15:59.37 | *** part/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 15:59.47 | *** join/#asterisk lowtek (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 16:02.16 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 16:02.58 | beniwtv | hi all... It appears that my asterisk is not sending SIGHUP to my AGI script. Anyone knows if this is working in 1.4.17? |
| 16:04.24 | *** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 16:09.06 | *** join/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 16:10.21 | ibm2 | hi ,i like to integrate h264 in my asterisk 1.2 ,it's possible ? |
| 16:10.42 | [TK]D-Fender | ibm2: * can at best do passthrough on that codec only |
| 16:12.36 | *** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl) |
| 16:12.42 | hi365 | with the new sounds folder layout, where does the custom folder go? |
| 16:12.57 | hi365 | under sounds or under en/custom? |
| 16:14.41 | [TK]D-Fender | hi365: folder go under whatever folder the base goes. |
| 16:15.20 | klictel | you should be under the language |
| 16:15.49 | klictel | you'll always play custom/blablabla |
| 16:16.16 | klictel | new world order |
| 16:16.54 | hi365 | so language/custom? |
| 16:16.59 | [TK]D-Fender | hi365: and "the custom" does not mean anything. The only reason to reference it is if you create it. There is no implied standard for "custom" sound files. |
| 16:17.18 | hi365 | good point |
| 16:17.25 | klictel | i agree |
| 16:17.33 | hi365 | quickly closes his freepbx window |
| 16:17.41 | *** join/#asterisk hfb (n=hfb@96.247.65.63) |
| 16:17.50 | [TK]D-Fender | ~cluebat hi365 |
| 16:17.51 | jbot | ACTION pulls out a ClueBat (tm) and thwaps hi365. |
| 16:17.57 | [TK]D-Fender | NOONE escapes the ClueBat! |
| 16:18.06 | hi365 | OUCH! that hurts! |
| 16:19.33 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 16:22.30 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 16:22.46 | pif | newbies used to get slapped with a large trout |
| 16:22.49 | *** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 16:24.47 | *** part/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 16:25.17 | pif | tzafrir_laptop: you there? |
| 16:26.21 | tzafrir_laptop | pif, yes |
| 16:26.27 | tzafrir_laptop | ~traut |
| 16:26.57 | tzafrir_laptop | ~trout |
| 16:27.11 | pif | with vanilla 1.4.22 no more 100% cpu or crashes, i'm not sure the bristuff patch in 1.4.21.2 is kosher |
| 16:27.14 | *** join/#asterisk [netman] (n=netman@200.Red-88-25-139.staticIP.rima-tde.net) |
| 16:34.25 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 16:34.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 16:34.37 | ber_ | is there an easy way to clear out zombie SIP channels? |
| 16:34.45 | ber_ | other than reloading the asterisk process |
| 16:34.57 | *** part/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 16:35.06 | rob0 | Only a bullet to the head, or it will rise again. |
| 16:35.13 | gsiener | hi all. is there an easy way to record audio clips via a phone? e.g. set up an extension that records audio in a certain format? |
| 16:35.26 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 16:35.29 | drmessano | ./silver_bullet |
| 16:35.36 | drmessano | But that may invoke mod_porn too |
| 16:35.40 | Katty | dies. |
| 16:35.51 | Katty | i'm hungry. |
| 16:36.04 | pif | eat me |
| 16:36.20 | Katty | :< |
| 16:36.27 | drmessano | make her? |
| 16:36.38 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
| 16:37.11 | pif | make: *** No rule to make target `her'. Stop. |
| 16:37.25 | drmessano | old joke |
| 16:37.29 | drmessano | Scroll back two hours |
| 16:37.51 | beniwtv | pif: you gotta cd into the directory, then use unzip on the file :P |
| 16:39.01 | drmessano | Im sure if you unwrapped your tarball anywhere near katty she would stab you in the throat |
| 16:39.53 | pif | scratches his tarballs |
| 16:40.16 | drmessano | If not, theres 7 or 8 of us that would |
| 16:40.32 | *** join/#asterisk mascool (n=george@c-68-84-164-71.hsd1.mi.comcast.net) |
| 16:40.35 | drmessano | stands in front of katty with a veggie corndog.. half eaten.. stick showin' |
| 16:42.01 | pif | 'katty' is prolly a bald overweight male freebsd user |
| 16:42.05 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:42.44 | drmessano | I seriously doubt she uses BSD.. no one really uses BSD |
| 16:43.21 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 16:43.28 | hardwire | hi |
| 16:43.45 | Katty | pif: i'm very much female. |
| 16:43.56 | giovani | drmessano: haha ... no one uses bsd? haha |
| 16:44.09 | hardwire | down johnny boy. |
| 16:44.23 | hardwire | drmessano is a freak, he's admitted it openly. |
| 16:44.48 | drmessano | Im sure if you did a real survey, you would find of the 100,000 copies of BSD downloaded and burned over the last 7 years, only 3 copies have actually been put to use. The rest are host headers in Apache on Windows boxes, and t-shirt sales |
| 16:45.04 | giovani | drmessano: you're off your rocker |
| 16:45.24 | tzanger | hahahaha |
| 16:45.31 | hardwire | heh |
| 16:45.38 | hardwire | he may just be right. |
| 16:45.46 | hardwire | amazing |
| 16:46.15 | mascool | why would asterisk not detect when a call to voicemail was hung up ? using asterisk 1.4.19 with ztdummy and 99.995% accuracy |
| 16:46.31 | drmessano | Everyone likes the IDEA of BSD.. Like having a 7 foot ex-football player as a 24/7 home security guard.. or like the fake "Beware of Pitbull" signs on their front doors.. but actual use? No.. Have you actually tried to install BSD? Its intentionally impossible |
| 16:46.41 | [TK]D-Fender | mascool: And none of that matter. |
| 16:46.42 | hardwire | mascool: the channel just stays up? what kind of channel? |
| 16:46.48 | mascool | SIP |
| 16:46.48 | giovani | drmessano: I know many companies that use large freebsd deployments |
| 16:47.00 | [TK]D-Fender | mascool: Timing has no impact on knowing if a channel dropped and you failed to even describe the caller |
| 16:47.09 | hardwire | mascool: what kind of phone? |
| 16:47.11 | giovani | drmessano: if you are unable to install free/open/netbsd ... you need to find another profession |
| 16:47.15 | mascool | but only calls to voicemail |
| 16:47.17 | mascool | polycom phone |
| 16:47.23 | mascool | jeez [TK]D-Fender sorry |
| 16:47.27 | [TK]D-Fender | mascool: Show us |
| 16:47.29 | [TK]D-Fender | ~pb |
| 16:47.29 | 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 |
| 16:47.30 | hardwire | mascool: pastebin how your call routes to voicemail. |
| 16:47.32 | mascool | i've failed you :) |
| 16:47.35 | Carlos_PHX | The only way to get BSD without pain is to buy a Mac. |
| 16:47.37 | mascool | ok |
| 16:47.39 | drmessano | People ask.. "Why are there so few security holes in BSD?" In reality, there's not much to look for. |
| 16:47.43 | thedonvaughn | yah freebsd is the quickest and easiest install i've ever seen. openbsd install, sure a little different since u need to be able to count in 512k blocks but still.... |
| 16:47.56 | giovani | Carlos_PHX: OS X isn't really BSD ... don't get me started |
| 16:48.04 | hardwire | frelling os x |
| 16:48.13 | rob0 | has never seen the need to try BSD, but he will as soon as Linux fails him |
| 16:48.16 | thedonvaughn | all tho up until freebsd 7.0, BSD wasn't that useful to me because it's SMP scaling was just horrid. |
| 16:48.25 | Katty | wonders why everyone thinks she's an old fat male. |
| 16:48.31 | tzanger | I've never done anything with *BSD aside of installing it once in the late 90s |
| 16:48.31 | mascool | this is the dialplan: '977' => 1. VoiceMailMain(${CALLERID(NUM)}) |
| 16:48.34 | *** join/#asterisk kaii (n=kai@ciphron.de) |
| 16:48.37 | drmessano | katty: I dont think you're old |
| 16:48.37 | mascool | and not all calls to voicemail stay up |
| 16:48.42 | giovani | freebsd has some amazingly rock-solid drivers |
| 16:48.44 | tzanger | Linux has done absolutely everything I've ever needed in a Unix style OS |
| 16:48.56 | Katty | hrmmphs. |
| 16:49.00 | mascool | only a couple every now and then |
| 16:49.03 | hardwire | mascool: you silly man. pastbin as much as you can. |
| 16:49.05 | giovani | they really put a lot of effort into coding things correctly, rather than getting the support for something new in quickly |
| 16:49.14 | mascool | hardwire, i wish i had more to paste |
| 16:49.19 | pif | loves Linux since 1995, (yeah i'm old too) |
| 16:49.25 | *** join/#asterisk _joe (n=joseph@74.51.109.60) |
| 16:49.25 | hardwire | mascool: so 977 is in the same context as the sip phone? |
| 16:49.29 | thedonvaughn | pif: yah about same time here too. |
| 16:49.29 | drmessano | Really, check a FreeBSD apache install.. I too wondered if FreeBSD was some windows app.. its not. |
| 16:49.30 | mascool | yes hardwire |
| 16:49.33 | tzanger | giovani: I have heard that argument before, and it's crap. |
| 16:49.40 | [TK]D-Fender | mascool: Show us the hung call. Show us you calling it and hanging up, all with SIP debug enabled, etc. |
| 16:49.46 | Katty | drmessano: so i'm just a fat male, is that it? :P |
| 16:49.56 | giovani | tzanger: "it's crap" is one of the worst arguments I've ever heard |
| 16:50.05 | drmessano | Katty: old fat male -old, yes |
| 16:50.05 | tzanger | the quickest way to get decent hardware support is to get *something* out so anyone can hack on it, instead of having one guy slave away over bit packing and trying to emit a perfect driver on the first release. |
| 16:50.08 | giovani | right up there with "yo mamma" |
| 16:50.12 | mascool | [TK]D-Fender, I would but I can't get one to stay connected now, like I said it happens every now and then |
| 16:50.13 | _joe | hey folks, sorry for being a complete n00b, but what might a bit of dialplan code that changes caller id on an outgoing call based on what extension is dialing out look like? |
| 16:50.16 | hardwire | [TK]D-Fender: You said you wanted to make a checklist at some point? |
| 16:50.29 | Katty | [TK]D-Fender: people really think i'm a fat male?! |
| 16:50.36 | hardwire | [TK]D-Fender: maybe it should be full of "how to debug" F.A.Q. |
| 16:50.37 | pif | Katty: and hair under yo arms |
| 16:50.46 | Katty | file: people think i'm a fat male :< |
| 16:50.49 | rob0 | Katty: your meatspace attributes are irrelevant |
| 16:50.51 | tzanger | giovani: you are intentionally ignoring myu explanation |
| 16:50.58 | giovani | tzanger: nobody said that the driver just appears out of nowhere, there's a lot of collaboration, but the bsd community is tighter, and overally, the contributors are higher-quality coders, in my experience, I know a number of bsd devels for free, open, and net bsd |
| 16:51.03 | mascool | just like when going to the car dealership, I can't re-create the problem now |
| 16:51.13 | giovani | tzanger: no ... I'm not, I just responded to it |
| 16:51.15 | drmessano | I should make a movie where I venture to find the one guy who installed NetBSD.. and at the end, he admits in a stoned, burned out haze that he may have actually installed Windows Me, but he's not really sure.. |
| 16:51.15 | tzanger | giovani: you must not have much experience :-) |
