| 00:00.36 | *** join/#asterisk _370z-hsv (~E63AMG@207.98.192.200) |
| 00:01.16 | *** join/#asterisk _370z-hsv (~E63AMG@207.98.192.200) |
| 00:03.43 | *** join/#asterisk mazpe (~lesterm@c-174-61-72-8.hsd1.fl.comcast.net) |
| 00:06.53 | _370z-hsv | mazpe, its 100% dead in here |
| 00:07.14 | mazpe | ? |
| 00:07.22 | _370z-hsv | why join ,if no one is talking? |
| 00:07.28 | _370z-hsv | It's idle central in here now |
| 00:07.50 | mazpe | the asterisk vampires come alive every once in a while |
| 00:07.54 | _370z-hsv | So to kill time, im listening to DJ Khaled - take it to the head, waiting for Diablo 3 to come out, watching preview clips |
| 00:09.18 | _370z-hsv | mazpe, nice website |
| 00:09.19 | _370z-hsv | http://www.lesterarielmesa.com/ |
| 00:09.42 | _370z-hsv | https://lh6.googleusercontent.com/--qW706gtNNY/TYALrapYrgI/AAAAAAAACvY/G-256tjYot4/s1104/191065_1576522983504_1548500312_31242229_381419_o.jpg |
| 00:10.36 | _370z-hsv | Now you see the world in "SuperFocus" Lester |
| 00:11.23 | mazpe | you must be really bored… lol |
| 00:11.28 | _370z-hsv | :) |
| 00:11.46 | _370z-hsv | I have a script that has SET/Maltego integration |
| 00:11.50 | mazpe | da hell is skinny protocol |
| 00:11.57 | _370z-hsv | that auto searches people, and i can seelct if i want to view output or not |
| 00:12.04 | _370z-hsv | auto sets, and auto maltegos people on join |
| 00:13.09 | WIMPy | mazpe: Why? |
| 00:14.46 | mazpe | WIMPy: getting ERROR[14898]: chan_skinny.c:4760 handle_register_message: Rejecting Device ATA001E7A2576D2: Device not found |
| 00:14.56 | mazpe | Wonder if its safe to noload |
| 00:15.18 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 00:15.18 | WIMPy | Do you use any SCCP phones? |
| 00:15.39 | mazpe | nope |
| 00:15.47 | WIMPy | Then sure, yes. |
| 00:15.54 | mazpe | I don't even know hat SCCP is… lol |
| 00:16.04 | mazpe | ah the cisco call manager stuff |
| 00:16.22 | WIMPy | That's what Cisco phones use, unless thay have a SIP firmware. |
| 00:16.54 | WIMPy | But from the message, someone seems to try to use it. |
| 00:18.08 | [TK]D-Fender | _370z-hsv: You joined and within 1 MINUTE you declared this room 100% dead. You are officially off the case study due to lack of understanding of sample size & distribution.... |
| 00:18.19 | WIMPy | But it makes sense to get rid of stuff you don't need. At least as far as it communicates. |
| 00:18.30 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 00:18.36 | mazpe | WIMPy yup. |
| 00:20.43 | _370z-hsv | heh |
| 00:24.54 | mazpe | how about this one: db.c: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied |
| 00:25.00 | mazpe | dirving me insane… loading. the drivers |
| 00:25.29 | mazpe | loading the log file i mean. |
| 00:27.02 | mazpe | my astdb is owned by root |
| 00:27.03 | mazpe | -rw-r--r--. 1 root root 20480 May 13 20:25 /var/lib/asterisk/astdb |
| 00:27.19 | mazpe | asterisk is running as root |
| 00:31.52 | [TK]D-Fender | O RLY? |
| 00:34.04 | mazpe | ? |
| 00:38.44 | *** part/#asterisk trung (~trung@ec2-23-23-179-94.compute-1.amazonaws.com) |
| 00:50.35 | darrenlooby | Hello all. I'm running "sip show peers" and the first column - Name/username is too short... is there a way I can change the length, as we have quite long usernames to deal with |
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| 02:18.53 | *** join/#asterisk Yourname` (~Administr@unaffiliated/yourname/x-837320) |
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| 02:37.40 | *** join/#asterisk Yourname` (~Administr@unaffiliated/yourname/x-837320) |
| 02:39.55 | Yourname` | Howdy! |
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| 03:05.28 | Mercury_Zero | I can't seem to get the slin44 codec working between two asterisk boxes with SIP. Is it not supported? |
| 03:07.46 | Mercury_Zero | It lets me specify it in the config, but it insists that the peer has no compatible codecs. |
| 03:16.30 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 03:16.53 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 03:17.13 | ruben23 | hi guys anyone have successfully install dahdi on VPS..any idea guys |
| 03:17.51 | Mercury_Zero | I'm trying to do a CD quality conference bridge so i can record a podcast. Apparently slin44 isn't working, and I can't get chan_alsa to support more than one sound device at the same time. |
| 03:24.44 | *** join/#asterisk JamKo (~JamKo@173-162-11-94-naples.hfc.comcastbusiness.net) |
| 03:34.27 | *** join/#asterisk JamKo (~JamKo@173-162-11-94-naples.hfc.comcastbusiness.net) |
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| 04:02.09 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-yymmpqofwchfulat) |
| 04:07.42 | ruben23 | hi guys anyone have successfully install dahdi on VPS..any idea guys |
| 04:36.36 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 04:57.45 | lanning | dahdi is a hardware driver. you have no dahdi supported hardware in a VPS. |
| 05:11.16 | *** join/#asterisk screenn (~screenn@37.46.237.217) |
| 05:14.14 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
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| 05:20.12 | [TK]D-Fender | lanning: incorrect |
| 05:20.34 | [TK]D-Fender | lanning: It is a generic interface layer that includes mixing, etc |
| 05:20.55 | [TK]D-Fender | lanning: And yes there is plenty of VM support depending on implementation |
| 05:21.20 | *** join/#asterisk btse (~btse@c83-253-249-43.bredband.comhem.se) |
| 05:30.29 | lanning | how do you get correct timing? (even standard TOD doesn't really work) |
| 05:35.11 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 05:54.55 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 05:56.10 | bulkorok | good morning |
| 05:59.50 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 05:59.52 | schmidts | good morning |
| 06:01.21 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 06:03.33 | ghghz | morning. what is the best way to do two outgoing call bridging? |
| 06:03.33 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 06:04.07 | *** join/#asterisk mpe (~mpe@gate.ipvision.dk) |
| 06:06.50 | bulkorok | ghghz: Dial(SIP/trunk1/1234&SIP/trunk2/1235) |
| 06:07.27 | ghghz | they would talk together? |
| 06:07.47 | bulkorok | no... just the one how picks up the call |
| 06:08.01 | bulkorok | if you want more than 2 partners you need MeetMe |
| 06:08.09 | ghghz | only 2 i need |
| 06:09.04 | bulkorok | so you can use Dial with & but thats connect the caller with one of the two called... not the two that are called! |
| 06:10.01 | bulkorok | if you want this you can use Originate (core show application Originate) |
| 06:10.52 | ghghz | seems that not so I need. I need that 2 outgoing calls originated from asterisk will be bridged |
| 06:11.09 | ghghz | asterisk calls to 991 and to 992 |
| 06:11.14 | bulkorok | then use Originate |
| 06:11.17 | ghghz | they should be bridged and talk together |
| 06:11.24 | qakhan | i have TDM800p it doesnt hangup call |
| 06:12.16 | qakhan | until caller hangup his phone and my line get busy |
| 06:13.13 | bulkorok | you call Originate(SIP/bla/991,exten,context,exten,prio) and if 991 picks up, the call will be send to kontext, exten, prio you specify... |
| 06:13.20 | bulkorok | and there you can make a dial to 992 |
| 06:13.46 | ghghz | bulkorok: good point |
| 06:14.09 | bulkorok | qakhan: did you try h-extension!? |
| 06:14.42 | bulkorok | or put a Hangup() at the end od your dialplan!? |
| 06:14.53 | qakhan | bulkorok can u tell me how i use h ext |
| 06:15.02 | qakhan | i m using Hangup() |
| 06:15.13 | bulkorok | exten => h,1,Hangup() |
| 06:15.37 | qakhan | well i m use Hangup() but it doesnt work |
| 06:15.53 | bulkorok | or try to specifay a hangup-cause... e.g Hangup(16) for busy |
| 06:16.42 | kaldemar | 16 is normal clearing, not busy. |
| 06:16.45 | qakhan | bulkorok my card hangup call for only 2 operators out of 8 |
| 06:16.46 | [TK]D-Fender | no such thing for ANALOG |
| 06:17.00 | kaldemar | qakhan: what did your telco say? |
| 06:17.13 | [TK]D-Fender | bed time, I'm off |
| 06:17.43 | qakhan | they said they are not providing disconnect supervision |
| 06:17.51 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 06:17.53 | qakhan | check you pbx |
| 06:17.57 | qakhan | your* |
| 06:18.22 | bulkorok | ghghz: little failure... use Originate(SIP/bla/991,context,exten,prio) |
| 06:18.27 | kaldemar | qakhan: what did you ask them? |
| 06:19.50 | qakhan | kaldemar i asked them i have asterisk pbx and my call doesnt hangup. so they said, did you checked line by attaching it with phone and make call |
| 06:20.15 | qakhan | i said i did it and phone hangup call |
| 06:20.35 | qakhan | so they said its mean our line doesnt has issue its your pbx |
| 06:22.41 | kaldemar | that's just useless. you should ask how they think they detect a hangup. |
| 06:24.50 | qakhan | kaldemar seriously i m tried of it |
| 06:25.01 | qakhan | it is useless to talk to them |
| 06:25.25 | ghghz | bulkorok: thank you, it works. |
| 06:25.33 | bulkorok | great :-) |
| 06:42.38 | ghghz | hm, what is wrong here? http://p.defau.lt/?pf5nP1chOLxE7bfISlyShQ |
| 06:43.50 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
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| 07:11.32 | *** join/#asterisk gavimobile1 (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 07:11.51 | gavimobile1 | can one of the admins please unban me on my other nick gavimobile |
| 07:14.54 | gavimobile1 | also, is there a way to exclude specific dial patterns in a dialplan, for example if ext 411 is used for information, and my dialplan looks like this exten => _NXX,1,Dial(${EXTEN}) |
| 07:15.12 | wdoekes2 | gavimobile1: just add 411 and it will take precedence |
| 07:15.30 | gavimobile1 | wdoekes2: right, but is there an order difference? |
| 07:15.40 | gavimobile1 | I understand asterisk finds the closest dial pattern |
| 07:16.03 | gavimobile1 | but wouldn't it be safer for extra percautions to create some sort of exclude rule as well? |