| 16:51.16 | mascool | I can show you the 2 hung calls ... |
| 16:51.16 | hardwire | mascool: good.. glad we could help. |
| 16:51.18 | Deeewayne | Katty: I think you are the Easter Bunny |
| 16:51.19 | Katty | rob0: people /have/ actually met me at cluecon you know (= |
| 16:51.32 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:51.32 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:51.32 | mascool | SIP/417-b6d9ee68 numberplan-custom-1 977 1 Up VoiceMailMai 417 417 211:48:2 (None) |
| 16:51.33 | drmessano | Find some total stoner from the 60s |
| 16:51.34 | Katty | Deeewayne: creepy. |
| 16:51.47 | rob0 | Yes, and I followed a link to your blog once, so I believe you. :) |
| 16:51.49 | drmessano | "I totally like, no, I dont know man" |
| 16:51.59 | mascool | soft hangup doesn't work either |
| 16:52.12 | mascool | the only way to get rid of them is to restart asterisk |
| 16:52.16 | drmessano | "I just smoked a bowl, and that could have been Windows Me or like Redhat.. I so thought it was BSD, shit man" |
| 16:52.36 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:52.45 | mascool | any ideas why this is happening ? |
| 16:52.51 | Katty | this is all crazy talk. |
| 16:52.52 | file | Katty: :( |
| 16:53.02 | russellb | Katty: you're face is crazy talk! |
| 16:53.05 | russellb | your. |
| 16:53.06 | russellb | :( |
| 16:53.08 | russellb | grammar fail |
| 16:53.15 | drmessano | I keep burned FreeBSD CD's around so people think i'm a hard ass.. in reality, they're just labeled FreeBSD and have my anime collection on them |
| 16:53.15 | Katty | comforts russellb |
| 16:53.27 | tzanger | giovani: the popularity of linux vs bsd, especially in areas such as industrial control and clustering would argue that linux has higher performance and stability |
| 16:53.53 | Katty | russellb: should i do the come back with your mom's face? |
| 16:54.00 | Katty | russellb: or shall we just let it be? |
| 16:54.00 | pif | pops a beer and watches linux vs. freebsd |
| 16:54.01 | drmessano | FreeBSD hasn't had one security hole discovered in 11 years.. Neither has the Commodore PET.. coincidence? |
| 16:54.03 | rob0 | Whether or not I believe that Katty is female doesn't change the irrelevance of gender in IRC. |
| 16:54.10 | russellb | Katty: it's your call. |
| 16:54.20 | Katty | calls russellb's mom. |
| 16:54.24 | giovani | tzanger: unless you have a study to back that up, I wouldn't be quoting something so specific |
| 16:54.24 | russellb | ooh |
| 16:54.32 | Katty | mom: hai, mom, can you make me some cookies? |
| 16:54.42 | giovani | tzanger: I run mostly linux, I'm not a bsd-zealot, but I acknowledge its strengths |
| 16:54.44 | rob0 | cookies are RELEVANT |
| 16:54.53 | tzanger | giovani: top 500 supercomputer list? |
| 16:54.56 | Katty | this reminds me of the Matrix on Windows. |
| 16:55.01 | tzanger | I mean bsd doesn't even have the title of most ported unix os anymore |
| 16:55.29 | drmessano | Hmm, and DOS 6.22 has had as many security patches released as FreeBSD in the last 10 years |
| 16:55.36 | drmessano | Coincidence? |
| 16:56.05 | pif | invoking DOS ends any argument |
| 16:56.08 | hardwire | Katty: call the banker! |
| 16:56.35 | tzanger | heh |
| 16:56.44 | tzanger | it's the godwin of operating systems |
| 16:56.54 | *** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 16:57.31 | drmessano | DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS DOS |
| 16:57.31 | drmessano | Theres your 50 DOS salute |
| 16:57.39 | drmessano | Do I win now? |
| 16:57.59 | tzafrir_laptop | DoS-s drmessano |
| 16:58.00 | pif | *general protection fault* |
| 16:58.57 | drmessano | I would love to port Asterisk to DOS.. but then like, I wouldnt see sunlight for years |
| 16:59.11 | rob0 | Uno DOS, dos DOS |
| 16:59.11 | mvanbaak | drmessano: use cygwin |
| 16:59.43 | drmessano | I would need cygdos |
| 16:59.45 | drmessano | lol |
| 16:59.48 | ajohnson | GAHHH ffs I hate DAHDI |
| 16:59.50 | mascool | hey guys take a look here: http://pastebin.ca/1254447 |
| 16:59.57 | mascool | I found something about one call in the logs |
| 17:00.01 | ajohnson | Why did we lose our make menuselect |
| 17:00.02 | drmessano | ajohnson: Tell us how you really feel |
| 17:00.05 | denon | who's your DAHDI |
| 17:00.13 | mascool | seems like voicemail can't read the password and the channel never hangs up |
| 17:00.31 | *** part/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 17:00.39 | *** part/#asterisk ibm2 (n=Administ@196.203.192.179) |
| 17:00.49 | *** join/#asterisk jjshoe (i=jjshoe@cpe-76-175-157-237.socal.res.rr.com) |
| 17:05.11 | mascool | I found the second call also: http://pastebin.ca/1254451 |
| 17:05.29 | mascool | hardwire, [TK]D-Fender ? |
| 17:06.15 | hardwire | mascool: add in the bits about the sip call.. and the sip call hanging up |
| 17:06.38 | mascool | you mean the full sip debug ? |
| 17:06.49 | hardwire | that would be handy.. but no |
| 17:07.00 | hardwire | just grep the logs for SIP/316-b6a3f300 |
| 17:07.06 | mascool | k |
| 17:07.30 | [TK]D-Fender | mascool: and show us the call still being up |
| 17:07.44 | mascool | it's exactly what I've pasted on pastebin |
| 17:08.40 | *** join/#asterisk synchris (n=synchris@athedsl-4380083.home.otenet.gr) |
| 17:08.53 | mascool | http://pastebin.ca/1254455 |
| 17:09.01 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 17:10.27 | *** join/#asterisk devilsoulblack (n=aandaluz@200.93.197.157) |
| 17:10.45 | *** join/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-12a11e3fc9af7bb0) |
| 17:10.59 | *** part/#asterisk M1s3ry (n=M1s3ry@nat/digium/x-12a11e3fc9af7bb0) |
| 17:11.33 | *** join/#asterisk ManxPower (n=manxpowe@33.sub-75-251-147.myvzw.com) |
| 17:12.51 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 17:13.03 | mascool | damn it pasted the wrong call for 316 |
| 17:13.48 | mascool | here it is: http://pastebin.ca/1254460 |
| 17:14.07 | mascool | it's all I have in the logs for SIP/316-b6d8c080 |
| 17:15.21 | mascool | shouldn't the SIP channel have a different id each time a call is made ? |
| 17:17.27 | *** join/#asterisk awk_r (n=rawk@nat/digium/x-e06991ea2c8ed48c) |
| 17:18.59 | mascool | Y/N/Maybe? |
| 17:19.10 | *** join/#asterisk joako (n=joako@adsl-074-170-252-213.sip.gnv.bellsouth.net) |
| 17:19.21 | beniwtv | hmmm... anyone knows what is the correct way of detecting when a call is hangup in an AGI? |
| 17:19.48 | joako | I use Linux on my desktop. How can I edit Asterisk sound files? I tried audacity but it always saves the files as 44100 |
| 17:20.10 | hardwire | mascool: no, it reuses UUID's |
| 17:20.22 | mascool | i see |
| 17:20.28 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:20.30 | hardwire | mascool: I'm not sure whats going on.. change priority 1 to Answer() for your voicemail extension.. |
| 17:20.42 | hardwire | I dunno if VoiceMailMain initiates an Answer or not.. |
| 17:20.50 | mascool | I'll try that |
| 17:21.31 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-186-176-190.bflony.east.verizon.net) |
| 17:21.35 | mascool | thanks hardwire |
| 17:24.34 | hardwire | I have no idea if that was the solution |
| 17:25.59 | beniwtv | maybe it was... |
| 17:26.37 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 17:27.14 | *** join/#asterisk joshaidan (n=joshaida@S01060090f8009fa6.tb.shawcable.net) |
| 17:29.17 | drmessano | Maybe he was like |
| 17:29.26 | drmessano | "Thanks for nothing, peace out bitches" |
| 17:29.36 | beniwtv | Anyway, I found out that * _does_ send SIGHUP. hmm... Now the question remains: If I use DIAL() from an AGI (which is the last command I send), and then exits, why does * execute DeadAGI() inmediately, even if if the call is not hung up? How to prevent that? Nobody using DIAL() in an AGI? :p |
| 17:30.43 | drmessano | Too much wordmouth |
| 17:30.55 | *** join/#asterisk r0land (n=r0land@212.36.209.1) |
| 17:30.59 | r0land | hello all |
| 17:31.16 | drmessano | H Roland, I love your synthesizers |
| 17:31.16 | r0land | hope someone could help me out with a prob am facing.. |
| 17:31.40 | ber_ | does anyone know how to clear out dead sip channels |
| 17:31.49 | ber_ | other than restarting asterisk |
| 17:32.09 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 17:32.33 | r0land | i have 3 lines connected to my Asterisk. 1- PSTN and two Callcentric sip accounts.. sometimes, when i make a call to my pstn, and then dial a CALLcentric line, i hear my welcoming message thats set for PSTN! why do u think that happens ? |
| 17:32.45 | drmessano | ber_: Not possible |
| 17:32.49 | awk_r | ber_, rtptimeout? |
| 17:32.51 | ber_ | ah |
| 17:32.56 | awk_r | just kidding |
| 17:33.06 | ber_ | i wonder why they are accumulating |
| 17:33.13 | ber_ | maybe it is some error in an AGI script I have |
| 17:33.13 | drmessano | ber_: If they're "zombie" channels, asterisk doesn't know they're zombies, hence them being zombies |
| 17:33.22 | ber_ | gotcha |
| 17:33.26 | drmessano | Thats like asking if asterisk can detect its hung and restart itself |
| 17:33.30 | ber_ | how can a zombie channel get created |
| 17:33.32 | drmessano | No, because its hung |
| 17:33.48 | drmessano | Nuclear fission, usually |
| 17:33.56 | ber_ | solar flares, etc |
| 17:34.05 | ber_ | well i was thinking you could issue a command to clear/reset all channels |
| 17:34.11 | drmessano | You can |
| 17:34.16 | ber_ | ah what is that |
| 17:34.23 | drmessano | But asterisk doesn't know it's there |
| 17:34.29 | drmessano | Its a ZOMBIE |
| 17:34.32 | ber_ | well sip show channels doe |
| 17:34.33 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 17:34.34 | ber_ | does |
| 17:34.41 | ber_ | so how can sip show channels detect it and asterisk cant |
| 17:34.59 | drmessano | Ok, suit yourself.. I am dont explaining that you're not gonna find some clever way of doing something that cant be done |
| 17:35.05 | drmessano | done* |
| 17:35.26 | ber_ | it doesnt seem to be causing any system resource issues |
| 17:35.33 | ber_ | so its just a superficial annoyance |
| 17:35.44 | ber_ | ill probably just restart the process periodically from cron |
| 17:36.04 | *** join/#asterisk rdgr (n=rich@82.46.0.91) |
| 17:37.11 | ber_ | how can a zombie channel get created? would it be my AGI not properly closing the channel? |
| 17:37.19 | ber_ | i would think asterisk would handle channel setup and teardown |
| 17:37.26 | ber_ | and hand an open channel to AGI |
| 17:37.55 | r0land | ber_ usualy zombies gets created due to asterisk not detecting the disconnect tone |
| 17:37.56 | *** join/#asterisk jsmith (n=njsmith@72.21.36.138) |
| 17:37.56 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 17:38.10 | jjshoe | ber_ and you know, there's things called software bugs |
| 17:38.15 | ber_ | but i go through a sip provider |
| 17:38.21 | ber_ | so there is no tone to detect |
| 17:38.23 | awk_r | jjshoe nevar! |
| 17:38.34 | ber_ | well asterisk has been pretty good about low bugginess |
| 17:38.42 | ber_ | i always assume I am the cuase of the problems not the program |
| 17:38.42 | drmessano | coughs |