| 07:16.43 | wdoekes2 | if you're banned from freenode, you should probably ask for unbanning somewhere else |
| 07:16.59 | gavimobile1 | wdoekes2: no, im banned from the channel |
| 07:17.10 | wdoekes2 | what did you do to get banned? |
| 07:17.12 | gavimobile1 | Qwell: banned me for multiple disconnections apparently |
| 07:17.24 | gavimobile1 | my router is f'ed |
| 07:17.31 | gavimobile1 | but I had it taken care of |
| 07:17.32 | wdoekes2 | mm |
| 07:18.13 | wdoekes2 | those with op-powers aren't awake at this hour. you'll have to wait until mid-day UTC |
| 07:18.31 | wdoekes2 | as for precautions.. against bugs? |
| 07:18.52 | gavimobile1 | anyways in php we use extra percautions, for example if ($_POST['EXTEN'] != 411 && $_POST['EXTEN'] == _NXX){ exten => _NXX,1,Dial(${EXTEN}) } |
| 07:19.10 | bulkorok | gavimobile1: you can make sth like this: exten => [235-9][02-9][02-9],1,Dial() |
| 07:19.10 | wdoekes2 | the matcher will find the most narrow match, unless it is broken |
| 07:19.23 | wdoekes2 | bulkorok: that won't work.. now 511 won't match either |
| 07:19.42 | gavimobile1 | do you guys think its necessary? |
| 07:19.45 | wdoekes2 | no |
| 07:20.11 | gavimobile1 | ok and what about the order? is there an order asterisk reads through the extensions.conf file? |
| 07:20.17 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 07:20.19 | bulkorok | why shouldn't 511 match!? (except for forgetting the underscore ^^) |
| 07:20.20 | gavimobile1 | no, right? |
| 07:20.48 | ruben23 | hi guys any one have successfully setup dahdi on a virtual server..? |
| 07:20.59 | bulkorok | asterisk hits the "best matching" extension firts... |
| 07:21.07 | wdoekes2 | bulkorok: ... because you're not matching a 1 at pos 1 nor 2 |
| 07:21.13 | wdoekes2 | 501 won't match either |
| 07:21.20 | gavimobile1 | alright, thanks folks |
| 07:21.21 | wdoekes2 | or 401 for that matter |
| 07:21.46 | bulkorok | mmh |
| 07:21.51 | gavimobile1 | so I should in addition make a context for services 411 etc and emergencies 911 etc |
| 07:21.52 | gavimobile1 | right |
| 07:21.53 | gavimobile1 | ? |
| 07:22.08 | wdoekes2 | gavimobile1: if you want to be certain that you didn't forget the 411, you could add a GotoIf($[${EXTEN}=411]?...) on prio 1 of _NXX instead |
| 07:22.17 | bulkorok | from voip-info: [1237-9] matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9) |
| 07:22.42 | bulkorok | [] indicates one digit or letter at this position |
| 07:22.54 | gavimobile1 | wdoekes2: will the above solution do fine? |
| 07:22.59 | gavimobile1 | " in addition make a context for services 411 etc and emergencies 911 etc" |
| 07:22.59 | bulkorok | so [1][2][3] is the same like 123 |
| 07:23.06 | gavimobile1 | ? |
| 07:23.11 | gavimobile1 | will that do? |
| 07:23.19 | bulkorok | you have to include this context then ^^ |
| 07:23.25 | gavimobile1 | right |
| 07:23.25 | wdoekes2 | gavimobile1: yes: you won't need to exclude 411/911 fron _NXX if you add separate 411 and 911 extens |
| 07:23.38 | gavimobile1 | but is that the recommended way? |
| 07:23.53 | gavimobile1 | the "good practice" way? |
| 07:24.02 | wdoekes2 | bulkorok: you can quote voip-info all you want, it still won't match 4?? or ?1? or ??1 |
| 07:24.05 | bulkorok | it's a working way... I would do this, because you can include in other contextes too |
| 07:24.20 | wdoekes2 | (and voip-info is never and authoritative source) |
| 07:24.23 | gavimobile1 | so how would you do it? |
| 07:24.25 | wdoekes2 | s/and/an |
| 07:24.55 | wdoekes2 | that's what I do gavimobile1. I have _1! for certain special extens, and 112 for the european 911 |
| 07:25.08 | bulkorok | wdoekes2: isn't it that what gavi wanted?! |
| 07:25.18 | bulkorok | everything except 411 !? |
| 07:25.45 | gavimobile1 | bulkorok: no, 411 was just an example that there are extentions which are not extentions but outbound calls rather |
| 07:25.51 | wdoekes2 | .. and except 201 and except 211 and except 210 and except 301 |
| 07:25.51 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:6017:2b0d:dcd4:dea4) |
| 07:25.58 | wdoekes2 | et cetera |
| 07:26.12 | gavimobile1 | this is a list of local extentions I don't want to use http://www.d.co.il/emergency-phones.html |
| 07:26.24 | bulkorok | mmh... |
| 07:27.00 | wdoekes2 | bulkorok: you're making a logic error.. replace your first _[2-35-9] with _N and check what you're left with |
| 07:27.57 | bulkorok | _N stands for 2-9 and _[2-35-9] is 2-9 without 4 ... so 4. doesn't match... |
| 07:28.56 | bulkorok | mmh |
| 07:29.03 | gavimobile1 | if we are not recommended to use numbers as peers, does that mean its not possible to make 1 dial rule for all the internal extentions and I need to make a new separate rule for each peer? |
| 07:29.17 | bulkorok | I see... if he wants 401 it's a prob ^^ |
| 07:29.30 | wdoekes2 | finally |
| 07:29.40 | ghghz | bulkorok: did you tried to use channel originate SIP/.. application Macro ? |
| 07:29.59 | wdoekes2 | gavimobile1: you can so _s-! and Dial(SIP/${EXTEN:2}) |
| 07:30.03 | gavimobile1 | for example, if I wanted to make this universal |
| 07:30.22 | gavimobile1 | http://pastebin.com/ES9fYcER |
| 07:30.43 | wdoekes2 | or something.. but since I do use numbers as peers, I don't know what the recommended practise is |
| 07:30.59 | gavimobile1 | :-( |
| 07:31.08 | bulkorok | ghghz: I don't use Macro anymore... it's not recommended anymore... Using GoSub instead... |
| 07:31.26 | gavimobile1 | anyone who does use names as peers? |
| 07:31.29 | wdoekes2 | 09:29 < wdoekes2> gavimobile1: you can so _s-! and Dial(SIP/${EXTEN:2})\ |
| 07:31.55 | gavimobile1 | wdoekes2: that was for the previous subject, no? |
| 07:32.12 | ghghz | bulkorok: but how to give arguments? |
| 07:32.18 | gavimobile1 | I thought we agreed that I was going to create a new context with services and emergencies |
| 07:32.28 | wdoekes2 | _s-! matches s-somepeer, which will then dial SIP/somepeer |
| 07:32.35 | bulkorok | ghghz: wait |
| 07:32.49 | gavimobile1 | wdoekes2: ahh |
| 07:32.49 | *** join/#asterisk cstachris (~chrismylo@202.182.147.81) |
| 07:32.53 | gavimobile1 | let me look in to this |
| 07:32.54 | gavimobile1 | thanks |
| 07:34.14 | gavimobile1 | what does "s" stand for? |
| 07:34.22 | bulkorok | ghghz: GoSub(context,exten,prio,arg1,arg2,argN) |
| 07:34.54 | gavimobile1 | don't see it listed in the definitive guide |
| 07:35.33 | bulkorok | gavi: https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 07:35.44 | *** part/#asterisk Lo-lan-do (~roland@polymir.placard.fr.eu.org) |
| 07:36.04 | gavimobile1 | bulkorok: thanks for the link |
| 07:36.35 | gavimobile1 | great link!!! |
| 07:37.41 | wdoekes2 | gavimobile1: s doesn't stand for anything per se.. although it is used for 'start' in certain scenario's (e.g. as the default extension for channel drivers that don't get an exten) |
| 07:38.50 | bulkorok | ghghz: you can get every info for applications or functions with: core show application/function NAME |
| 07:38.59 | gavimobile1 | the description talks about usage for macros or analog calls |
| 07:39.00 | bulkorok | in asterisk CLI |
| 07:39.14 | gavimobile1 | im going to test with it |
| 07:39.23 | bulkorok | gavi: it works in every tech |
| 07:39.41 | bulkorok | sip, dhadi, iax... doesn't matter |
| 07:40.47 | gavimobile1 | im not exactly clear still :-( |
| 07:40.57 | gavimobile1 | my logic is incorrect |
| 07:41.05 | gavimobile1 | exten => _ZXX,1,Dial(SIP/_s-!${EXTEN} <--still incomplete im thinking |
| 07:41.29 | gavimobile1 | but im guessing the right way is exten =? _s-! not sure why |
| 07:41.42 | wdoekes2 | er.. that is wrong indeed.. now you're mixing things up |
| 07:41.49 | gavimobile1 | exten =_s-! * |
| 07:42.03 | wdoekes2 | where is your mapping between extens and devices? |
| 07:42.58 | gavimobile1 | peers have names in sip.conf, the extensions.conf file as of now has a dial rule for each peer |
| 07:43.14 | gavimobile1 | so for example exten => 113,1,Dial(SIP/zoiperHPlaptop,10) |
| 07:43.23 | gavimobile1 | 113 will dial zoiperHPlaptop |
| 07:43.40 | gavimobile1 | I hope I answered your question accordingly |
| 07:43.49 | wdoekes2 | that will work.. you can't expect asterisk to magically know that zoiperHPlaptop wants to get dialed when you press 113 |
| 07:44.15 | wdoekes2 | so you need a mapping like that or through a different mechanism |
| 07:44.46 | gavimobile1 | yup, I know it works, however im trying to cleanup my code here though and instead of having 100 dial rules, 1 for each peer, I want to create 1 rule for all |
| 07:44.51 | gavimobile1 | using the global variables |
| 07:45.13 | gavimobile1 | but the only logical way that I can think of to do this is instead of names I will need numbers for peers |
| 07:45.24 | gavimobile1 | and than I can take advantage of the EXTEN variable |
| 07:45.59 | bulkorok | gavi: you can put the names in asterisk DB with the according numbers... yo you just have to ask your DB for the name |
| 07:46.00 | gavimobile1 | wdoekes2: I know asterisk won't know, but im still not clear on what _s will do, or how I specify which peer to dial |
| 07:46.31 | gavimobile1 | bulkorok: that's a great idea, I just didn't get to the developing side of asterisk yet |
| 07:47.26 | wdoekes2 | gavimobile1: _s- is for after the mapping.. e.g. you have 113,1,Goto(s-somepeer,1) and then _s-!,1,stuff-before-dial and then _s-!,the-actual-dial |
| 07:47.27 | gavimobile1 | the ari's or whatever its called etc. |
| 07:47.37 | bulkorok | it's bit boring to put 100 peers in the DB but they won't disappear after reboot too... |
| 07:48.18 | gavimobile1 | I think im going to leave it as is, and continue reading |
| 07:48.33 | gavimobile1 | I am trying to jump ahead of myself |
| 07:48.38 | bulkorok | :) |
| 07:48.42 | gavimobile1 | :-D |
| 07:48.45 | gavimobile1 | back to the books |
| 07:48.52 | gavimobile1 | thanks for the advice folks |
| 07:49.34 | ghghz | bulkorok: what I need to do, have you one more advice? :) http://p.defau.lt/?XRkbWRQ4YvW_Dv36lmsJzw |
| 07:53.57 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:54.13 | bulkorok | ghghz: mmh... at first: try not to use Macro (core show application Macro) |
| 07:54.25 | bulkorok | and second,, why not sending to an extension!? |
| 07:54.55 | ghghz | bulkorok: but I need to forward some arguments |
| 07:55.09 | ghghz | is it possible to send to extension arguments? |
| 07:55.11 | bulkorok | ghghz you can do that with gosub too |
| 07:55.53 | ghghz | okey, I will figure it out |
| 07:58.38 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:01.28 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:07.19 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 08:09.53 | ruben23 | hi guys any one have successfully setup dahdi on a virtual server..? |
| 08:22.39 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 08:23.21 | aberrios | Why when doing a queue show would some agents show as their names and some show as Local/blah ?? The Local/Blah's all seem to be softphones and the ones with names are on hardphones.... |
| 08:24.22 | ghghz | is it possible to give arguments to extension? |
| 08:44.39 | *** join/#asterisk bulkorok (~bulkorok@ad96ec5e1.dsl.de.colt.net) |
| 08:49.28 | *** join/#asterisk cstachris (~chrismylo@202.182.147.115) |
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| 08:56.51 | *** join/#asterisk timahvo1 (~rogue@41.81.95.239) |
| 09:10.03 | verywiseman | when i make a call from sip client to another ,the sender can't hear the receiver , where is the problem? |
| 09:11.16 | *** join/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 09:11.19 | v0lZy | hello |
| 09:11.49 | v0lZy | U gave a quick question |
| 09:12.18 | v0lZy | is it possible to configure asterisk in such a way that as soon as i lift the handset, a call is placed? |
| 09:12.29 | v0lZy | or is this a phone specific setting? |
| 09:19.04 | Yourname` | What does it mean when a remote agent calls another remote agent via the same pbx that is behind nat? |
| 09:19.09 | Yourname` | And there's no audio. |
| 09:23.11 | verywiseman | when i make a call from sip client to another ,the sender can't hear the receiver , where is the problem? |
| 09:24.27 | v0lZy | i guess NAT in both casses |
| 09:27.07 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:29.46 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 09:34.38 | *** join/#asterisk arekm (matrix157@pld-linux/arekm) |
| 09:35.20 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 09:36.06 | *** join/#asterisk amessina (~amessina@2001:470:1f11:a4:d6be:d9ff:fe8d:7c1e) |
| 09:38.29 | arekm | hi, I'm trying to connect asterisk 1.4.43 with 1.8.11.1 via iax2. Config is like this: http://pastebin.com/SPmNqKPs Connection 1.8->1.4 works fine but 1.4->1.8 tells me chan_iax2.c:9470 socket_process: I don't know how to authenticate asterisk-pbx to 192.168.1.112 |
| 09:38.44 | arekm | any clues? |
| 09:39.49 | Yourname` | v0lZy: thanks |
| 09:40.22 | *** join/#asterisk hehol (~hehol@217.9.101.222) |
| 09:42.42 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 09:46.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 09:46.59 | arekm | If I add secret=something on both ends then I get "Call rejected by 192.168.1.112: No authority found" |
| 09:47.16 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 09:47.40 | arekm | and on 112 I get "chan_iax2.c:10991 socket_process: Host 192.168.1.111 failed to authenticate as asterisk-pbx" |
| 09:51.49 | arekm | make no sense to me sine both ends have nearly identical iax.confs hmm |
| 10:01.06 | arekm | blah, context didn't match |
| 10:03.37 | *** join/#asterisk timahvo1 (~rogue@41.90.160.210) |
| 10:04.01 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:07.34 | *** join/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 10:08.36 | PBXman | Hello I'd like to make a demo for a large scale asterisk platform, that is to say at least 1500 extensions, can anybody give me some advice on how to do this? |
| 10:17.06 | v0lZy | hm.. softphoens. |
| 10:17.08 | v0lZy | softphones* |
| 10:22.03 | *** join/#asterisk danfromuk (~IceChat77@2.26.250.118) |
| 10:25.39 | aberrios | can anyone help me out with a bit of PRI debugging? |
| 10:25.52 | WIMPy | ~ask |
| 10:25.52 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 10:25.57 | WIMPy | aberrios: ^^ |
| 10:26.00 | aberrios | just about to paste it |
| 10:26.39 | aberrios | http://pastebin.com/0fED3mE8 |
| 10:26.54 | aberrios | Doesn't seem to be causing much of an issue but its annoying |
| 10:31.21 | WIMPy | wonders why these debus have different formats each and every time. |
| 10:31.34 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a151e.pool.mediaWays.net) |
| 10:32.04 | WIMPy | Unless that's a response to something you sent, there isn't much to do about that. |
| 10:34.11 | qakhan | WIMPy can you tell me now what i do |
| 10:35.27 | WIMPy | Get a supported card. |
| 10:35.38 | qakhan | its TDM800p |
| 10:36.06 | WIMPy | You got an original now? |
| 10:36.34 | qakhan | yes i checked orignal |
| 10:36.40 | qakhan | but same result |
| 10:36.50 | ectospasm | is it registered? |
| 10:36.54 | WIMPy | The best on only safe solution wouuld be to get rid if the analog ah^Wstuff. |
| 10:37.28 | WIMPy | If it's original, contact Digium support. |
| 10:37.37 | qakhan | ok |
| 10:37.56 | qakhan | ectospasm you told me something |
| 10:38.00 | qakhan | i lost it |
| 10:38.11 | ectospasm | opermode=PAKISTAN |
| 10:38.24 | ectospasm | ...in /etc/modprobe.d/dahdi.conf |
| 10:38.33 | ectospasm | and you should keep IRC logs. |
| 10:45.29 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 10:45.47 | qakhan | ectospasm you tell me some other line |
| 11:04.19 | *** join/#asterisk cstachris (~chrismylo@202.182.147.115) |
| 11:06.35 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 11:07.05 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 11:11.57 | *** join/#asterisk arekm (matrix157@pld-linux/arekm) |
| 11:13.11 | arekm | hi, two asterisks connected with iax2, one responds -- Auto fallthrough, channel 'IAX2/asterisk-pbx-272' status is 'CHANUNAVAIL' but the other sees it as "IAX2/asterisk-dongle-18654 is circuit-busy" which ends up as CONGESTION ? Why it doesn't transmit CHANUNAVAIL properly? |
| 11:15.45 | *** part/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 11:19.20 | *** join/#asterisk hariom (~hariom@115.118.173.77) |
| 11:25.59 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 11:28.55 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 11:33.06 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 11:36.08 | *** join/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 11:46.39 | *** join/#asterisk Seppoz (~Invincibl@unaffiliated/seppoz) |
| 11:47.01 | Seppoz | hey, im looking for a suitable hardware for an asterisk server for 5-10 users |
| 11:56.57 | v0lZy | im sures theres people who have extra working hardware and will give it to you for free just so u take it out of their garage |
| 11:59.04 | *** join/#asterisk Bullmoose (~Bullmoose@71-33-10-124.bois.qwest.net) |
| 12:06.48 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:07.14 | *** join/#asterisk Yourname` (~Administr@unaffiliated/yourname/x-837320) |
| 12:09.41 | Yourname` | Asterisk as a SIP server behind nat, clients on the outside behind a second NAT connecting to Asterisk. Port forwarding done on both ends. STUN server also being used. Qualify=yes, nat=yes, canreinvite=no for the remote clients, and the local clients. Choppy call audio from remote sip - remote sip extensions. Choppy calls when I log in *11 and when checking voicemail. Basically choppiness all around. When an inbound call comes into |
| 12:09.42 | Yourname` | the server that comes to the remote agent - 9 out of 10 cals are clear. Any idea what's up anyone? Oh and when I set externip there's dead audio |
| 12:10.18 | *** join/#asterisk `paul (70c65b54@gateway/web/freenode/ip.112.198.91.84) |
| 12:11.36 | ruben23 | hi guys any one have successfully setup dahdi on a virtual server..? |
| 12:12.14 | [TK]D-Fender | ruben23, "virtual server" doesn't actually say anything specific to qualify it. |
| 12:12.23 | `paul | im using ast 1.8 when i do originate call from php "Channel: Local/100@context1; Exten: 101; Context: context2" i can save custom cdr fields in extension 100 but not in 101. please help |
| 12:12.43 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 12:14.19 | ruben23 | [TK]D-Fender: i rented a hosted VPS and planning somehow to setup asterisk on it but im stock with dahdi-install, askign for linux source..i guess with kernel source issue, any idea how to make this work..? |
| 12:14.47 | WIMPy | Why do you want to install dahdi on a VPS? |
| 12:15.11 | *** join/#asterisk cstachris_ (~chrismylo@202.182.147.115) |
| 12:15.36 | [TK]D-Fender | ruben23, depends on the kernel, the virtualization method, etc. |
| 12:15.54 | [TK]D-Fender | ruben23, You don't ssem to have any details at all to offer us. We cannot answer your question. |
| 12:16.12 | [TK]D-Fender | ruben23, have you even tried yet? Do you have error reports to show us? |
| 12:16.44 | [TK]D-Fender | ruben23, and yes, if it's asking for kernel source, you do indeed need the headers. Go ask your provider |
| 12:16.59 | [TK]D-Fender | ruben23, You need them. This is not an option. |
| 12:22.21 | tzafrir | ruben23, what type of VPS? |
| 12:23.23 | *** join/#asterisk MarKsaitis (~MarKsaiti@195.59.185.18) |
| 12:24.30 | ruben23 | i will try to cehck with provider..and get back right away, thanks |
| 12:26.54 | `paul | im using ast 1.8 when i do originate call from php "Channel: Local/100@context1; Exten: 101; Context: context2" i can save custom cdr fields in extension 100 but not in 101. please help |
| 12:27.11 | zamba | i have a problem when dialing out.. if it's busy in the other end, i don't hear the busy tone in my sip client.. what could be the problem here? |
| 12:27.24 | zamba | instead of hearing a busy tone, the conversation just hangs up |
| 12:27.49 | WIMPy | zamba: That's your cient. |
| 12:28.05 | zamba | and i even get the dial feedback before the connection has started setting up |