| 17:38.43 | drmessano | Oh? |
| 17:38.54 | r0land | ber_ it still happening iwth me on my Asterisk to sipura pstn line. i've set cron to restart asterisk every night at 12:01 am, as well as emailing me whenever a channel is opened for more than 30 min that way i could check if its a zombie or not |
| 17:39.20 | drmessano | ber_ which version of asterisk? |
| 17:39.23 | ber_ | 1.2.13 |
| 17:39.47 | ber_ | or 1.2.17 depending on the box |
| 17:40.37 | drmessano | So you're using 2 year old code? |
| 17:40.43 | drmessano | No |
| 17:40.50 | drmessano | Not quite 2 years |
| 17:41.07 | drmessano | yeah, you may want to update at some point.. a lot of those sort of issues have been resolved |
| 17:41.51 | mocker | w/ new issues along the way. :) |
| 17:42.02 | ber_ | lets put it this way |
| 17:42.06 | drmessano | The last few months things have been stable |
| 17:42.08 | ber_ | i pay for trixbox call center edition for something |
| 17:42.13 | ber_ | and they use 1.2.17 |
| 17:42.16 | drmessano | lol |
| 17:42.19 | ber_ | it has been flawless stable |
| 17:42.25 | drmessano | Ok then |
| 17:42.29 | ber_ | i have used all kinds of different switches and ACD for call center |
| 17:42.42 | ber_ | to get something comparably stable has cost me 200k before |
| 17:42.48 | ber_ | (and featureful) |
| 17:42.51 | drmessano | This convo is losing it's IQ fast |
| 17:43.27 | ber_ | sorry i cant equal your irc troll status |
| 17:43.39 | drmessano | lol |
| 17:43.45 | jjshoe | actually by engauging his negative comments you are equaling his troll status |
| 17:43.46 | jjshoe | *shrug* |
| 17:43.47 | drmessano | Yeah, because none of what i am saying makes sense |
| 17:44.21 | drmessano | You're using OLD asterisk code and inquiring about problems that could well be fixed, or could be added to by the package choice you made |
| 17:44.32 | jblack | jjshoe: And by engaging drmessano's engaging of the trolls negative comments, you are equalling his troll status too. |
| 17:44.39 | jjshoe | jblack indeed I am |
| 17:44.46 | ber_ | i am complaining abut one thing which is essentially superficial |
| 17:44.48 | ber_ | and which i can deal with |
| 17:44.54 | jblack | Oh my god! Trollism is a communicable disease! |
| 17:44.57 | drmessano | Frankly, if you're gonna use Fonality code, you need to complain to them |
| 17:45.00 | jjshoe | ber_ which itsp are you using? |
| 17:45.07 | ber_ | I use global crossing |
| 17:45.11 | jjshoe | who? |
| 17:45.12 | jjshoe | yuck |
| 17:45.14 | ber_ | i dont use anyone that calls themselves an ITSP |
| 17:45.19 | ber_ | why yuck |
| 17:45.25 | ber_ | they are a huge carrier |
| 17:46.04 | ber_ | I also use XO for some US stuff |
| 17:46.10 | ber_ | they are ok |
| 17:46.23 | ber_ | i stay away from L3, they are difficult to deal with |
| 17:46.45 | ZB2 | ok |
| 17:47.00 | ber_ | anyone who buys from one of those guys i dont do business with |
| 17:47.05 | *** part/#asterisk jsmith (n=njsmith@72.21.36.138) |
| 17:47.17 | drmessano | jblack: The cycle gets even worse than that when you find the person is essentially complaining about something in asterisk that turns out to be old code, or custom binaries from a company known to produce crap code, which is borderline trolling in itself |
| 17:47.21 | drmessano | The cycle is endless |
| 17:47.23 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 17:48.20 | drmessano | jblack: and if you don't blow sunshine up their ass and tell them how pretty they look, then that starts yet another GoSub to a troll routing, which doesn't have a return |
| 17:48.27 | drmessano | It's drmessano |
| 17:48.30 | drmessano | not messano |
| 17:48.32 | drmessano | But thanks |
| 17:48.44 | drmessano | routine* |
| 17:49.22 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 17:49.49 | jblack | huh. fuckedcompany.com is fucked. |
| 17:50.08 | drmessano | I think a lot of those "zombie channel" issues were resolved in the 1.2.1x releases and about the same for 1.4 |
| 17:50.11 | jblack | When did that happen? Looks like it happened a while ago. |
| 17:50.16 | drmessano | It did |
| 17:50.18 | drmessano | Dunno when |
| 17:50.38 | drmessano | techcrunch bought them I think |
| 17:51.45 | jblack | Shame. They'd have a lot of fodder these days. |
| 17:53.22 | coppice | fuckedcompany.com is ripe for a second cuming |
| 17:56.48 | *** join/#asterisk iamhrh (n=iamhrh@66-162-29-114.static.twtelecom.net) |
| 17:57.58 | iamhrh | i'm having a problem with my t1 configuration - no matter what i use for the signalling parameter (pri_cpe or pri_net) asterisk tells me that the other end is the same thing |
| 17:58.53 | iamhrh | i messed around in zttool, hit the loop button - could that be causing it? |
| 17:59.41 | drmessano | probably |
| 18:00.08 | iamhrh | how can i undo it? |
| 18:00.17 | iamhrh | i don't see an "unloop" option :-/ |
| 18:03.47 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 18:06.58 | iamhrh | ok, rebooting the iad did it |
| 18:07.10 | iamhrh | apparently once in loop mode you have to reboot to get out of it |
| 18:11.11 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 18:11.52 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:12.03 | *** join/#asterisk VoipForces (n=courchea@67.55.25.219) |
| 18:13.05 | joako | How can I edit Asterisk sound files? |
| 18:13.13 | VoipForces | Hi, anyone knows a way to pickup an answered and ringing zap channel from a sip phone? I have an analog line that rings an analog phone (not connected to the asterisk server) this analog line is also connected to asterisk, but just go to a context that does a wait(1000). |
| 18:13.15 | [TK]D-Fender | joako: audacity <- |
| 18:13.29 | VoipForces | Joako: Try wavepad |
| 18:13.29 | joako | I tried Audacity but it always saves in 44100Hz |
| 18:13.47 | *** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu) |
| 18:13.48 | VoipForces | Wavepad is win32, but runs fine under wine |
| 18:13.58 | joako | I was about to say... |
| 18:15.02 | Qwell | joako: it "always" saves in 44k because you aren't telling it to do differently |
| 18:16.30 | joako | Where is the option to save in another frequency? the file is already 8000Hz... |
| 18:16.34 | awk_r | Qwell, well thats clearly a bug if i ever saw one. It should always save in random formats unless I say otherwise |
| 18:16.44 | joako | anyways wavepad actually plays the audio.. audacity does not |
| 18:16.45 | Qwell | awk_r: there's an option for that |
| 18:17.00 | Qwell | joako: In audacity, you click the "Play" button. |
| 18:17.27 | joshaidan | Does anyone know if "show applications" still works in 1.6? |
| 18:17.41 | Qwell | awk_r: does the _r signify that you're a reentrant version of awk? |
| 18:19.17 | awk_r | Qwell, neg r just happens to be the first letter of my name |
| 18:19.34 | awk_r | and its fun to say 'awker' |
| 18:19.58 | *** join/#asterisk feeds (n=feeds@85-135-244-202.adsl.slovanet.sk) |
| 18:20.09 | joako | Qwell: LOL Yes I do press the "PLAY" button. It shows the eq but no sound comes out of the speaker |
| 18:20.17 | Qwell | joako: you're doing it wrong :p |
| 18:20.55 | joako | joshaidan: 1.4 says "The 'show applications' command is deprecated and will be removed in a future release. Please use 'core show applications' instead." |
| 18:21.24 | iamhrh | ~book |
| 18:21.25 | jbot | i guess book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 18:21.36 | joshaidan | ah, thanks joako! |
| 18:23.00 | *** join/#asterisk Un1x (n=Shaanay@CPE001d7e1e2ddb-CM001692fa22fa.cpe.net.cable.rogers.com) |
| 18:24.44 | *** join/#asterisk qdk (n=qdk@212.27.23.141.bredband.3.dk) |
| 18:24.59 | *** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net) |
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| 18:27.52 | n8n8 | hi guys, my asterisk 1.4 has been running for a while, i noticed (before setting up the sip account in sip.conf) that my phone is failing to register but when i make an outgoing call, it works. is there a security issue there? how can i patch this up? |
| 18:29.40 | awk_r | n8n8, allowguest=no in [general] in sip.conf usually works |
| 18:29.50 | n8n8 | tnx |
| 18:31.12 | LeddyHM | is it possible to have one user with multiple registrations in sip.conf? |
| 18:31.54 | LeddyHM | i.e. [extension5] and ha a phone register with it as well as x-lite w/o any problems? |
| 18:32.19 | joako | VoipForces: wavepad works perfectly, thanks |
| 18:34.28 | VoipForces | Joako: Nice piece of software indeed. |
| 18:34.48 | VoipForces | FYI for my question about picking up an answered channel, found this: http://www.pbxfreeware.org/archives/2005/06/new_download_--.html |
| 18:35.03 | VoipForces | app_intercept: Intercept an unanswered channel: |
| 18:36.07 | doug | working on my script now... |
| 18:36.19 | *** part/#asterisk iamhrh (n=iamhrh@66-162-29-114.static.twtelecom.net) |
| 18:36.32 | doug | "If you wish to speak to Doug, press 1. If you wish to speak to Allen, press 2. If you want to hear about my latest disasterous date, press 3." |
| 18:37.26 | awk_r | 3? |
| 18:37.51 | doug | that's a number on the telephone keypad. |
| 18:38.03 | klictel | LeddyHM: so you want multiple devices using the same username/pwd to register? |
| 18:38.14 | awk_r | presses 3 and then #. |
| 18:40.31 | *** join/#asterisk jaytee (i=d8cff501@gateway/web/ajax/mibbit.com/x-d52e673c98e90693) |
| 18:41.38 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 18:41.38 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 18:41.47 | lmadsen | has anyone seen this before? [Nov 12 13:38:08] WARNING[7813]: app.c:615 __ast_play_and_record: No audio available on SIP/OfficeTrunk-083fc970?? |
| 18:42.34 | giovani | lmadsen: possibly codec negotiation problem? |
| 18:42.41 | lmadsen | this is when a call goes: analog --> asterisk_A --> asterisk_B --> asterisk_A --> analog (call cancelled due to Dial() timeout) --> asterisk_B app_voicemail |
| 18:42.41 | kfife | DAHDI question: etc/asterisk/zapata.conf Where's the DAHDI equivalent |
| 18:42.43 | giovani | try an allow=all to make sure |
| 18:42.52 | jblack | I have, but it's been a long long time, I think it was a codec problem. |
| 18:43.03 | lmadsen | giovani: I don't think so, because if I answer the call on the cell, I have audio -- I also hear the voicemail message |
| 18:43.17 | *** join/#asterisk jsmith (n=njsmith@72.21.36.138) |
| 18:43.17 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 18:43.31 | jblack | could there be a redirect in the middle of that? |
| 18:43.34 | lmadsen | odd... maybe after coming back from ringing the cell asterisk is getting confused on the codec |
| 18:43.36 | giovani | asterisk_A --> asterisk_B --> asterisk_A -- uhh, what? |
| 18:44.09 | lmadsen | giovani: call comes from asterisk A on hardware... rings a desk phone attached to asterisk B, when that doesn't answer, it sends it back to Asterisk A to ring a cell phone |
| 18:44.21 | giovani | ah, ok |
| 18:44.38 | giovani | well, googling leads me to codec negotiation issues |
| 18:44.44 | lmadsen | then the Dial() times out (on Asterisk_B), and delivers the call from Asterisk_A --> Asterisk_B to the app_voicemail on asterisk-b |