| 12:28.25 | zamba | i'm using SFLphone |
| 12:30.40 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 12:31.30 | [TK]D-Fender | zamba, that is how SIP actually works. |
| 12:31.50 | [TK]D-Fender | zamba, OOB signalling and indications |
| 12:31.55 | *** join/#asterisk kaldemar (~kaldemar@unaffiliated/kaldemar) |
| 12:33.17 | zamba | any way i can get some more analogue responses back into my client? |
| 12:34.08 | WIMPy | No |
| 12:34.14 | [TK]D-Fender | zamba, Answer the call in * before you dial out |
| 12:34.25 | [TK]D-Fender | zamba actually.... more than than... |
| 12:34.25 | WIMPy | Neitehr Asterisk nor SIP support audio after the call has ended. |
| 12:34.40 | [TK]D-Fender | zamba, You'll need to check your DIALSTATUS, use Playtones, etc |
| 12:34.57 | [TK]D-Fender | zamba, And delay things in the dialplan. |
| 12:35.11 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 12:35.11 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 12:35.44 | WIMPy | Yes, you can do it manually. |
| 12:36.15 | WIMPy | But you neither get the announcement from your telco nor wil your CDRs be usable for billing. |
| 12:38.39 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-alopdbcmalaaxjsh) |
| 12:38.56 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-lemgslyxeizjksks) |
| 12:40.25 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 12:44.42 | WIMPy | I still vote for a 'disconnected' channel state in Asterisk. |
| 12:45.14 | *** join/#asterisk elzid (~IceChat77@80.168.5.31) |
| 12:50.19 | *** join/#asterisk JamKo (~JamKo@mail.jaguardata.com) |
| 12:55.49 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:55.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:57.38 | *** join/#asterisk rossand (~aross@CPE009400809a9c-CM78cd8ed45eb5.cpe.net.cable.rogers.com) |
| 13:02.35 | elzid | Hi - do I need to implement a STUN server to allow a sip client to communicate with another sip client via asterisk 1.8? The sip clients are typically end users behind ADSL (NAT) routers. |
| 13:02.54 | WIMPy | No |
| 13:03.04 | WIMPy | You usually dont even want to use stun. |
| 13:03.49 | elzid | @wimpy - maybe I dont understand - how can I ensure that there are no nat issues between these sip callers? |
| 13:04.04 | elzid | (I thought thats why I'd need to use STUN) |
| 13:05.13 | WIMPy | Let Asterisk handle it. |
| 13:05.35 | elzid | @winpy - is that through sip.conf nat settings etc? |
| 13:05.46 | WIMPy | Exactely. |
| 13:06.51 | elzid | perfect - beyond that, could there be any other issues? a sip client will connect to my asterisk, which will connect to another sip switch (3rd party freeswitch), which will connect to the other sip client |
| 13:08.51 | [TK]D-Fender | elzid, "other issues" can mean absolutely anything |
| 13:09.14 | [TK]D-Fender | elzid, * talks to your phone, calls out to FS. What that does is not *'s problem. |
| 13:09.30 | [TK]D-Fender | elzid, * works fine behind NAT 99% of the time. |
| 13:11.11 | elzid | Thanks - I'm aware of one way audio issues from the past, that's all - will default asterisk sip.conf settings for nat, one-way audio should not be an issue (outside of FS)? |
| 13:12.48 | [TK]D-Fender | <PROTECTED> |
| 13:13.35 | elzid | ok, thanks - curious what is all the talk about a need for sip proxies/nat traversal kit to handle "sip nat issues"? Any links I can read to get up to speed? |
| 13:14.35 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-wbtcsvddlndnbcjf) |
| 13:14.35 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:14.47 | [TK]D-Fender | elzid, It's all in sip.conf and we cannot validate what you claim to have "heard". |
| 13:15.13 | [TK]D-Fender | elzid, Howver anti-psychotics are in the bowl to the left of the main door... |
| 13:15.40 | elzid | great! thanks |
| 13:16.55 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:24.54 | *** join/#asterisk faithlove (proxyuser@188.27.133.9) |
| 13:25.37 | faithlove | hi. how can i set up a blacklist for inbound? some numbers keep calling me and i just want to block them from asterisk |
| 13:25.48 | *** join/#asterisk serafie (~erin@nat/digium/x-rifqicujpctarnsv) |
| 13:25.52 | [TK]D-Fender | faithlove, "core show application gotoif" |
| 13:25.59 | [TK]D-Fender | faithlove, "core show function CALLERID" |
| 13:26.12 | [TK]D-Fender | faithlove, and/or basic dialplan patterns on those extens. |
| 13:26.14 | [TK]D-Fender | ~book |
| 13:26.14 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:26.15 | [TK]D-Fender | ^^^ |
| 13:26.22 | faithlove | thanks |
| 13:27.38 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 13:28.28 | *** join/#asterisk alexscott (~alexscott@37.122.202.194) |
| 13:29.42 | *** part/#asterisk v0lZy (~chatzilla@mail.silk-group.net) |
| 13:29.57 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 13:30.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:30.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:36.25 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
| 13:36.36 | MrTelephone | under what conditions would asterisk not respond to register requests? |
| 13:36.55 | MrTelephone | My servers just stopped responding this morning to any messages what so ever |
| 13:37.25 | schmidts | MrTelephone sounds like a Deadlock problem |
| 13:37.56 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:38.37 | [TK]D-Fender | MrTelephone, Can you show us that attempts coming in with full SIP debug enabled? |
| 13:38.57 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 13:39.57 | MrTelephone | what is the command for full debug? |
| 13:40.40 | [TK]D-Fender | ... |
| 13:40.42 | [TK]D-Fender | sip set debug on |
| 13:40.49 | MrTelephone | yeah nothing shows up |
| 13:41.15 | [TK]D-Fender | If nothing shows us... that means the register isn't even REACHING * |
| 13:41.23 | [TK]D-Fender | Which is not *'s problem |
| 13:41.35 | MrTelephone | I see the traffic with tcpdump |
| 13:41.37 | schmidts | [TK]D-Fender> thats not true |
| 13:41.59 | [TK]D-Fender | schmidts, When does SIP debug fail to show a packet inbound at all? |
| 13:42.05 | schmidts | if a deadlock happens inside the sip do_monitor loop sip debug wouldnt show anything |
| 13:42.15 | [TK]D-Fender | schmidts, Failing to open an outbound socket is something else, but "in"? |
| 13:42.21 | [TK]D-Fender | How is * responsible for that? |
| 13:42.37 | MrTelephone | it's an odbc problem |
| 13:42.38 | schmidts | sip debug happens after the packet is taken from the buffer and processed in some functions |
| 13:42.41 | MrTelephone | i removed odbc and it's working |
| 13:43.38 | [TK]D-Fender | schmidts, I suppose if something cataclysmically bad happens... but then again... how many more cases of this symptom should we have seen? |
| 13:44.23 | schmidts | [TK]D-Fender it depends on the asterisk version but i have seen this much too often with version before 1.8.x |
| 13:44.57 | [TK]D-Fender | schmidts, I've never heard of it in my 8 years of working with * |
| 13:45.58 | pabelanger | Hey, anybody using asterisk with an Avaya ip office ? |
| 13:46.12 | schmidts | pabelanger only instead but not "with" ;) |
| 13:46.29 | *** join/#asterisk MK92 (~coming@151.66.219.212) |
| 13:46.32 | pabelanger | schmidts: have you tried hooking asterisk to it? |
| 13:46.44 | pabelanger | Working on a new quote for a customer and they have an existing one |
| 13:47.09 | schmidts | [TK]D-Fender imho you could see outgoing packets from asterisk to some clients, atleast when these packets are not coming from the sip scheduler |
| 13:47.21 | *** part/#asterisk cstachris_ (~chrismylo@202.182.147.115) |
| 13:47.48 | MrTelephone | odbc caused a deadlock for no reason this morning |
| 13:47.57 | MrTelephone | never touched the machines for months |
| 13:48.02 | MrTelephone | and all of a sudden, bang |
| 13:48.07 | schmidts | MrTelephone do you use sip realtime over odbc? |
| 13:48.09 | MrTelephone | yeah |
| 13:48.14 | MrTelephone | I had to disable it |
| 13:48.26 | schmidts | i see ;) |
| 13:48.36 | MrTelephone | what good is it if it decides to quit working one morning |
| 13:48.58 | MrTelephone | needs more investigation |
| 13:52.28 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 13:52.46 | qakhan | hi all |
| 13:53.20 | qakhan | i want to retrive and save data in MS SQL 2008 through dialplan |
| 13:53.26 | qakhan | is it possible |
| 13:54.05 | qakhan | ? |
| 13:54.49 | [TK]D-Fender | qakhan, ODBC... go read the book |
| 13:54.50 | [TK]D-Fender | ~book |
| 13:54.50 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:54.52 | [TK]D-Fender | ^^^^ |
| 13:55.04 | qakhan | ok |
| 13:55.19 | qakhan | is ODBC support asterisk 1.4.38? |
| 13:56.18 | [TK]D-Fender | qakhan, Yes |
| 14:03.49 | ruben23 | <PROTECTED> |
| 14:04.16 | qakhan | [TK]D-Fender i didnt enable ODBC in menuselect while i was installing asterisk |
| 14:04.27 | qakhan | shoudl i run again menuselect? |
| 14:04.30 | ruben23 | thats the reason i cant install header to fit the kernel source |
| 14:04.55 | [TK]D-Fender | qakhan, Clearly |
| 14:05.05 | [TK]D-Fender | ruben23, Go ask your provider |
| 14:06.00 | *** join/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 14:06.12 | ruben23 | <PROTECTED> |
| 14:06.34 | [TK]D-Fender | ruben23, Then go google toher users of your service. |
| 14:10.45 | MrTelephone | I must have did an update at some point and odbc got updated or something |
| 14:10.57 | MrTelephone | what version of myodbc do you need that is known to work well |
| 14:16.33 | *** join/#asterisk kami (~user@unaffiliated/kami-) |
| 14:16.36 | kami | Hello |
| 14:18.54 | *** join/#asterisk teknoprep (~tek@unaffiliated/teknoprep) |
| 14:18.56 | teknoprep | hi all |
| 14:19.25 | teknoprep | i was looking at asterisk as a B2BUA for my existing SIP PBX... i want to be able to use Asterisk as the gateway between my ITSP and my Internal LAN |
| 14:20.07 | teknoprep | is this something that Asterisk would be good at doing ? i just want to pass all traffic from the ITSP to an IP address on the internal side and then pass all traffic from the internal side to the ITSP |
| 14:20.34 | teknoprep | dial plans would be done by the internal PBX... so i would just have a simple dial plan of . |
| 14:20.42 | *** join/#asterisk tnt (~tnt@osmocom/tnt) |
| 14:20.46 | *** join/#asterisk hfb (~hfb@cpe-98-151-249-95.socal.res.rr.com) |
| 14:21.20 | tnt | Anyone has experience running B410p cards in virtualized environment ? Which driver is recommended ? dahdi / misdn v1 / misdn v2 ? |
| 14:21.51 | WIMPy | tnt: http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 14:25.59 | [TK]D-Fender | teknoprep, of course |
| 14:26.27 | teknoprep | cool |
| 14:26.50 | teknoprep | my question would also be... i want to setup asterisk on the LAN side to not need authentication of any sort from the IP of the PBX |
| 14:27.44 | [TK]D-Fender | teknoprep, Fine... |
| 14:28.31 | teknoprep | what would be the type for this connection in asterisk settings |
| 14:29.14 | [TK]D-Fender | "peer" like 99% of everything anyone needs |
| 14:30.14 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 14:34.15 | tnt | WIMPy: thanks. Any comment about the 'in a virtual environment' part ? |
| 14:34.39 | WIMPy | Sounds risky. |
| 14:35.13 | WIMPy | Oh, and my conclusion is: If you want features, go dahdi. If you want stability, go LCR> |
| 14:36.07 | tnt | I'm actually trying to build misdn v2 right now. (Andreas recommended to build the latest rather than use the kernel supplied ones), but the makefile seem to have trouble on the latest ubuntu. |
| 14:36.25 | *** join/#asterisk timahvo1 (~rogue@197.178.121.125) |
| 14:37.22 | tzafrir | ruben23, with openvz you use the host's kernel. You normally can't load your own modules |
| 14:37.27 | WIMPy | Yes, git *should* be ok again. |
| 14:37.30 | MrTelephone | the linkage between t1 drivers, kernel, zaptel, and asterisk is very frustrating |
| 14:37.41 | WIMPy | tnt: I used the kernel version so far. |
| 14:37.48 | tzafrir | zaptel? |
| 14:38.32 | tnt | WIMPy: ok, maybe i'll stick with that at first and only check git if there is issues. |
| 14:38.51 | ruben23 | tzafrir: so tis not possible..? |
| 14:38.52 | tnt | it's a shame there is no packages afaict |
| 14:38.57 | ruben23 | dahdi |
| 14:40.26 | ruben23 | tzafrir: its dahdi |
| 14:41.05 | tzafrir | ruben23, it's technically possible, but the host (your provider) has to explicitly allow you |
| 14:41.18 | tzafrir | or do that for you (more likely) |
| 14:41.46 | tzafrir | If you provider allows you to load modules, your provider allows you to root the host |
| 14:41.53 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 14:43.11 | ruben23 | tzafrir:provider cant help tahts my issue now |
| 14:43.37 | MrTelephone | can't get my pri to dial out now. very frustrating. Probably something simple |
| 14:43.59 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:44.06 | *** join/#asterisk Naikrovek (~sbrule@unaffiliated/naikrovek) |
| 14:44.14 | *** join/#asterisk din3sh (2988f40b@gateway/web/freenode/ip.41.136.244.11) |
| 14:44.26 | *** join/#asterisk darrenlooby (~darrenloo@mail.likeminded.co.uk) |
| 14:44.28 | WIMPy | MrTelephone: Like you have configured a "dialplan" (<>unknown) in chan_dahdi.conf? |
| 14:44.32 | din3sh | gd evening all |
| 14:44.48 | Hive | Does the warning: "chan_sip.c: Remote host can't match request CANCEL to call 'hashedNumber@trunk'. Giving up." look familiar to anyone. Google is not being very helpful here :( |
| 14:46.12 | darrenlooby | Is there any way to modify the output of sip show peers? |
| 14:46.37 | [TK]D-Fender | darrenlooby, You've got the source just like the rest of us :) |
| 14:46.57 | MrTelephone | wimpy, not that I'm aware of |
| 14:47.04 | MrTelephone | do you need it? |
| 14:47.56 | MrTelephone | keep getting cause 27 hangup |
| 14:48.00 | WIMPy | No. Usually you must not have it. But IIRC yu have it by default. |
| 14:48.24 | WIMPy | How much is that in hex? |
| 14:48.43 | WIMPy | Oh |
| 14:48.53 | WIMPy | {0x1B, "Destination out of order"}, |
| 14:49.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:49.48 | MrTelephone | On debug I see send attempts but no receives (pri debug) |
| 14:49.53 | MrTelephone | hmm |
| 14:49.55 | darrenlooby | [TK]D-Fender, bugger... I thought you were going to say that |
| 14:49.57 | MrTelephone | better check my cables |
| 14:50.14 | darrenlooby | I just want to see all the logged in users - but their usernames are cut off |
| 14:50.20 | darrenlooby | Because they are too long |
| 14:50.34 | [TK]D-Fender | darrenlooby, Look at AMI's command set for more complete options |
| 14:51.07 | WIMPy | MrTelephone: Hmm. so it generates that for a local failure? Interesting. |
| 14:52.03 | MrTelephone | It just keeps repeating these lines. PRI Span: 1 SAPI/TEI=0/0 Kick starting link PRI Span: 1 TEI=0 Sending SABME PRI Span: 1 Changing from state 4(TEI assigned) to 5(Awaiting establishment) |
| 14:52.46 | darrenlooby | [TK]D-Fender, I'm currently using a PHP class to call commands to asterisk |
| 14:52.52 | WIMPy | TEI assigned on a PRI??? |
| 14:53.00 | MrTelephone | No |
| 14:53.06 | MrTelephone | I don't even know what that is |
| 14:53.21 | WIMPy | But it is a PRI? |
| 14:54.12 | MrTelephone | Yeah |
| 14:54.34 | *** part/#asterisk PBXman (c335d9a4@gateway/web/freenode/ip.195.53.217.164) |
| 14:54.40 | MrTelephone | 20 channels |
| 15:00.06 | MrTelephone | Is there anything new in chan_Dahdi.conf for asterisk 10+? |
| 15:00.49 | *** join/#asterisk CNpigCA (~CNpigCA@123.62.25.7) |
| 15:02.36 | WIMPy | AOC is from 10 I think. Possibly others. |
| 15:07.01 | MrTelephone | wow |
| 15:08.47 | leifmadsen | CHANGES is a good place to start for changes between versions |
| 15:14.14 | kami | I have an installation of asterisk 1.4 and would like to use an HFC dongle to connect a BRI line with misdn2 and lcr to asterisk. Do you recommend an upgrade of asterisk, before I start, or is there a chance of getting it to work without an upgrade? |
| 15:15.12 | WIMPy | It Should still work with 1.4, but I'd recommend at least 1.8. |
| 15:17.11 | MrTelephone | had the t1 in the wrong port |
| 15:17.21 | MrTelephone | one server has horizontal slots and the other has a riser card |
| 15:17.44 | kami | WIMPy: my box is an ubuntu intrepid 8.10 with kernel 2.6.27. Will asterisk 1.8 work with that kernel? |
| 15:18.22 | leifmadsen | asterisk itself doesn't care about kernel |
| 15:18.37 | leifmadsen | dahdi might |
| 15:18.39 | *** join/#asterisk sickgrinder (~sickgrind@office2.vicidial.com) |
| 15:18.49 | kami | leifmadsen: and misdn2? |
| 15:19.03 | WIMPy | kami: Sure. But it's the very first to support misdn2. So maybe the standlone version might be interesting on that one. |
| 15:20.05 | kami | WIMPy: does "it" in your sentence refer to asterisk. Is LCR obsolete with asterisk 1.8? |
| 15:20.25 | WIMPy | yes and no. In that order. |
| 15:20.47 | WIMPy | There is a can_misdn2, but that's not publicly available. |
| 15:21.02 | *** join/#asterisk HyperNerdV2 (~HyperNerd@cpe-98-149-122-251.socal.res.rr.com) |
| 15:21.03 | kami | If I have to build asterisk anyway, should I take 1.8 or newer? |
| 15:21.33 | kami | Are there known combinations of lcr/misdn2/asterisk which work best? |
| 15:22.20 | MrTelephone | wow that was a nice 3 hour failover |
| 15:22.44 | MrTelephone | I will be having a going out of business sale if anyone is interested |
| 15:22.50 | pabelanger | Any know who sells used 601s? Hopefully in Ontario |
| 15:22.53 | pabelanger | need 1 |
| 15:23.11 | MrTelephone | try ebay |
| 15:23.14 | MrTelephone | I have an old 501 |
| 15:23.16 | WIMPy | kami: As I said: I'd use at leas 1.8. Any release version of 1.8 or 10 is fine. SVN TRUNK does not work at the moment. |
| 15:24.06 | teknoprep | on a new install where is the [default] context |
| 15:24.12 | teknoprep | or do i have to create it |
| 15:24.41 | [TK]D-Fender | teknoprep, You have to create everything |
| 15:25.00 | [TK]D-Fender | teknoprep, And contexts are .... extensions.conf |
| 15:25.14 | teknoprep | thats what i thought |
| 15:25.17 | MrTelephone | I can't beleive I blew up a xeon server. Those things should last 20 years |
| 15:25.25 | teknoprep | there is alot of example code in the Asterisk Default install |
| 15:25.52 | WIMPy | MrTelephone: What did you do to it? |
| 15:25.56 | [TK]D-Fender | teknoprep, You've been here for what ... 5 years now? |
| 15:26.27 | teknoprep | its been 5 years since i worked with asterisk |
| 15:26.34 | teknoprep | i have been working with FreeSwitch and SipXecs |
| 15:27.13 | [TK]D-Fender | teknoprep, I see a bunch of activity in 2010... |
| 15:27.29 | [TK]D-Fender | not THAT far back... anyway.. nothing significant has changed in this respect. |
| 15:27.41 | teknoprep | ok.. just getting my bearings back |
| 15:27.45 | [TK]D-Fender | Dialplan is dialplan. SIP peers are SIP peers.... |
| 15:27.56 | [TK]D-Fender | etc... |
| 15:29.22 | kami | WIMPy: thank you. |
| 15:29.25 | *** part/#asterisk kami (~user@unaffiliated/kami-) |
| 15:40.21 | *** join/#asterisk paryl (~par@168.215.170.202) |
| 15:43.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-196.mobile.uci.edu) |
| 15:43.37 | teknoprep | [May 14 20:42:59] WARNING[26175] chan_sip.c: Retransmission timeout reached on transmission 187ab53d-2a4c8da-33e1a4d3@192.168.1.19 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 15:43.37 | teknoprep | Packet timed out after 6403ms with no response |
| 15:44.06 | teknoprep | so the call never establishes |
| 15:44.06 | WIMPy | Did you look at that URL? |
| 15:44.06 | teknoprep | yes |
| 15:44.42 | WIMPy | And what's the question not yet answered? |
| 15:46.02 | teknoprep | i believe its a misconfiguration on my peer for SIP PBX < - > Asterisk... so its not accepting communication of the SIP packets on one side or the other |
| 15:47.06 | teknoprep | if i have multiple IP address and multiple NIC which NIC / IP will asterisk use ? |
| 15:47.09 | teknoprep | i think that is the issue |
| 15:47.38 | WIMPy | Yes. That doesn't work. |
| 15:48.13 | [TK]D-Fender | teknoprep, Multiple LAN = OK. Multiple WAN can pose issues |
| 15:48.22 | teknoprep | i have 1 wan and 1 lan |