| 18:44.47 | giovani | so, I'd look into that |
| 18:44.55 | lmadsen | coolio, I'll give that a shot |
| 18:45.28 | [TK]D-Fender | kfife: /etc/asterisk/chan_dahdi.conf |
| 18:46.17 | kfife | [TK]D-Fender: Thanks. |
| 18:47.15 | *** join/#asterisk psy0nid3 (n=IT@bookit-dev.com) |
| 18:51.12 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:55.26 | *** part/#asterisk jaytee (i=d8cff501@gateway/web/ajax/mibbit.com/x-d52e673c98e90693) |
| 18:55.44 | *** join/#asterisk jaytee (i=d8cff501@gateway/web/ajax/mibbit.com/x-d52e673c98e90693) |
| 18:56.49 | Tuxguy | What is PCMU? |
| 18:56.57 | *** join/#asterisk xchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 18:57.36 | joako | On the grandstream phones? Means it's using u-Law |
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| 18:58.39 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 18:58.41 | tzafrir_laptop | Tuxguy, ulaw (g.711u) |
| 18:59.53 | lmadsen | giovani: oh interesting... seems to be some sort of reinvite issue |
| 19:00.13 | lmadsen | canreinvite=no saves the day again |
| 19:00.22 | giovani | haha ... I dunno about "saves the day" |
| 19:00.35 | giovani | you're increasing load significantly to serve 3 calls instead of one :) |
| 19:00.45 | giovani | not to mention bw increases between servers |
| 19:00.49 | giovani | and latency added |
| 19:02.41 | *** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com) |
| 19:06.17 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
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| 19:08.39 | *** part/#asterisk ManxPower (n=manxpowe@22.sub-75-250-92.myvzw.com) |
| 19:08.48 | *** join/#asterisk ManxPower (n=manxpowe@22.sub-75-250-92.myvzw.com) |
| 19:09.52 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:10.40 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 19:10.54 | magronez | is back |
| 19:14.51 | LeddyHM | klictel: yes multiple devices using the same sip registration |
| 19:15.50 | *** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net) |
| 19:15.55 | *** part/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 19:15.59 | *** join/#asterisk hardwire (n=hardwire@srv001.gandi.brutetech.com) |
| 19:16.18 | Katty | waaaaahhhhhhhhhhhhhhh!!!!!!!! |
| 19:16.19 | Katty | asplodes. |
| 19:16.28 | bijit | can someone help me with this error? Everyone is busy/congested at this time (1:0/0/1) |
| 19:17.06 | Qwell | Katty: :( |
| 19:17.36 | [TK]D-Fender | bijit: that is a meaningless generic warning. |
| 19:17.51 | [TK]D-Fender | bijit: Look at what caused it include channel debug. |
| 19:23.08 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-217-192.phlapa.east.verizon.net) |
| 19:27.11 | *** join/#asterisk feeds (n=feeds@85-135-244-202.adsl.slovanet.sk) |
| 19:31.10 | Tuxguy | tzanger: Can you convert that to mp3? |
| 19:33.42 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
| 19:36.02 | *** join/#asterisk Pagautas (n=bigman@ns.voip.ktu.lt) |
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| 19:38.32 | lmadsen | giovani: no biggie -- all on a LAN |
| 19:38.42 | lmadsen | and latency is only a few milliseconds |
| 19:38.54 | lmadsen | plus better for now since at least I get voicemails |
| 19:39.26 | mocker | Guh, so my problem with DTMF not being detected turned out to be a bad handset. |
| 19:39.38 | mocker | cries. |
| 19:39.43 | lmadsen | heh |
| 19:40.01 | mocker | Analog handsets are never supposed to go bad. :) |
| 19:40.02 | [TK]D-Fender | waits for it to collect into a puddle and holds mocker's head under... |
| 19:40.50 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 19:41.42 | Tuxguy | Anyone know a way to conver MP3 PCMU? |
| 19:41.44 | Tuxguy | to ^ |
| 19:42.19 | mocker | sox? |
| 19:42.31 | mocker | Never tried, but it's what I use for all my other conversions. :) |
| 19:43.05 | *** join/#asterisk loconut (n=blt@webtrotter.com) |
| 19:43.26 | VoipForces | asterisk -rx "file convert /tmp/file_in /tmp/file_out" |
| 19:43.32 | loconut | hello. Is there a way to override the DID on an incoming trunk so that I can make all inbound calls on a sip trunk match an incoming rule? |
| 19:44.06 | VoipForces | mocker: asterisk -rx "file convert /tmp/file_in /tmp/file_out" |
| 19:44.44 | [netman] | IMHO sox is much better |
| 19:45.06 | mocker | loconut: exten => s,1,NoOp(Here I'm starting my incoming call?) |
| 19:45.20 | VoipForces | netman: sometimes, but there is nothing better that asterisk itself to convert file format tht it will understand... |
| 19:45.52 | mocker | VoipForces: Does it just go by extension for desired output? |
| 19:46.06 | [netman] | VoipForces: I got some results with sox that I couldn't get with asterisk... |
| 19:46.12 | loconut | im using elastix/freepbx and I have two sip peers/trunks whose dids both show up as "s" and I was hoping to make an inbound route for each that wold direct each to a specific extension |
| 19:46.15 | bijit | Sorry I left. Had problems with firewall. Is this meaningless since my called said he never ended the call Channel 0/3, span 1 got hangup request, cause 17 |
| 19:46.28 | [netman] | some type of conversions |
| 19:46.39 | *** join/#asterisk Aurs (n=Ove_Aurs@apb9hb.ip.ssc.net) |
| 19:46.46 | [TK]D-Fender | loconut: FreePBX is not supported here. |
| 19:46.52 | [TK]D-Fender | ~freepbx |
| 19:46.52 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:47.10 | loconut | okie. thanks i suppose |
| 19:47.12 | *** part/#asterisk loconut (n=blt@webtrotter.com) |
| 19:47.20 | Tuxguy | VoipForces: me? |
| 19:48.10 | feeds | is all happy he can now call his * server |
| 19:48.22 | feeds | can now go to bed |
| 19:49.27 | Aurs | does anyone know the difference between 2345-11500-030.sip.ld and 2345-11500-040.sip.ld here? They are both firmware files for polycom spip 501 |
| 19:50.01 | mocker | Aurs: Looks like a difference of 10 |
| 19:50.12 | drmessano | Anyone here using CarrieXchange? |
| 19:51.02 | Aurs | mocker: aha! of course |
| 19:51.17 | [TK]D-Fender | Aurs: First thought : file size limit and had to be split in 2. |
| 19:51.30 | [TK]D-Fender | Aurs: or profitable in so doing. |
| 19:52.10 | Aurs | [TK]D-Fender: sounds unlikely... since you can use a file sip.ld which is all the other files combined... but I dont know |
| 19:52.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:53.55 | [TK]D-Fender | Aurs: see part 2 |
| 19:54.14 | [TK]D-Fender | Aurs: Maybe coming in pieces made their dev easier in some way |
| 19:56.36 | Aurs | maybe. but not my dev |
| 19:56.54 | Aurs | I have to change my plan. hehe |
| 19:58.37 | *** join/#asterisk fede2 (n=alvaro@201.192.28.246) |
| 19:58.52 | fede2 | Guys, does anyone know where to get the firmware for 3com 3102 phone to make them sip? |
| 19:59.57 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 20:00.10 | *** join/#asterisk fede2 (n=alvaro@201.192.28.246) |
| 20:00.20 | Aurs | so how should i setup the 000000000000.cfg file for polycom 501? hmm |
| 20:00.51 | *** join/#asterisk giovani (n=giovani@unaffiliated/giovani) |
| 20:00.51 | Aurs | I'm trying to avoid APP_FILE_PATH="sip.ld", because that file is 38M |
| 20:01.41 | LeddyHM | so back to my original question. Is it possible for a physical phone and a softphone to share a sip entry in sip.conf? |
| 20:01.42 | *** join/#asterisk stoffell (n=stoffell@d51A4D324.access.telenet.be) |
| 20:02.13 | LeddyHM | we are currently using 100 for the phone, and 100remote for the softphone |
| 20:02.21 | LeddyHM | the only difference is the nat enabled |
| 20:02.41 | LeddyHM | I just don't know if 2 devices can properly share 1 entry |
| 20:03.49 | Aurs | LeddyHM: as far as I know, you have to have 2 entries in sip.conf if you want both to be registered |
| 20:04.10 | LeddyHM | that's what I was afraid of |
| 20:04.26 | LeddyHM | makes sense, however I was hoping for the moon |
| 20:04.44 | Aurs | in openser, you can register several devices to the same account |
| 20:04.59 | LeddyHM | openser? |
| 20:05.01 | Aurs | but I guess that doesn't help you much, right |
| 20:05.03 | Aurs | hehe |
| 20:06.13 | LeddyHM | correct |
| 20:06.30 | *** join/#asterisk klictel (n=klictel@nat/digium/x-1566cded8048658b) |
| 20:07.13 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
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| 20:10.34 | *** part/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 20:10.38 | edibrac | i upgraded my asterisk box to 1.4 from 1.2 -- now when i check voicemail it prompts me for my extension first -- is this a setting in voicemail.conf? |
| 20:10.58 | *** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200) |
| 20:11.07 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:11.09 | [TK]D-Fender | edibrac: pastebin the call. |
| 20:11.18 | [TK]D-Fender | ~pb |
| 20:11.19 | 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 |
| 20:11.46 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 20:12.57 | jasonwoot | I'm trying out AgentCallbackLogin, but when using autologoff, the extension they are using gets paused. Issuing an "unpause" won't unpause their agentID though |
| 20:12.58 | rhousand | I am trying to setup voice mail and i am some what new to *. I am dialing my own sip extension but it does not go to voice mail even when i click ignore. My * console gives me the following error "status is 'CONGESTION'". does this sound like an issue with my dial plan? |
| 20:13.59 | edibrac | [TK]D-Fender: here's when I just hit the voicemail button on my linksys SPA941: http://pastebin.com/m334e573f |
| 20:14.58 | [TK]D-Fender | edibrac: Executing [2500@intern-ext:2] VoiceMailMain("SIP/1901-081bc268", "") in new stack <-- well you aren't telling it what BOX # to enter... |
| 20:15.26 | edibrac | [TK]D-Fender: that's just it..it used to do it automatically |
| 20:15.26 | *** join/#asterisk Vale-ICS (n=vale@user-54460330.lns1-c12.dsl.pol.co.uk) |
| 20:15.29 | [TK]D-Fender | edibrac: This is you not having read "channelvariables.txt" and "upgrade.txt" |
| 20:15.46 | *** join/#asterisk hi365_m (n=hi365@213.151.54.53) |
| 20:15.55 | *** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com) |
| 20:15.55 | [TK]D-Fender | edibrac: beacuse one of the variables you were almost certainly using in there no longer exists |
| 20:15.56 | edibrac | maybe not - it's an upgrade from 1.2 to 1.4 so.. like our great former Defense Secretary.. it looks like part of.. |
| 20:16.03 | edibrac | the unknown unknowns :) |
| 20:16.25 | Tuxguy | How can I convert PCMU -> mp3? |
| 20:16.32 | Aurs | lol |
| 20:16.43 | Aurs | [TK]D-Fender: diff <file1> <file2> |
| 20:16.45 | Aurs | no output |
| 20:16.47 | Aurs | :P |
| 20:17.07 | jasonwoot | how do I "unpause" an agent? |
| 20:17.16 | [TK]D-Fender | Aurs: Well it was a thought... go read the docs and see for yourself |
| 20:17.24 | Aurs | it is just because the polycom 501 apparently has 2 different part numbers, that can be used as variables in the config files |
| 20:17.39 | [TK]D-Fender | jasonwoot: "core show applications like pause" |
| 20:18.24 | edibrac | do some phones send the extension for you? |
| 20:18.28 | edibrac | for voicemail |