| 15:48.34 | Qwell | It'll use whatever your routing table says |
| 15:48.39 | teknoprep | ok |
| 15:48.44 | [TK]D-Fender | teknoprep, more than fine |
| 15:48.44 | teknoprep | that answers that then |
| 15:48.52 | WIMPy | Two different nets if ok. |
| 15:51.24 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 15:55.23 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:55.23 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:55.35 | teknoprep | ahh i figured it out |
| 15:55.45 | teknoprep | i have the asterisk box over a VPN connection |
| 15:55.48 | teknoprep | must be a routing issue |
| 15:55.58 | teknoprep | now that i have put the asterisk box in a virtual device on the same network |
| 15:56.01 | teknoprep | everything is ok |
| 15:56.07 | teknoprep | i'll figure out the routing issues later |
| 15:56.22 | teknoprep | this is just for testing so i will not be keeping everything inside vmware |
| 15:56.46 | teknoprep | pFsense has issues with routing a remote LAN to the local ip address of the firewall over IPSEC |
| 15:56.53 | *** join/#asterisk cstachris (~chrismylo@202.182.147.115) |
| 16:00.57 | teknoprep | i am getting one way audio... audio from internal to external is working.. but not the other way |
| 16:01.07 | teknoprep | is there a way to see how the call is being established ? |
| 16:01.13 | teknoprep | other than wireshark ? |
| 16:01.21 | wdoekes2 | sip set debug on |
| 16:01.23 | wdoekes2 | rtp set debug on |
| 16:02.41 | *** join/#asterisk tnt (~tnt@osmocom/tnt) |
| 16:03.43 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 16:05.34 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 16:13.45 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 16:13.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:18.08 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 16:19.10 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:19.10 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:21.08 | teknoprep | its all routing issues over IPSEC |
| 16:21.13 | teknoprep | wasn't the dial plans |
| 16:23.40 | *** join/#asterisk dirty (~johnny@70.99.103.122) |
| 16:28.18 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 16:34.25 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 16:34.28 | cusco | hi |
| 16:35.13 | cusco | if one was to connect about 30 rj11 phones to asterisk? |
| 16:35.23 | cusco | I guess I would need about 30 fxo ports, right? |
| 16:35.51 | tzafrir | right |
| 16:36.26 | teknoprep | do i still need to define the sip nat settings for an asterisk box behind a firewall ? |
| 16:36.51 | teknoprep | localnet and externaaddr |
| 16:37.41 | cusco | is there hardware recomended for this? |
| 16:38.36 | *** join/#asterisk timahvo1 (~rogue@197.176.22.46) |
| 16:44.09 | [TK]D-Fender | cusco, FXS, not FXO |
| 16:44.19 | *** join/#asterisk paryl (~par@168.215.170.202) |
| 16:44.27 | [TK]D-Fender | teknoprep, if it has to talk to the outside world, yes |
| 16:45.20 | cusco | ow, right |
| 16:45.37 | cusco | gah.. client has a alcatel easy lynq, in his hotel. I didn't quite understand the cabling, phones are all rj11, but they all connect with rj45: http://ovh.tretas.eu/~cusco/img/a.jpg |
| 16:46.41 | *** join/#asterisk teknomega (~tek@cpe-static-bluecloudcons-rtr.cmts.eph.ptd.net) |
| 16:48.55 | [TK]D-Fender | You can plug a RJ-11 pug into an RJ-45 jack... |
| 16:49.38 | [TK]D-Fender | cusco, and for 30 I'd grab a larger gatweway device for this |
| 16:49.50 | teknomega | patton makes a nice gateway |
| 16:49.51 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:50.14 | teknomega | i believe they have one which has 32 ports on it |
| 16:50.24 | teknomega | what is the best way to track down one way audio issues |
| 16:50.31 | teknomega | i have my external ip address set corectly |
| 16:50.38 | teknomega | i have my internal ranges set |
| 16:50.46 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:51.24 | teknomega | when adding multiple internal subnets... do i just add multiple localnet= address |
| 16:51.31 | teknomega | or do i put them on the same line somehow |
| 16:51.59 | [TK]D-Fender | multiple lines |
| 16:52.05 | teknomega | ok |
| 16:52.13 | [TK]D-Fender | PB your configs masking only passwords |
| 16:52.23 | [TK]D-Fender | ~pb |
| 16:52.24 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 16:52.26 | [TK]D-Fender | ^ |
| 16:55.03 | teknomega | http://pastebin.com/bdYX9tD3 |
| 16:55.32 | MrTelephone | teknomega, what kind of audio issues? |
| 16:55.55 | MrTelephone | You need to seriously think about upgrading servers every 2-3 years |
| 16:55.58 | teknomega | one way... outbound from my polycom phone works... inbound from cell to my desk phone does not |
| 16:55.59 | *** join/#asterisk l2trace99 (~jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 16:56.23 | MrTelephone | teknomega, cell -> pstn -> asterisk -> desk phone? |
| 16:56.32 | MrTelephone | what are the issues though. Echo is hard to beat |
| 16:56.59 | teknomega | desk phone -> PBX -> Asterisk -> NexVortex (ITSP - SIP) -> Cell Phone |
| 16:57.07 | teknomega | no echo .. i am getting one way audio |
| 16:57.14 | teknomega | i posted my configs they are very simple |
| 16:57.23 | [TK]D-Fender | teknomega, disable canreinvite/directmedia |
| 16:57.51 | [TK]D-Fender | teknomega, quality=no <- and that's supposed to be "qualify=yes |
| 16:58.06 | [TK]D-Fender | well .. not yes/no ... just QUALIFY |
| 16:58.09 | [TK]D-Fender | (spelling) |
| 16:58.29 | [TK]D-Fender | teknomega, Also, nexvortex should be "nat=no" |
| 16:58.38 | MrTelephone | I type in voicemail show users and asterisk console crashes |
| 16:58.53 | teknomega | i do not have a wan addres on this box |
| 16:59.07 | *** join/#asterisk adeel (~adeel@72.53.65.233) |
| 16:59.51 | [TK]D-Fender | teknomega, And you need your WAN IP in the externhost field (which you don't have there) |
| 17:00.01 | [TK]D-Fender | oops, missed the newer one below |
| 17:00.05 | [TK]D-Fender | Scratch that |
| 17:00.14 | p3nguin | Register statements can only go in the general section, too. |
| 17:00.24 | [TK]D-Fender | Ok, fix the rest and you should be OK with all else being equal |
| 17:00.35 | teknomega | where is this directmedia you are talking about ? |
| 17:00.35 | [TK]D-Fender | p3nguin, Correct. |
| 17:00.44 | [TK]D-Fender | teknomega, absent. That is the problem. |
| 17:00.55 | [TK]D-Fender | teknomega, put it in [general] and your peers |
| 17:01.12 | teknomega | general under extensions.conf ? |
| 17:01.25 | [TK]D-Fender | No, sipconf |
| 17:01.28 | teknomega | ok |
| 17:04.30 | MrTelephone | I installed libodbc in '/usr/local/lib/odbc/libmyodbc.so instead of /usr/local/lib. How do I relink the libraries to the alternate location? |
| 17:04.40 | [TK]D-Fender | ldconfig |
| 17:05.14 | teknomega | audio works |
| 17:05.51 | [TK]D-Fender | teknomega, The problem is once the call tried to set up * let your phone and provider try to get in direct communication with each other |
| 17:05.59 | [TK]D-Fender | teknomega, that = fail. |
| 17:06.05 | teknomega | ok |
| 17:06.18 | teknomega | but why would i not do directmedia on the other contexts ? |
| 17:06.37 | [TK]D-Fender | teknomega, Your phone doesn't know your WAN IP and when it tells the provider "send audio to IP 1.2.3.4" it'll be the LAN IP it has.... which is of course non-contactable |
| 17:06.50 | [TK]D-Fender | teknomega, I did suggest yuo put it in all peers |
| 17:07.28 | p3nguin | Calls between a device outside the NAT and a device inside the NAT should always keep asterisk in the media path to deal with the internal/external addressing problem. |
| 17:08.06 | p3nguin | Calls between devices on the same LAN as asterisk never passing the NAT can usually use directmedia safely. |
| 17:08.09 | teknomega | we are not using Asterisk for the PBX functionality |
| 17:08.18 | teknomega | so NAT is compensated on the internal PBNX |
| 17:08.20 | teknomega | PBX |
| 17:08.30 | [TK]D-Fender | teknomega, its a grey line as to when you call something a "PBX" |
| 17:08.38 | teknomega | ok |
| 17:08.51 | teknomega | features are not being done by asterisk |
| 17:08.57 | [TK]D-Fender | teknomega, My * makes me coffe and is my jukebox.... |
| 17:08.58 | teknomega | they are being done by freeswitch |
| 17:09.07 | [TK]D-Fender | coffee* |
| 17:09.25 | MrTelephone | tkD-fender, did you have to compile odbc yourself or did you use the distribution package? |
| 17:09.31 | TriJetScud | Lolokol |
| 17:09.37 | teknomega | how do i allow for refers and reinvites to work on the [eZuce] peer |
| 17:10.39 | *** join/#asterisk elkng (~elkng@unaffiliated/elkng) |
| 17:13.13 | MrTelephone | The distribution package is working now |
| 17:13.22 | MrTelephone | It wasn't a couple years ago |
| 17:13.23 | MrTelephone | lol |
| 17:13.32 | p3nguin | You would have to have another peer involved in a call with eZuce for a reinvite to even be possible in the first place. |
| 17:14.03 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 17:15.18 | *** join/#asterisk gusto (~gusto@88.128.164.124) |
| 17:15.39 | elkng | is asterisk alternative for skype ? |
| 17:16.00 | p3nguin | Nope. |
| 17:16.05 | teknomega | the eZuce context is the server that our phones are connected to |
| 17:16.11 | teknomega | so .. it goes like this |
| 17:16.21 | [TK]D-Fender | MrTelephone, I always use the distro provided packages and works fine. I did install in manual ONCE about 5-6 years ago maybe while still running slackware |
| 17:16.35 | teknomega | polycom phone -> eZuce -> Asterisk -> ITSP -> PSTN |
| 17:16.48 | teknomega | p3nguin, there is nother peer involved |
| 17:17.25 | p3nguin | If you are going from inside the NAT to outside the NAT and the call goes through asterisk, you don't want to reinvite. |
| 17:17.40 | [TK]D-Fender | elkng, It can be part of such a solution |
| 17:18.55 | teknomega | p3nguin, ok |
| 17:19.01 | teknomega | p3nguin, its al inside the same network |
| 17:19.27 | p3nguin | Not from what you just said. |
| 17:19.37 | p3nguin | ITSP = Internet ...... |