| 20:18.35 | edibrac | or that's a standard feature |
| 20:19.37 | Tuxguy | whoops |
| 20:20.11 | [TK]D-Fender | edibrac: Executing [2500@intern-ext:2] VoiceMailMain("SIP/1901-081bc268", "") in new stack <-- hello... it IS dialing 2500. That # didn't come out of thin air. its your DIALPLAN that is wrong. |
| 20:20.45 | [TK]D-Fender | edibrac: Your call to VMM is no telling it which box. And I even told you what kind of thing changed and in what docs to read about it. |
| 20:20.47 | edibrac | i know i'm talking about the phone' extension |
| 20:20.48 | *** join/#asterisk sniper_voip (n=sniper_v@87.236.144.38) |
| 20:20.50 | [TK]D-Fender | edibrac: this is NOT a guessing game. |
| 20:21.07 | [TK]D-Fender | edibrac: phones are not "extensions". |
| 20:21.10 | jasonwoot | Thanks Fender, I had no idea you could "pause" an agent |
| 20:21.34 | sniper_voip | hi all,I'm trying to install bison-devel on centos 4 but it seems that they do not support this package..Is there any equivalent package for centos 4 that I can install? |
| 20:23.07 | *** part/#asterisk loompek (n=NoName@noname.rula.net) |
| 20:23.11 | Qwell | sniper_voip: what does that have to do with Asterisk? O.o |
| 20:23.56 | Tuxguy | I meant, can I convert mp3 into PCMU ? |
| 20:23.56 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 20:24.05 | edibrac | [TK]D-Fender: yes but this linksys SPA941 let's you configure it via the web -- i figured then it might not be a dialplan issue but something on the phone. |
| 20:24.15 | sniper_voip | Qwell, I'm following the manual on http://www.asteriskguru.com/tutorials/asterisk_installation_compilation_centos.html and they are asking to install this package |
| 20:24.16 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 20:24.55 | bijit | logs on link http://pastebin.com/m2da6f595 can someone help .. Please? |
| 20:24.59 | *** mode/#asterisk [+b *!*@87.236.144.38] by [TK]D-Fender |
| 20:25.01 | *** kick/#asterisk [sniper_voip!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 20:25.22 | [TK]D-Fender | Qwell: Ban evasion from our previous discussion |
| 20:25.25 | Qwell | eh? |
| 20:26.11 | bkw_ | why was he kicked? |
| 20:26.27 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:27.50 | bkw_ | hope you're correct otherwise you come off lookin like a prick |
| 20:28.06 | jaytee | was jeev banned? I haven't been annoyed by him on over a week now. |
| 20:28.47 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 20:29.39 | *** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it) |
| 20:29.42 | bijit | ~book |
| 20:29.43 | jbot | methinks book is probably 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://tfot.leifmadsen.com or see ~buybook |
| 20:30.16 | drmessano | So it looks like these damn RTP300's use a diff provisioning schema than the other linksys stuff |
| 20:30.17 | bijit | ~buybook |
| 20:30.18 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 20:31.08 | joako | How can I play audio files to the called channel before the call is bridged? |
| 20:31.19 | Qwell | bkw_: it is correct |
| 20:31.40 | bkw_ | its still not the right way to deal with it in my opinion |
| 20:31.49 | Qwell | different issue. |
| 20:32.02 | *** join/#asterisk oh2gma (n=oh2gma@xdsl-83-150-94-231.nebulazone.fi) |
| 20:32.23 | edibrac | [TK]D-Fender: ahh my problem is htis; exten => 2500,2,VoicemailMain(${CALLERIDNUM}) ..deprecated |
| 20:32.27 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-184-199.telkomadsl.co.za) |
| 20:32.28 | edibrac | thanks |
| 20:32.38 | xacatecas | hi folks, any programmers in here? |
| 20:32.52 | [TK]D-Fender | edibrac: Congratulations on finding it. You do also see what replaces it, right? |
| 20:33.12 | edibrac | [TK]D-Fender: yeah i fixed it for other parts but not that one |
| 20:34.02 | edibrac | it was a case of, knowing the solution but not implementing it all the way :( |
| 20:34.18 | [TK]D-Fender | edibrac: Excellent, then its 1 part lesson, 1 part reminder, and 1 part vigilence. |
| 20:35.05 | edibrac | and then seeing the results of the problem and looking in the wrong place. then heading to irc and misusing technical terminology and looking like a FOOOL :) |
| 20:35.39 | [TK]D-Fender | edibrac: Ok, you can stop beating yourself up now :) |
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| 20:45.08 | *** mode/#asterisk [-b %jeev!*@*] by lmadsen |
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| 20:46.45 | Qwell | O.o |
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| 20:49.17 | lmadsen | hides |
| 20:50.36 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 20:50.54 | murdock_ut | Does HPEC work with DAHDI? |
| 20:51.01 | murdock_ut | I love acronymns. |
| 20:51.56 | Qwell | murdock_ut: yes |
| 20:52.36 | murdock_ut | Qwell: So If I have it running on 1.4 and want to move to 1.6 I can use my same license file? |
| 20:52.43 | Qwell | murdock_ut: probably |
| 20:53.39 | murdock_ut | What is the general consensus about 1.6. Is it stable enough to be put in production? |
| 20:54.24 | murdock_ut | I've got 1.6 running at home, but it is not a very busy system. |
| 20:55.17 | lmadsen | murdock_ut: only testing will determine if that is true -- asterisk is a very complex system. Depending on what you are doing with it, it may be quite stable, or it may not |
| 20:55.28 | lmadsen | that is pretty much true with any multi-versioned software |
| 20:57.47 | drmessano | I love IRC over TCPIP, it's much better than APC over ROTFL |
| 20:58.15 | murdock_ut | lmadsen: Well when 1.4 came out I think the feeling was to wait for a few point releases. I haven't heard that much negative stuff for 1.6 |
| 20:58.48 | drmessano | Of course it's not HVAC or HIPAA compliant, so I am running T.2 over BR549 to compensate for the lack of RFC6981 signalling in the ICUP stack |
| 20:59.10 | *** part/#asterisk |||Mad||| (n=mad@mail.rubbusa.com) |
| 20:59.25 | lmadsen | murdock_ut: I'd say the underlying major infrastructure changes is not as great between 1.4 and 1.6 as there was between 1.2 and 1.4. Plus the developers have gotten a lot better with more practice :) |
| 20:59.28 | lmadsen | imho |
| 21:00.40 | drmessano | I would agree with that |
| 21:00.44 | *** join/#asterisk _mary_kate_ (i=river@loreley.flyingparchment.org.uk) |
| 21:01.08 | drmessano | 1.4 to 1.6 is less traumatic, and 1.6 really does improve on the changes made in later 1.4 releases |
| 21:01.13 | drmessano | Which thus far have been awesome |
| 21:01.24 | _mary_kate_ | i'm looking for a small system to run linux+asterisk on... what sort of CPU would be required for 1-2 phones + IAX? (would something like this work? http://www.soekris.com/net5501.htm) |
| 21:01.35 | murdock_ut | There are some features that I want to implement that 1.6 has, so I'm anxious to move to it. |
| 21:01.55 | Qwell | murdock_ut: throw it on a dev box |
| 21:02.03 | drmessano | and even 1.6.1 is adding some new stuff like removing one of the timing dependencies, IIRC |
| 21:02.09 | Qwell | see how it works for you |
| 21:02.32 | drmessano | Is it conferences that won't need dahdi timing? |
| 21:02.48 | drmessano | Not sure I phrased that correctly |
| 21:03.06 | murdock_ut | Qwell: I am running 1.6 on my home phone system. Seems to be running fine. |
| 21:03.16 | Qwell | well then.. |
| 21:03.17 | RobH | _mary_kate_: I am checking the asterisk sites but I am not finding any kind of minimum requirements for running it =P |
| 21:03.28 | _mary_kate_ | RobH: fancy seeing you here |
| 21:03.46 | *** join/#asterisk vk4akp (n=Ken@c122-104-157-145.ipswc2.qld.optusnet.com.au) |
| 21:03.47 | murdock_ut | Qwell: It just isn't a very demanding role. Not like a business... Just seeing what others thought. |
| 21:03.47 | RobH | i figure i answer your question in here and thus draw the channels attention to it, i am sneaky. |
| 21:04.07 | vk4akp | Some one help me with soem Zaptel stuff please? |
| 21:04.21 | RobH | someone needs to compile a list of the crap they have run * on so we can see the cheapest solution =] |
| 21:04.23 | drmessano | Ah |
| 21:04.49 | drmessano | DAHDI won't be needed for timing, but will be needed for mixing, in the short term |
| 21:04.57 | vk4akp | ANyone have a link that explains all the options well for Chan_Dahdi ? |
| 21:05.03 | murdock_ut | I'm running it on one of these: http://www.newegg.com/Product/Product.aspx?Item=N82E16883220002 |
| 21:05.07 | lmadsen | _mary_kate_: for the requirements you just mentioned... pretty much anything beyond a P3 500 with 256megs of RAM should work fine. Yes, a soekris net5501 will be fine for that you just mentioned, but you'll need to use an embedded distro such as astlinux |
| 21:05.25 | Qwell | lmadsen: net5501 isn't x86? |
| 21:05.28 | RobH | of course, lmadsen has asnwers ;] |
| 21:05.32 | vk4akp | I'm trying ot get a PayPhone workign on a TDM400P for receiving calls. |
| 21:05.39 | RobH | answers even |
| 21:05.40 | drmessano | So no dahdi for timing in 1.6.1 (im assuming that frees IAX2 from needing dahdi) and soon conferencing wont need dahdi at all |
| 21:05.40 | lmadsen | Qwell: oh maybe it is... I thought it was geode |
| 21:05.45 | Qwell | ahh |
| 21:05.52 | Qwell | I don't know. I was asking :p |
| 21:05.56 | vk4akp | It mustent be getting the correct signalling to realise an incomming call after the ring and pickup. |
| 21:06.12 | lmadsen | 433 to 600 Mhz AMD Geode LX single chip processor with CS5536 companion chip |
| 21:06.27 | lmadsen | but... |
| 21:06.29 | lmadsen | This compact, low-power, low-cost, advanced communication computer is based on an up to 500 Mhz 586 class processor. |
| 21:06.31 | [TK]D-Fender | Geode = x*^ IIRC |
| 21:06.39 | [TK]D-Fender | x86 |
| 21:06.42 | lmadsen | heh |
| 21:06.53 | lmadsen | wonders if you still need to cross-compile for that... |
| 21:07.29 | [TK]D-Fender | lmadsen: the NET5501 was a LX-700 or LX-800 |
| 21:07.43 | [TK]D-Fender | lmadsen: I was seriously considering it as a gateway device.. |
| 21:07.47 | lmadsen | [TK]D-Fender: ok... that means absolutely nothing to me |
| 21:08.00 | drmessano | Hmm.. Says here in July that the bridging API would free conferencing from needing Dahdi and that it was almost done and needed polishing.. anyone know if this was merged? |
| 21:08.11 | Qwell | drmessano: not yet |
| 21:08.18 | lmadsen | I think file got distracted by other things |
| 21:08.19 | [TK]D-Fender | vk4akp: Gigabyte board? |
| 21:08.22 | drmessano | Is it gonna make 1.6.1? |
| 21:08.29 | lmadsen | doubtful |
| 21:08.30 | Qwell | drmessano: no, 1.6.1 is branched |
| 21:08.39 | lmadsen | oh right |
| 21:08.45 | drmessano | Duh, yeah... it's in beta |
| 21:08.54 | vk4akp | D-Fender: Its a Real Digum TDM400P |
| 21:09.12 | *** join/#asterisk BBHoss (n=bbhoss@user-24-214-210-231.knology.net) |
| 21:09.14 | drmessano | So next line would be 1.6.2 since it's a new feature, right? |
| 21:09.15 | [TK]D-Fender | vk4akp: I meant your nick :) |
| 21:09.24 | Qwell | drmessano: yeah |
| 21:09.46 | [TK]D-Fender | vk4akp: Looks almost exactly likea model I had a vew years ago |