| 17:21.33 | teknomega | the reinvite would come from eZuce |
| 17:21.39 | teknomega | the phone would put a call on park |
| 17:22.15 | teknomega | park is an extension on the ezuce box... so if i transfer the call to a parking lot extension that is not at the phone... do i not have to reinvite the call to the location of the parking lot ? |
| 17:23.45 | p3nguin | A reinvite, for most purposes of discussion, causes the call to go directly between two peers. What does that have to do with a parking lot? |
| 17:24.22 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:24.23 | teknomega | p3nguin, how does a call get moved from one sip location to another sip location |
| 17:24.30 | teknomega | a reinvite no ? |
| 17:24.45 | p3nguin | I don't know what you mean by a "sip location." |
| 17:25.22 | teknomega | 101@fqdn.com |
| 17:25.26 | teknomega | sip locations |
| 17:25.42 | teknomega | i have a call at 101@fqdn.com... i want to transfer it to... 102@fqdn.com |
| 17:25.47 | teknomega | a reinvite will do this no ? |
| 17:26.01 | teknomega | that is the nature of SIP |
| 17:26.18 | p3nguin | no |
| 17:26.34 | teknomega | no? |
| 17:26.37 | p3nguin | You can transfer the call without doing reinvite. The transfer will have an invite. |
| 17:27.34 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 17:30.11 | teknomega | i can't transfer call's when asterisk is the gateway for my PBX |
| 17:30.20 | teknomega | i know it might not be a PBX.... |
| 17:33.46 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 17:37.03 | *** join/#asterisk Champi (Champi@rootshell.fr) |
| 17:41.53 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 17:43.45 | *** join/#asterisk rhg412 (~harrisong@68-73-223-152.ded.ameritech.net) |
| 17:53.03 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 17:54.10 | rhg412 | anyone want to help a newb connecting a D70 to his asterisk server? |
| 17:55.34 | Qwell | rhg412: Are you using DPMA? |
| 17:57.04 | Qwell | rhg412: Please don't message people without asking first. To answer your question, however, yes. You really aught to be using it. |
| 17:57.57 | Qwell | It makes all the setup rather trivial, and gives you a bunch of awesome stuff. |
| 17:58.03 | *** join/#asterisk ThinkGNU- (~ThinkGNU-@216.67.183.210) |
| 17:58.40 | rhg412 | Qwell thanks for the suggestion |
| 17:59.12 | *** join/#asterisk wtf911 (~wtf911@cpe-67-251-103-137.stny.res.rr.com) |
| 18:00.07 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 18:00.22 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:02.18 | *** join/#asterisk imcdona (~imcdona@c-71-227-200-25.hsd1.wa.comcast.net) |
| 18:13.09 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 18:17.27 | MrTelephone | my voicemail server drive filled up and mysql failed to respond to queries and it left asterisk in a deadlock |
| 18:17.32 | MrTelephone | wow |
| 18:20.07 | MrTelephone | Why would mysql not respond because of disk full. |
| 18:20.55 | *** join/#asterisk mik0 (~Andrey@88.87.93.132) |
| 18:22.45 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 18:25.45 | MrTelephone | wow, lesson learned guys. Monitor disk usage on your odbc servers |
| 18:26.26 | bdfoster | that's why you partition your drives (programs on one, data on the other) |
| 18:27.43 | MrTelephone | It was some cheap backup script I was using |
| 18:29.17 | MrTelephone | The path was on the programs partition |
| 18:33.27 | wtf911 | hmm perhaps someone isn't overly busy and would be willing to take some time to help me out? ...i can explain what i want to do...i'm open tousing teamviewer and what not to help set me up :O |
| 18:34.54 | p3nguin | Describe your goals. |
| 18:35.46 | MrTelephone | to establish a multimillion dollar enterprise using asterisk |
| 18:36.21 | wtf911 | No just to get cool free calling lol... |
| 18:36.30 | wtf911 | Well in a nutshell I would like to get this setup... http://www.rootsecure.net/?p=reports/callerid_spoofing |
| 18:37.51 | wtf911 | I've already made a http://www.ipkall.com/ account and voipwise.com account (i could always change this one if it can't forward the caller id info?) to hopefully have the outgoing/incoming covered |
| 18:37.55 | p3nguin | That page has a lot of information on it, but it does not describe any particular concept. |
| 18:38.44 | p3nguin | You simply want to be able to send the caller id number of your choice? |
| 18:39.29 | MrTelephone | Man I get a lot of overruns from telco compared to local channel bank. |
| 18:39.45 | MrTelephone | Would it be crap service? |
| 18:39.49 | wtf911 | Yes...that and have free calls...it would be an upgrade coming from $30 a month for only local calls and no caller-id from verizon (yes stupid i know) |
| 18:40.02 | p3nguin | Ask your termination provider if they allow you to pass your own caller id number. |
| 18:40.41 | wtf911 | I would be running asterisk from my router running optware(ipkg) to have it running 24/7 and once setup I would buy a SIP phone |
| 18:41.14 | *** part/#asterisk tnt (~tnt@osmocom/tnt) |
| 18:43.42 | wtf911 | I would have installing asterisk taken care of. I was hoping someone here wouldn't mind taking some time to help me set asterisk up though. :) |
| 18:45.43 | *** join/#asterisk scgm11 (~Sebastian@r186-52-2-57.dialup.adsl.anteldata.net.uy) |
| 18:45.45 | p3nguin | ~book |
| 18:45.45 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 18:47.13 | wtf911 | :/ i was hoping someone would be able to give some hands on help specific to my scenario ... cough cough anyone who wouldn't mind? |
| 18:50.36 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:50.44 | anonymouz666 | where I can find the CEL .sql structure? |
| 18:51.04 | *** part/#asterisk scgm11 (~Sebastian@r186-52-2-57.dialup.adsl.anteldata.net.uy) |
| 18:52.13 | *** join/#asterisk Neptu (~Neptu@c-6c96e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 19:04.32 | ThinkGNU- | what is a good brand of small ata? I'm looking for something that is maybe 2 FXS ports and doesn't need a router built-in |
| 19:05.03 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 19:05.20 | paulc | ThinkGNU- I've always liked the PAP2, cheap and cheerful, works well |
| 19:05.37 | ThinkGNU- | I'm hearing a lot about that one |
| 19:05.51 | ThinkGNU- | A company I used to work for used those too |
| 19:05.57 | ThinkGNU- | I'll have to get one of those |
| 19:06.10 | ThinkGNU- | anything you'd choose as a second option? |
| 19:06.27 | paulc | They're pretty good, super configurable.. the Obihai devices are worth checking out too, but I don't think they have a 2 port unit yet.. (or do they? something just got released with a built in router, might have been 2 FXS no FXO) |
| 19:06.58 | paulc | Obihai is from the guys who were at Sipura before they got bought by Linksys/Cisco.. so they know what they're doing.. |
| 19:07.25 | ThinkGNU- | I'll check that out too |
| 19:07.31 | ThinkGNU- | just need a couple of options |
| 19:08.11 | teknomega | what is the simplest way to route all calls that come into the asterisk box from the ITSP to a specific extension or context |
| 19:09.44 | *** join/#asterisk ZenMaster (~zenmaster@unaffiliated/auckla) |
| 19:11.17 | ZenMaster | 1 Willshire. :) |
| 19:11.19 | ZenMaster | What a place. |
| 19:11.27 | ThinkGNU- | I'm no expert here but I was thinking about just sending an incoming call to an extension teknomega |
| 19:11.37 | ThinkGNU- | maybe someone can expand on that idea or others |
| 19:11.50 | teknomega | i was thinking the same thing |
| 19:11.57 | teknomega | i am using asterisk as a B2BUA |
| 19:12.27 | ThinkGNU- | sorry, B2BUA? |
| 19:12.29 | teknomega | we will be putting many of them all over the coutry to allow for outbounds calls to directly go out the local B2BUA gateway instead of having to traverse RTP traffic over MPLS and VPN connections |
| 19:12.34 | teknomega | back to back user agent |
| 19:12.35 | kaldemar | teknomega: callbackectension in the register statement to itsp and context option in a matching peer definition. |
| 19:12.37 | ThinkGNU- | I'm terrible with acronyms |
| 19:13.06 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:14.27 | wtf911 | if anyone wants to lend me a hand i have vnc open... server: wtf911.hopto.org pass:help |
| 19:15.31 | *** join/#asterisk capt-rogers (~IceChat77@mail1.decisioningsolutions.com) |
| 19:16.40 | capt-rogers | is there anything in asterisk that controls audio volume? my voip phone volume is low..even if i max the volume on the sip phone |
| 19:17.41 | kaldemar | capt-rogers: for voip, nothing but func VOLUME |
| 19:20.22 | capt-rogers | Function volume..is that inside asterisk? |
| 19:20.32 | *** join/#asterisk timahvo1 (~rogue@197.176.22.46) |
| 19:20.45 | kaldemar | capt-rogers: yes, it is a dialplan function |
| 19:21.25 | capt-rogers | hmm..and it controls the default volume on my voip phones? I need to check that... |
| 19:22.16 | ZenMaster | This is an error I am getting trying to connect Xlite softphone to an asterisk box. http://pastebin.com/y5nhn19z |
| 19:23.27 | p3nguin | capt-rogers: It changes the amplitude on a channel. |
| 19:24.18 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:37.40 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:40.38 | ThinkGNU- | how about a good conference phone? |
| 19:40.52 | ThinkGNU- | does anyone have any suggestions on brand? |
| 19:42.51 | *** join/#asterisk gavimobile1 (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 19:43.10 | gavimobile1 | I need one of the admins to please unban me.. gavimobile |
| 19:43.48 | ThinkGNU- | I'm looking at a polycom soundstation ip 5000 and 6000 |
| 19:44.46 | gavimobile1 | had a power outage today now my sip commands aren't working |
| 19:47.27 | ZenMaster | I am having a hard time trying to find where I add a user to asterisk? |
| 19:47.34 | ZenMaster | Or am I just crazy. |
| 19:47.43 | gavimobile1 | ======= |
| 19:48.03 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 19:48.07 | gavimobile1 | I tried running core restart now, but still sip commands aren't working |