| 21:10.01 | vk4akp | D-Fender: It's a Ham Callsign for Australia, Also yes I have a Gigabyte AMD2 4000+ M/board in my workstation. |
| 21:10.05 | drmessano | Tell file to stop getting laid and ask him what his favorite beer is.. I got a donation for the cause |
| 21:10.28 | vk4akp | How can you tell I have a Gigabyte M/Board? |
| 21:11.34 | vk4akp | So, any idea's on my payphone problem? |
| 21:11.36 | _mary_kate_ | lmadsen: thanks |
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| 21:12.23 | lmadsen | _mary_kate_: I'd probably suggest using something else though as it'll be easier to manage a standard computer system rather than an embedded solution, unless power consumption is a real issue |
| 21:12.49 | lmadsen | I'd suggest using one of those micro boards so you can compile, and do other things without all the overhead required of something like a soekris |
| 21:13.22 | lmadsen | any modern computer put out in the last few years will be adequate to handle your modest requirements though |
| 21:13.34 | _mary_kate_ | lmadsen: well, i want to avoid an entire computer, there's already a router and a switch. but something similarly sized (physically) to the soekris would work |
| 21:13.49 | murdock_ut | Check out that Eee box |
| 21:13.53 | murdock_ut | It's small. |
| 21:13.53 | lmadsen | _mary_kate_: right, but there are quite small computers now as opposed to a full tower |
| 21:14.29 | *** join/#asterisk [8none1] (n=aherbert@cerberus.franklinamerican.com) |
| 21:14.59 | lmadsen | _mary_kate_: something like: http://www.stealth.com/littlepc.htm?gclid=COy8xOHH8JYCFSTaDAodvAl3rA |
| 21:15.05 | lmadsen | just clicked on the first random google link |
| 21:16.17 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:17.49 | jasonwoot | exten => 41,1,AgentCallbackLogin(,,${CALLERIDNUM}@INTERNAL) any suggestions on what I'm doing wrong here? It's not passing the extension |
| 21:18.16 | Qwell | jasonwoot: passing the extension where? |
| 21:18.43 | jasonwoot | to 3rd step of agentcallbacklogin |
| 21:19.03 | vk4akp | So no idea's, help, links, anything? HUmm. OK.. I'm out-a here. |
| 21:19.04 | [TK]D-Fender | jasonwoot: go read "channelvariables.txt" and "upgrade.txt" |
| 21:19.05 | Qwell | what do you mean it isn't passing it? I'm not seeing extension anywhere |
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| 21:19.16 | Qwell | you probably mean to use ${CALLERID(num)} though |
| 21:19.29 | [TK]D-Fender | jasonwoot: You are calling variables that no longer exist |
| 21:19.53 | jasonwoot | lol, my middle name is deprecated |
| 21:20.09 | Qwell | Jason Deprecated Woot? What a strange name you have. |
| 21:21.00 | [TK]D-Fender | Qwell: http://xkcd.com/327/ |
| 21:21.14 | Qwell | [TK]D-Fender: indeed |
| 21:26.03 | Tuxguy | I have placed a .call file in the outgoing directory of asterisk, but the call was never placed. |
| 21:26.06 | Kobaz | anyone know why when putting someone on hold, after about 30 seconds (in the middle of a track), the phoner will hang up |
| 21:26.21 | voxter | Kobaz: your music on hold sucks! |
| 21:26.30 | Kobaz | heh |
| 21:26.37 | Kobaz | i'm getting a sip debug on it now |
| 21:26.38 | Kobaz | but umm |
| 21:26.42 | Kobaz | that's really weird |
| 21:26.43 | voxter | Ive seen that on cisco phones actually |
| 21:26.51 | kb3ien | looking to unlock my polycom phones' abality to use EFK from whom can i get/buy a license? |
| 21:26.54 | Kobaz | this is on iax2 -> aastra |
| 21:27.01 | [TK]D-Fender | Tuxguy: if the timestamp is in the future it won't, and als if the file got there by any other means than a "mv" |
| 21:28.00 | giovani | uih oh ... |
| 21:28.00 | [TK]D-Fender | *b00m* |
| 21:28.10 | [8none1] | Kobaz: are you using tt-monkeys as MOH? |
| 21:29.08 | Kobaz | http://pastebin.ca/1254660 |
| 21:29.08 | Kobaz | [8none1]: heh no |
| 21:29.08 | Tuxguy | There isnt a timestamp. |
| 21:29.08 | Kobaz | Really destroying SIP dialog '2ce0380d0c5605737e7783aa504ee60c@192.168.24.12' Method: BYE |
| 21:29.10 | [TK]D-Fender | tuxand the 2nd part? |
| 21:29.10 | Tuxguy | I mvd it from /tmp to outgiong/ |
| 21:29.29 | Tuxguy | outgoing/ |
| 21:30.06 | [TK]D-Fender | Kobaz: Contact: <sip:Unknown@192.168.24.12> <-- your server is NOT correctly set up to work behind NAT. Go read : |
| 21:30.06 | [TK]D-Fender | ~sipnat |
| 21:30.07 | 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 |
| 21:30.07 | Kobaz | [TK]D-Fender: no nat |
| 21:30.07 | Kobaz | [TK]D-Fender: this is all local lan |
| 21:30.41 | [TK]D-Fender | Kobaz: Oh drat, I went cross-eyed. |
| 21:30.51 | [TK]D-Fender | Kobaz: Sorry about that... |
| 21:30.54 | Kobaz | heh |
| 21:31.11 | Kobaz | okay, i take that back about the aastra |
| 21:31.18 | Kobaz | this is zoiper (iax) to xten (sip) |
| 21:31.28 | Tuxguy | How often does asterisk look for .call files? 1 second? Is that on by defualt? |
| 21:31.31 | [TK]D-Fender | Tuxguy: What do you see in CLI when you did the MV? |
| 21:31.38 | [TK]D-Fender | tuxnearly instant. |
| 21:31.53 | [TK]D-Fender | Tuxguy: pastebin the call file. It could be bad as well |
| 21:31.58 | Kobaz | [TK]D-Fender: it plays the music... and then cuts off right in the middle of the song |
| 21:32.16 | Tuxguy | I didn't see anything in CLI |
| 21:32.38 | Tuxguy | oh |
| 21:32.44 | Tuxguy | Permission denied |
| 21:32.51 | Tuxguy | It was burried |
| 21:33.21 | [TK]D-Fender | ok, checkout time, back later. |
| 21:33.25 | Kobaz | awww |
| 21:34.04 | Tuxguy | What permissiosn should the file have, and who should it be chown'd to? |
| 21:35.31 | etm124 | I have a client using a TDM 400P, they want to upgrade to an 800P, besides rerunning dahdi_cfg, will i really have to do much but pop the card out and the new one on? |
| 21:36.45 | lmadsen | my guess would be no |
| 21:37.17 | etm124 | im assuming that, too. but you know what they say about assuming. |
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| 21:43.05 | etm124 | Thanks lmadsen. My boss is a big fan of your book. He says to me, "I can't believe Lief Madsen is in that channel answering our dumb questions. That is awesome." |
| 21:43.25 | lmadsen | lol |
| 21:43.26 | etm124 | s/Lief/Leif |
| 21:43.46 | lmadsen | oh don't worry, I ask just as many dumb questions as I answer, if not more :) |
| 21:43.48 | x86 | how do you get such a cool name as Leif in the first place? |
| 21:43.55 | etm124 | :) |
| 21:44.08 | lmadsen | x86: my dad |
| 21:44.09 | jasonwoot | can i ask another deprecated question? exten => 40,1,AgentCallbackLogin(,,#) should log the agent off, but I'm getting Extension '#' is not valid for automatic login of agent '501' |
| 21:44.19 | x86 | lmadsen: pretty cool guy :) |
| 21:44.35 | lmadsen | x86: it's also a danish name, and my late grandfather was from denmark :) |
| 21:44.42 | x86 | ah |
| 21:44.59 | lmadsen | doesn't speak a word of dansk though |
| 21:45.11 | murdock_ut | lmadsen: Ya, good book by the way. I've got both editions. When is version 3 coming out. |
| 21:45.15 | murdock_ut | :) |
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| 21:46.43 | x86 | wow |
| 21:46.44 | lmadsen | ummm... not for a while... need to find the motivation/time to work on books again... |
| 21:46.46 | lmadsen | I started a little bit on the cookbook, but then got derailed |
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| 21:47.26 | hardwire | hai |
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| 21:47.26 | x86 | ok can someone give me a direct download link for 1.4.22? |
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| 21:47.27 | SkramX | Hi all |
| 21:47.42 | x86 | links fails with the damn redirect (since digium has to use some script to count downloads with a redirect, instead of just parsing the logs for some reason) |
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| 21:48.03 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 21:48.04 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 21:48.04 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 21:48.04 | *** join/#asterisk frozty_sa (n=froztbyt@unaffiliated/frozt01100101) |
| 21:48.04 | *** join/#asterisk getsql08 (n=getsql05@173.101.136.54) |
| 21:48.04 | *** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi) |
| 21:48.04 | *** join/#asterisk zamba (i=marius@sveigde.hih.no) |
| 21:48.04 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 21:48.04 | *** join/#asterisk tmiw (i=mooneer@voldemort.lifeafterking.org) |
| 21:48.04 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 21:48.04 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 21:48.06 | x86 | and lynx wont render asterisk.org |
| 21:48.06 | bkruse | x86: Lol wow man, let me do that for you. |
| 21:48.30 | x86 | bkruse: ;) |
| 21:48.30 | bkruse | x86: http://downloads.digium.com/pub/asterisk/releases/asterisk-1.4.22.tar.gz |
| 21:48.30 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 21:48.30 | Tuxguy | What is the best way to record a gsm file? |
| 21:48.30 | x86 | bkruse: can you make asterisk.org more links/lynx friendly? :) |
| 21:48.30 | bkruse | Ya, that gets annoying for me too x86! But I have the path memorized now :P |
| 21:48.30 | etm124 | Tuxguy: i just make an extension and record it myself |
| 21:48.31 | lowtek | Tuxguy: With Record()? |
| 21:48.34 | SkramX | I have Cisco phones configured w/ SCCP- I want *everything* to be handled by an AGI.. can I not have: +.,1,AGI(XXXX) -- because it doesnt work :( |
| 21:48.37 | vk4akp | What effects incomming call detection / signalling on a FXS port ? |
| 21:48.41 | hardwire | anybody ever seen SIP used for something other than RTP? |
| 21:49.00 | etm124 | Tuxguy: http://www.voip-info.org/wiki/index.php?page_id=1056&tk=f9a20ff1ce2cfcaef758&comments_page=1 |
| 21:49.19 | etm124 | do a search on the page for Record(day_menu.gsm) |
| 21:49.30 | subdolus | Katty: See ya next netsplit ;) |
| 21:49.35 | Kobaz | SkramX: exten => X!,1,AGI(...) |
| 21:49.41 | subdolus | goes to find bandages |
| 21:49.42 | SkramX | ill try that |
| 21:49.44 | SkramX | thanks Kobaz |
| 21:49.50 | x86 | hardwire: sure, messaging, just not with asterisk ;) |
| 21:49.52 | Katty | scowls at subdolus |
| 21:50.05 | x86 | bkruse: thanks |
| 21:50.12 | hardwire | x86: I'm thinking |
| 21:50.13 | hardwire | ducks |
| 21:50.16 | x86 | bkruse: what about asterisk-addons and zaptel? |
| 21:50.16 | hardwire | of using it for VPN's |
| 21:50.39 | x86 | hardwire: that'd be dumb, but whatever ;) |
| 21:50.46 | hardwire | x86: so you say |
| 21:50.48 | x86 | just because something CAN be done, doesn't mean it should |
| 21:50.50 | hardwire | p2p vpn's are fun |
| 21:50.53 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-92e73147ce8fa7e6) |
| 21:50.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:50.54 | hardwire | x86: true nuff |
| 21:51.07 | hardwire | but I like using sip stacks to find peers |
| 21:51.18 | x86 | bkruse: ? |
| 21:51.26 | *** join/#asterisk nomadium_ (i=miguel@unaffiliated/nomadium) |
| 21:51.27 | SkramX | Kobaz: are you sure? |
| 21:51.58 | Katty | Qwell: 8 hours :> |
| 21:52.03 | SkramX | my call gets hung up right away |
| 21:52.09 | *** join/#asterisk telecos (n=sergio@84.166.219.87.dynamic.jazztel.es) |
| 21:52.18 | *** part/#asterisk nomadium_ (i=miguel@unaffiliated/nomadium) |
| 21:52.37 | *** join/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 21:53.07 | *** join/#asterisk Daejeo (n=chatzill@118.221.248.34) |
| 21:53.19 | x86 | ok can someone give me a direct download link for zaptel 1.4.12 and the newest asterisk-addons? |
| 21:53.26 | x86 | links fails with the damn redirect (since digium has to use some script to count downloads with a redirect, instead of just parsing the logs for some reason) |
| 21:53.47 | bkruse | x86: I got it |
| 21:53.53 | x86 | cool ;) |
| 21:54.03 | bkruse | Digium should be able to do that in the apache2 config, but no! :P |
| 21:54.11 | SkramX | Kobaz: ? |
| 21:54.15 | x86 | bkruse: right! :) |
| 21:54.15 | bkruse | I am not sure all the exact reasons, but I just work around it, and wget follows irt |
| 21:54.17 | bkruse | it* |
| 21:54.31 | Tuxguy | Does it have to be served as a GSM format? Can it serve as mp3? |
| 21:54.53 | x86 | yeah wget seems to work ok with it, but not links/lynx (so I can't browse to the right path, and I don't know the path off the top of my head to feed wget) |
| 21:54.53 | bkruse | x86: http://downloads.digium.com/pub/zaptel/releases/zaptel-1.4.12.1.tar.gz |
| 21:54.54 | snapper14 | x86: Try http://downloads.digium.com/pub/telephony/ as a start |
| 21:55.43 | lowtek | Tuxguy: Are you asking if Asterisk can play non-GSM formatted audio? |
| 21:56.01 | bkruse | x86: http://downloads.digium.com/pub/asterisk/releases/asterisk-addons-1.4.3.tar.gz |
| 21:56.12 | Tuxguy | lowtek: yes |
| 21:56.34 | *** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk) |
| 21:56.43 | lowtek | Tuxguy: Asterisk supports a number of codecs through transcoding. Do a "show codecs" to see what asterisk can play natively. |
| 21:56.56 | Tuxguy | ok |
| 21:57.05 | snapper14 | Can anyone post a working sipgate asterisk configuration? I've just spent the day compiling asterisk from source and although sipgate works fine if I configure it on my desk SIP phone asterisk just doesn't want to play ball. Thx |
| 21:57.32 | Tuxguy | mp3 is not in the list |
| 21:57.46 | *** join/#asterisk arpu (n=arpu@chello084114022060.14.vie.surfer.at) |
| 21:57.52 | lowtek | Tuxguy: I believe asterisk addons has an mp3 codec. |
| 21:58.08 | maxxim | hi. i'm answering to an incomming call using Asnwer, after that i'm playing a speach using PlayBack, after that i'm calling to a endpoit using Dial command. The problem is that the calling party is not hearing the ring tone during the call. Why? How can i fix it? |
| 21:58.43 | lowtek | Tuxguy: You can use sox to convert easily between various formats. |
| 21:58.56 | vk4akp | Where can I get a better description of what all teh fields do / are in Chan_Dahdi ? |
| 22:00.13 | x86 | how do i use chan_zap with 1.4.22? |
| 22:00.33 | lmadsen | you use chan_dadhi |
| 22:00.42 | hardwire | solution! |
| 22:00.59 | lmadsen | chan_dadhi in 1.4.22 will compile against latest zaptel driver |
| 22:01.11 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-33e5edeb860a0f09) |
| 22:01.11 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:01.12 | x86 | hmm ok |
| 22:01.14 | lmadsen | Zaptel-to-DAHDI.txt in your user source |
| 22:01.20 | x86 | awesome, thanks |
| 22:03.04 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust11.oxfd.cable.ntl.com) |
| 22:05.09 | *** join/#asterisk etech3 (n=chatzill@173.6.133.69) |
| 22:06.05 | Tuxguy | I tried using sox.. |
| 22:06.11 | Tuxguy | I am getting this error when trying to convert mp3 to gsm |
| 22:06.16 | Tuxguy | Do not understand format type: mp3 |
| 22:06.33 | ai-a | because you dont have mp3 decoder. |
| 22:06.46 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:06.55 | Tuxguy | Oh, I can play them though |
| 22:06.59 | ai-a | a quick 2 minutes www.google.com for "sox Do not understand format type: mp3" and you'll see why. |
| 22:07.57 | maxxim | hi. i'm answering to an incomming call using Asnwer, after that i'm playing a speach using PlayBack, after that i'm calling to a endpoit using Dial command. The problem is that the calling party is not hearing the ring tone during the call using Dial command. Why? How can i fix it? |
| 22:08.25 | Tuxguy | I have the lame installed though |
| 22:08.27 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 22:08.34 | ai-a | the calling party's phoen does the ring. |
| 22:08.46 | hardwire | maxxim: are you specifying the r flag in your Dial command? |
| 22:09.02 | ai-a | maxxim: do you mean the 'calling party' cant hear the ring. |
| 22:09.12 | ai-a | this is because your call is not playing any audio. |
| 22:09.31 | [TK]D-Fender | maxxim: Make sure you have a proper indications.conf file |
| 22:10.26 | *** part/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust11.oxfd.cable.ntl.com) |
| 22:11.50 | Katty | lmadsen: i volunteer. |
| 22:12.01 | lmadsen | heh |
| 22:12.14 | lmadsen | if only you were a few hundred KM's closer |
| 22:12.18 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:12.39 | snapper14 | Okay, I've figured out howto get an incoming call from sipgate but I cannot hear anything on the external line, any ideas? |
| 22:12.43 | maxxim | hardwire> yes, i am! the thins is, that if i'm not using the Playback action before the Dial, the ring persist. It doesn't work just if i'm usign DIal after Playback |
| 22:13.08 | hardwire | what about what [TK]D-Fender suggested? |
| 22:13.33 | maxxim | [TK]D-Fender> i hear the ring, if i'm using strait awayt the DIal command. THe problem is that i'm not hearing the ring, if i'm using Plyaback, and after that put the call using Dial command |
| 22:14.09 | [TK]D-Fender | maxxim: unload chan_brokenrecord.so |
| 22:14.23 | [TK]D-Fender | maxxim: I hinted what you should look for. Go look. |
| 22:14.31 | [TK]D-Fender | snapper14: Read up : |
| 22:14.33 | [TK]D-Fender | ~sipnat |
| 22:14.34 | 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 |
| 22:14.35 | [TK]D-Fender | ^^^ |
| 22:15.33 | maxxim | [TK]D-Fender> what do you mean by proper indications.conf ? this is mine: http://rafb.net/p/ffK6Hs45.html |
| 22:17.31 | maxxim | [TK]D-Fender> i don't have such module 'chan_brokenrecord.so'. i'm using * 1.6 |
| 22:17.42 | [TK]D-Fender | ... |
| 22:18.47 | [TK]D-Fender | maxxim: What are you dialing? |
| 22:20.23 | x86 | haha |
| 22:20.29 | x86 | chan_brokenrecord.so... |
| 22:20.35 | x86 | classic tk-d ;) |
| 22:20.36 | maxxim | [TK]D-Fender> look here , please: http://rafb.net/p/Z6xsB949.html |
| 22:21.39 | [TK]D-Fender | maxxim: Executing [808901@maximash:4] Dial("SIP/808901-0828f538", "OOH323/911,120,r") in new stack <- 1st, you are using "r". "r" = EVIL. Avoid unless absolutely necessary |
| 22:21.39 | maxxim | [TK]D-Fender> if i'm using the same dial plan but without PlayBack, the ringing tone could be heared by the calling party |
| 22:21.41 | Katty | 30 minutes till freedom! |
| 22:22.04 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 22:22.18 | maxxim | [TK]D-Fender> i was just trying everything to make it generate the ring tones :) , thanks for advice |
| 22:22.22 | etech3 | dialing a number and adding the # is that called fast dial? Can I add that to the dial plan? |
| 22:23.05 | [TK]D-Fender | etech3: that is nothing you add to the dialplan and only applies to Zaptel FXS channels |
| 22:24.24 | etech3 | outbound calls take 2-3 sec to connect on FXS 9|. as a test |
| 22:24.56 | maxxim | [TK]D-Fender> somebody had the same problem, but the post didn't finished with an result: http://lists.digium.com/pipermail/asterisk-dev/2006-January/018078.html |
| 22:25.09 | etech3 | k |
| 22:25.14 | [TK]D-Fender | etech3: What device? |
| 22:25.26 | [TK]D-Fender | maxxim: And the result of what I just told you? |
| 22:25.31 | etech3 | TDM 410P |
| 22:26.02 | *** join/#asterisk blaylock (n=blaylock@snap.helixsystems.com) |
| 22:26.06 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.2) |
| 22:27.11 | maxxim | [TK]D-Fender> i've removed the 'r' from Dial cmd. but it didn't solve the problem :( i still can't hear the ring tone... |
| 22:27.47 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 22:28.19 | *** join/#asterisk hfb (n=hfb@96.247.64.35) |
| 22:28.35 | *** join/#asterisk protocols (n=protocol@ip-88-153-209-241.unitymediagroup.de) |
| 22:28.49 | protocols | hi all |
| 22:29.12 | [TK]D-Fender | etech3: IIRC "#" will terminate your dial. |
| 22:29.25 | protocols | I have a problem with my asterisk setup behind a nat. when the server has more than one network-interface configured, I can not register sip from outside the nat |
| 22:29.43 | protocols | with one interface configured, everything works fine |
| 22:30.03 | lowtek | protocols: What version of asterisk? We use multiple nics (two inet facing, two private net facing) |
| 22:30.19 | protocols | 1.4.21 |
| 22:30.35 | etech3 | Fender That's what I mean, can I put that in the dial plan? |
| 22:30.39 | protocols | lowtek, is your asterisk-server behind a nat? |
| 22:30.59 | [TK]D-Fender | etech3: No. Its implicitly part of chan_zap |
| 22:31.03 | lowtek | protocols: Are you listening on all interfaces in sip.conf? |
| 22:31.08 | protocols | yep |
| 22:31.26 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:31.37 | lowtek | protocols: No, they are behind layer 2 firewalls. |
| 22:31.41 | [TK]D-Fender | protocols: describe your servers networks |
| 22:32.07 | protocols | asterisk <-> nat (router) <-> wan |
| 22:32.10 | Primer | Anyone know why in asterisk 1.4.22, calls from asterisk to a polycom phone registered via SIP would show calls as: sip:8885551212@1.2.3.4 where 8885551212 would be the actual callerid of a call that came in via POTS, and 1.2.3.4 is the IP of the asterisk in question? |
| 22:32.27 | Primer | in 1.4.20, this doesn't happen. I get 8885551212 |
| 22:32.37 | [TK]D-Fender | protocols: So far that shows only 1 interface on * |
| 22:32.50 | Primer | same happens for registered extension -> registered extension via asterisk. say I'm on ext 123 and I call 456, my call shows up as sip:123@1.2.3.4, not just 123, as I'd expect |
| 22:33.10 | protocols | at least 1 physical interface, but it happens when I add virtual interfaces |
| 22:33.12 | [TK]D-Fender | Primer: if * responds back from a subnet other than that of the phone. |
| 22:33.15 | lowtek | protocols: Are these multiple NICs or multiple IP's bound to a single NIC? |
| 22:33.24 | protocols | latter |
| 22:33.31 | etech3 | Fender How can I correct this 2-3 sec delay before the call terminates to the POTS line? |
| 22:33.44 | [TK]D-Fender | protocols: Show us an actual sample. Also place a call with SIP debug enabled and include your configs |
| 22:33.50 | lowtek | protocols: Ahh, it's probably your NAT device getting confused as the MAC address is the same for all interfaces. |
| 22:34.10 | lowtek | protocols: If you need multiple IP's, use multiple NICs with NAT. |
| 22:34.13 | Primer | [TK]D-Fender: this doesn't seem to be the case |
| 22:34.17 | Madkiss | uhm. i am trying to link two asterisks via IAX2 and over an openvpn-connection. the firewall is open for iax2, but in tcpdump on the gateway for one site, i see this: 23:32:55.335704 IP (tos 0xc0, ttl 63, id 28300, offset 0, flags [none], proto: ICMP (1), length: 71) 10.9.20.6 > 10.9.11.244: ICMP 10.9.20.6 udp port 4569 unreachable, length 51 -- any ideas? |
| 22:34.20 | [TK]D-Fender | etech3: there are *2* delays in effect here. |
| 22:34.57 | [TK]D-Fender | etech3: 1 is the day gettting your INPUT from the FXS. the 2nd delay is the time it takes * to DIAL the number you have chosen one te call is accepted into the fialplan in the first place. |
| 22:35.35 | [TK]D-Fender | etech3: You can fix the 1st by pressing "#" after you're done with the number. the 2nd delay is unavoidable |
| 22:35.50 | *** join/#asterisk sdaniels (n=chatzill@cpe-72-190-15-241.tx.res.rr.com) |
| 22:36.36 | etech3 | OK Thanks Fender |
| 22:37.31 | protocols | hmm ok thanks lowtek.. I will experiment with my router a little.. |
| 22:38.04 | lowtek | protocols: You could try 1:1 NAT insted of just port forwarding. |
| 22:38.19 | *** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 22:38.22 | Carlos_PHX | One power outage...how to go over on your SMS limit. |
| 22:38.23 | [TK]D-Fender | Primer: We haven't even seen a case yet. I see nothing, and trust even less until I do :) |
| 22:38.26 | protocols | you mean something like a dmz? |
| 22:38.54 | *** part/#asterisk jyfletcher (n=justin@2105ds4-ar.0.fullrate.dk) |
| 22:38.56 | lowtek | protocols: No, do 1:1 nat, then all private IP's would hear on the public IP's on port 5060. Depending on your NAT implementation, might work. |
| 22:39.16 | lowtek | protocols: What router/firewall? |
| 22:39.21 | Primer | [TK]D-Fender: You mean, a case where what's happening is what I've described, and had it NOT be the fact that the phone and asterisk not on the same subnet? |
| 22:39.31 | protocols | aah ok.. yes thats what I was thinking of.. or at least all private ips from my asterisk machine |
| 22:39.36 | protocols | dd-wrt |
| 22:39.51 | protocols | ehm.. wrt-gl that opensource router from linksys |
| 22:39.53 | lowtek | dd-wrt? What's that? |
| 22:40.01 | lowtek | Oh, dear god man, get a real firewall, lol |
| 22:40.06 | protocols | :D |
| 22:40.09 | Primer | I am 100% certain that the phones and asterisk are on the same subnet |
| 22:40.17 | Primer | yet, we are experiencing this issue |
| 22:40.35 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 22:40.49 | pcrane | Hi guys |
| 22:40.54 | pcrane | call recording question for you |
| 22:41.24 | pcrane | a.) why does asterisk (when sox is installed) merge the in and out parts of the file, but I only hear the called party? |
| 22:41.38 | pcrane | and b.) if I've got in and out files, how do I merge them in to the one file? |
| 22:41.47 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 22:42.11 | Carlos_PHX | pcrane: Are you aware you can now record both sides at once using mixmonitor? |
| 22:42.47 | pcrane | yep |
| 22:43.03 | pcrane | this stems from using the *1 to record the call |
| 22:43.07 | pcrane | (in features.conf) |
| 22:44.54 | *** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri) |
| 22:48.48 | Katty | Qwell: :> |
| 22:48.50 | Katty | Qwell: naptime! |
| 22:49.05 | Qwell | yes! |
| 22:49.13 | Qwell | hopefully I can get one :( |
| 22:49.21 | Katty | would be good :> |
| 22:49.40 | Deeewayne | passes out sleeping rugs |
| 22:49.56 | Katty | Deeewayne: sleeping on a rug is so yesterday. |
| 22:50.07 | Katty | Deeewayne: it's all about feather beds now. |
| 22:50.15 | Deeewayne | ah....I'm so lame |
| 22:50.34 | x86 | hmm, dahdi_scan sees my TDM808EF, but I'm having trouble configuring channels on it |
| 22:50.39 | *** part/#asterisk _mary_kate_ (i=river@loreley.flyingparchment.org.uk) |
| 22:50.59 | x86 | I guess it uses the wctdm24xxp driver |
| 22:51.07 | Carlos_PHX | I'd sleep on the grass if there was any in AZ |
| 22:53.17 | Qwell | x86: yes |
| 22:53.33 | x86 | Qwell: hold, i'll pastebin my config |
| 22:54.53 | x86 | Qwell: http://pastebin.ca/1254740 |
| 22:55.00 | x86 | what's wrong with my config? |
| 22:55.33 | Katty | ctrl ad |
| 22:55.35 | Qwell | if that's your entire config - a lot |
| 22:56.50 | x86 | Qwell: I only want to setup this single group of channels |
| 22:57.10 | x86 | hmm, I'm missing a group= I guess |
| 22:57.20 | x86 | I've only ever setup T1's with asterisk |
| 22:57.29 | x86 | (as far as analog is concerned) |
| 22:57.55 | protocols | [TK]D-Fender, it helped when I forwared the first virtual interface |
| 22:58.07 | protocols | e.g. I have eth0, eth0:0, eth0:1 |
| 22:58.21 | protocols | when forwarding port 5060 to eth0:0, I can register from outsited |
| 22:58.23 | x86 | Qwell: can you pastebin a config that will work? I'm sure I'm only missing a couple things? |
| 22:58.35 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:58.39 | snapper14 | <PROTECTED> |
| 22:58.44 | protocols | -d, when forwarding ports to the other ifterfaces it does not work |
| 22:59.01 | x86 | Qwell: not seeing anything in the sample config pertaining to anything but T1 interfaces and radio interfaces |
| 22:59.12 | [TK]D-Fender | snapper14: pastebin is your friend |
| 22:59.19 | x86 | well, T1, E1, BRI, PRI, and radio |
| 22:59.24 | [TK]D-Fender | snappe~pb |
| 22:59.30 | x86 | but not just a lowly 8-channel FXO card |
| 22:59.34 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 22:59.59 | snapper14 | <PROTECTED> |
| 23:00.09 | [TK]D-Fender | ~pb |
| 23:00.10 | 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 |
| 23:00.33 | lesouvage | I have a strange situation. dtmf digits seem to arive at the asterisk server in the wrong order with the result that I'm calling the wrong number. Never have seen this before. |
| 23:01.00 | lesouvage | 12:00 at night is a bad time for calling wrong numbers :-( |
| 23:02.11 | snapper14 | [TK]D-Fender http://pastebin.com/m527f5e10 |
| 23:03.19 | [TK]D-Fender | snapper14: you have skipped a very important parameter in the guide which was well commented. Go read it again |
| 23:04.46 | lesouvage | Operation panel triggers local channel for first leg and second leg by bridging the first leg to a local channel with read() to enter the number and Dial() to set up the second leg. The/n paramter is used in both local channels. |
| 23:05.13 | *** join/#asterisk CrazyTux (n=brandon@adsl-75-4-22-105.dsl.irvnca.sbcglobal.net) |
| 23:05.58 | lesouvage | Can the use of local channels cause mixing up the dtmf digits? |
| 23:07.10 | [TK]D-Fender | lesouvage: no latency exists. |
| 23:07.17 | [TK]D-Fender | lesouvage: it should be end-to-end |
| 23:08.01 | *** join/#asterisk sulan (n=sulan@89-253-95-143.customers.ownit.se) |
| 23:09.22 | doug | i must be specifying callerid wrong in my sip.conf... |
| 23:09.46 | doug | when i say 'callerid="Robert E. Lee" <8885894849>' |
| 23:09.52 | doug | the sip session only says: From: "Robert E. Lee" <sip:bob@voipprovider.com>;tag=as4f1a13f5M |
| 23:11.10 | lesouvage | [TK]D-Fender: So it could be a SIP provider issue? |
| 23:11.22 | [TK]D-Fender | lesouvage: So far could be anything |
| 23:13.02 | lesouvage | [TK]D-Fender: any suggestion on where to start? |
| 23:13.51 | lesouvage | changing dtmf mode? |
| 23:13.51 | *** join/#asterisk StephenF (n=none@198.144.201.106) |
| 23:13.51 | [TK]D-Fender | lesouvage: if you're getting scrambled digits, then its not the mode. |
| 23:13.51 | [TK]D-Fender | lesouvage: Or you wouldn't get anything |
| 23:16.08 | snapper14 | 86.4.193.46 |
| 23:16.08 | snapper14 | [TK]D-Fender I've had another look but the only setting I cld c that looked relevent was host=. Tried setting this to public and private IP but no difference |
| 23:16.34 | snapper14 | oops :-S |
| 23:17.00 | lesouvage | [TK]D-Fender: point is that I have other routines that use read() for digit input and they work ok. I configured FOP myself in a way that two local channels are bridged and both local channels set up an outgoing channel to a real phone. |
| 23:27.02 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:27.35 | lesouvage | If I enter them real slow then it works ok. |
| 23:27.37 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 23:28.29 | *** join/#asterisk jblack (n=jblack@pool-71-181-244-67.sctnpa.east.verizon.net) |
| 23:29.55 | snapper14 | [TK]D-Fender: Thanks for your help so far, giving up for the night try again tomorrow. Chow |
| 23:30.06 | *** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus) |
| 23:31.01 | [TK]D-Fender | So close... |
| 23:31.06 | *** part/#asterisk jaytee (i=d8cff501@gateway/web/ajax/mibbit.com/x-d52e673c98e90693) |
| 23:31.56 | [TK]D-Fender | Hint : IMPORTANT! phones must not be allowed to attempt to directly connect with each other |
| 23:32.01 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:33.58 | x86 | Qwell: only problem was the echo canceller thingy |
| 23:34.20 | x86 | Qwell: took out that hpec echo canceller thing and dahdi_cfg -vvvv worked fine, setup all 8 channels |
| 23:35.11 | lesouvage | I tried the VoiceChangeDial() application. Funny app, it can change the pitch of the voice of the caller interesting for kids who want to call to school and sound like there father ;-) |
| 23:35.38 | snapper14 | That close, don't give it away just yet. I'll have another look tomorrow (giggle) |
| 23:36.23 | *** join/#asterisk `paul (n=admin@125.252.70.126) |
| 23:37.33 | *** part/#asterisk snapper14 (n=snapper1@cpc3-reig1-0-0-cust301.hers.cable.ntl.com) |
| 23:37.48 | `paul | can i change a ring tone(tune) for a specific user... any clues? |
| 23:38.29 | [TK]D-Fender | 'pdepends on the phone |
| 23:39.02 | [TK]D-Fender | `paul: depends on the phone |
| 23:42.38 | *** join/#asterisk km2 (n=x@mobile-166-217-042-173.mycingular.net) |
| 23:46.57 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 23:48.42 | `paul | how bout an xlite softphone is it possible? |
| 23:48.48 | *** join/#asterisk ManxPower (n=manxpowe@212.sub-70-222-140.myvzw.com) |
| 23:49.12 | `paul | is it possible to play something right before dial() and stop it when its connected? :) |
| 23:49.45 | ManxPower | `paul: you are looking for MoH or one of the Dial options see "core show application dial" |
| 23:49.49 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-33e5edeb860a0f09) |
| 23:51.22 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 23:51.57 | [TK]D-Fender | `paul: go look in its manual |
| 23:53.43 | `paul | dial(SIP/111,35,Ttm testclass) <---right syntax? |
| 23:53.51 | ManxPower | Just remember FXO ports are considered "answered/connected" as soon as you are done dialing. |
| 23:54.02 | ManxPower | `paul: wrong syntax, look at extensions.conf.sample |
| 23:54.18 | killfill | hm.. so a 4-ports analog card cost almost the same as a PRI one?.. |
| 23:54.38 | killfill | just like 100 bucks less.. |
| 23:54.42 | ManxPower | `paul: you do not want both T and t, using both can open up a MAJOR security hole allowing people to use your PBX to make toll calls billed to you. |
| 23:54.54 | ManxPower | killfill: maybe same COST, but not same. |
| 23:55.17 | killfill | yeah.. i would guess pri's sells more... :P |