| 19:49.07 | gavimobile1 | ZenMaster: why do you mean? don't you mean you want to assign a user for running asterisk? |
| 19:49.20 | *** join/#asterisk twanny796 (~twanny@46.11.51.90) |
| 19:49.34 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 19:49.59 | *** join/#asterisk chazcafe (~chazcafe@cmr-208-124-160-122.cr.net.cable.rogers.com) |
| 19:50.19 | ZenMaster | gavimobile1: No, I am trying to connect xlite softphone to asterisk which is already running. I am getting a error about the user not being in my sip.conf. |
| 19:50.35 | ZenMaster | Thus I am trying to figure out how to solve that issue. |
| 19:50.39 | ZenMaster | All via CLI. |
| 19:51.04 | gavimobile1 | ZenMaster: think peer is the proper term rather than user |
| 19:51.15 | gavimobile1 | but go ahead and add the peer in to your sip settings |
| 19:51.42 | ZenMaster | I have never messed with telephony. I was just asked to try to make sense of this. Apparently I am going to have to add another book to my book shelf now. :D |
| 19:52.12 | gavimobile1 | ZenMaster: im still a noon, but yes your going to need to read |
| 19:52.15 | ZenMaster | gavimobile1: I posted a pastebin link to my error up above. |
| 19:52.22 | gavimobile1 | its not something that someone can explain to you in a sentence |
| 19:52.41 | gavimobile1 | search google for the definitive guide 3rd edition asterisk pdf |
| 19:52.47 | gavimobile1 | download that |
| 19:53.26 | p3nguin | zenmaster: You don't add users. If you want to add a phone, create a peer entry in sip.conf for it. |
| 19:53.27 | gavimobile1 | and read the section about the sip.conf |
| 19:53.38 | p3nguin | ~book |
| 19:53.39 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:54.06 | gavimobile1 | if your not interested in reading, then you may want to consider using freepbx or pbxiaf |
| 19:54.15 | gavimobile1 | you can add peers from the gui |
| 19:54.21 | p3nguin | That pastebin was empty from what I saw. |
| 19:54.26 | ZenMaster | gavimobile1: This is not the case. Please don't instantly label me. |
| 19:54.29 | ZenMaster | Thank you for the link. |
| 19:54.46 | gavimobile1 | ZenMaster: sorry budd! |
| 19:54.50 | gavimobile1 | just trying to help |
| 19:54.58 | gavimobile1 | ;-) |
| 19:55.06 | gavimobile1 | but that's the book |
| 19:55.22 | gavimobile1 | it has great explinations |
| 19:55.44 | ZenMaster | gavimobile1: No worries. I do read..... A lot. As my coment up above was about adding another book to my book shelf. I am a systems admin. I just walked into this project today. :) |
| 19:56.09 | p3nguin | ~book |
| 19:56.09 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:56.09 | ZenMaster | Thank you and I also think I found what I was looking for finally via google. I am tasked with trying to get a dial tone today. |
| 19:56.18 | gavimobile1 | ZenMaster: that's great!! do it the right way rather than using freepbx |
| 19:56.34 | gavimobile1 | that way you'll learn how to solve problems on your own and won't reach a dead end |
| 19:56.46 | ZenMaster | Of course. |
| 19:56.47 | ZenMaster | :D |
| 19:56.51 | gavimobile1 | :-) |
| 19:57.39 | gavimobile1 | before I reload my sip channel driver, is there something you folks can recommend to me why my sip commands aren't working ie. sip show peers |
| 19:57.52 | gavimobile1 | I tried running core restart now |
| 19:57.55 | p3nguin | module show like sip |
| 19:58.04 | *** join/#asterisk elkng (~elkng@unaffiliated/elkng) |
| 19:58.35 | gavimobile1 | it says 1 module loaded |
| 19:58.43 | p3nguin | chan_sip.so ? |
| 19:58.50 | gavimobile1 | p3nguin: yes |
| 19:58.55 | p3nguin | module unload chan_sip.so |
| 19:58.55 | p3nguin | then |
| 19:58.58 | p3nguin | module load chan_sip.so |
| 19:59.05 | p3nguin | Watch the console for errors. |
| 19:59.44 | gavimobile1 | from the cli? its not unloading |
| 20:00.03 | gavimobile1 | Unable to unload resource chan_sip.so |
| 20:00.28 | p3nguin | When you did core restart now, did it actually restart? |
| 20:00.41 | gavimobile1 | yes, according to my knowledge |
| 20:00.43 | p3nguin | core show channels |
| 20:00.52 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:00.55 | p3nguin | Maybe there is a call right now. |
| 20:00.55 | gavimobile1 | 0 active channels |
| 20:01.01 | gavimobile1 | 0 active calls |
| 20:01.04 | p3nguin | Bizarre. |
| 20:01.10 | gavimobile1 | 0 calls in progress |
| 20:01.29 | gavimobile1 | maybe I should reload from a backup? |
| 20:01.44 | gavimobile1 | maybe it wasn't the power outage and I have an error in one of my config files? |
| 20:01.46 | p3nguin | I would probably stop asterisk, set noload on chan_sip, start up asterisk, then manually load chan_sip after everything else is up. |
| 20:02.04 | gavimobile1 | lets try |
| 20:02.07 | gavimobile1 | service asterisk stop |
| 20:02.51 | gavimobile1 | how do I set noload on chan_sip |
| 20:02.58 | gavimobile1 | I need to disable the module from the config? |
| 20:03.18 | gavimobile1 | from modules.conf? |
| 20:03.42 | gavimobile1 | yes, that's what you mean.. k one sec |
| 20:04.36 | gavimobile1 | I don't need to reload anything if doing this when asterisk is stopped I assume |
| 20:04.48 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 20:05.28 | p3nguin | When asterisk is stopped, there is nothing to reload. Everything has already been removed from memory. |
| 20:05.47 | gavimobile1 | its a no go |
| 20:06.00 | p3nguin | When you manually tried to load chan_sip, what happened? |
| 20:06.09 | gavimobile1 | I got 4 lines |
| 20:06.11 | gavimobile1 | let me pastebin |
| 20:06.31 | gavimobile1 | its like it failed at first than It looks like it loaded but it didn't |
| 20:06.37 | michael-i | Was wondering if anyone ran into problems with Playtones() when using G.722. It seems Asterisk forces a 8000 sample rate in the ast_playtones_start function. Playtones() isn't reliable after >=3500hz |
| 20:07.25 | gavimobile1 | p3nguin: http://pastebin.com/TYfJhLSj |
| 20:08.05 | gavimobile1 | noload => chan_sip.so this was in my modules.conf |
| 20:08.17 | p3nguin | Wow, so overly descriptive and clear. :/ |
| 20:08.21 | gavimobile1 | meaning, that's what I added |
| 20:08.39 | p3nguin | It gave absolutely no indication of what is wrong, why it could not load. Just that it failed to load. |
| 20:08.49 | gavimobile1 | ? |
| 20:08.55 | p3nguin | the paste |
| 20:08.57 | gavimobile1 | maybe my hd is failing? |
| 20:09.06 | gavimobile1 | or bad ram? |
| 20:09.14 | p3nguin | I would not conclude that based on this single problem. |
| 20:09.30 | gavimobile1 | it happened a power otage for a split second |
| 20:09.33 | p3nguin | Do you have a backup of sip.conf? |
| 20:09.40 | p3nguin | Do you have a backup of chan_sip.so? |
| 20:09.47 | gavimobile1 | p3nguin: yup, no |
| 20:09.51 | gavimobile1 | yes to sip.conf |
| 20:10.04 | p3nguin | Start with comparing the current sip.conf to the backup. |
| 20:10.28 | gavimobile1 | no to chan_sip.so, unless chan_sip.so is locaed in the download dir /usr/src/downloads/1.8/bla bla bla |
| 20:10.33 | p3nguin | Don't forget that you have set a noload for chan_sip.so. That could cause a headache later. |
| 20:10.46 | p3nguin | If you did not clean the source tree, it should still be there. |
| 20:11.02 | gavimobile1 | ok... I don't think its sip.conf cause id didn't play with sip today, I did play with extentions.conf |
| 20:11.07 | gavimobile1 | extensions.xonf |
| 20:11.11 | gavimobile1 | extensions.conf |
| 20:11.23 | gavimobile1 | you think it may be corrupted? |
| 20:11.55 | p3nguin | sip.conf and extensions.conf are completely unrelated. |
| 20:12.47 | gavimobile1 | ill rename sip.conf to sip.conf.old and move a copy of my bu of sip.conf to the correct location |
| 20:13.02 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 20:13.36 | p3nguin | I would also compare checksums of the chan_sip.so which is in place and the one saved in the source tree. |
| 20:13.43 | p3nguin | md5sum |
| 20:15.10 | gavimobile1 | ok your first diagnostic method worked |
| 20:15.18 | gavimobile1 | p3nguin: the sip bu worked just fine |
| 20:15.31 | p3nguin | Oh, good. |
| 20:15.35 | gavimobile1 | im going go over it.. maybe some text got moved rather than edit |
| 20:15.37 | gavimobile1 | edited |
| 20:15.42 | p3nguin | Take care of that noload you set during testing. |
| 20:15.44 | gavimobile1 | I use nano |
| 20:15.52 | gavimobile1 | p3nguin: will do friend! |
| 20:15.58 | gavimobile1 | thanks for your great assistance |
| 20:16.07 | p3nguin | You can compare the two files using diff or sdiff. |
| 20:16.21 | gavimobile1 | p3nguin: naaa, ill go over it manually |
| 20:16.25 | gavimobile1 | the bu is quite old |
| 20:17.11 | p3nguin | I would start with sdiff -s sip.conf sip.conf.old, but that's just me. |
| 20:17.24 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:18.45 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 20:20.26 | gavimobile1 | again! thanks for the help |
| 20:20.39 | fprior | Hi, I'm testing TE121 card with loopback connector. In CLI appear "PRI Error on span 1: We think we're the CPE, but they think they're the CPE too.", can I ignore this message or is better change conf ? |
| 20:21.40 | paulc | fprior: If you both think you're CPE, things won't work right... so it's best changed I reckon ;-) |
| 20:24.58 | jpsharp | You can't really loopback test a PRI like that. |
| 20:25.59 | teknomega | i am having a hard time understanding what i need to do to route ALL CALLS that come from my ITSP to a single extension |
| 20:26.11 | teknomega | outbound was easy |
| 20:26.50 | jpsharp | You can loopback test it insofar as you can tell the transmitter and receiver in the card work, but that's abou tit. |
| 20:27.44 | fprior | jpsharp, this is what I would, test E1 card as "standalone", is possible ? |
| 20:28.12 | p3nguin | teknomega: The context setting for the peer controls where calls from the peer will go. |
| 20:28.47 | p3nguin | teknomega: The extension in the register statement (sometimes) determines which extension calls will go to. |
| 20:28.48 | jpsharp | fprior: Well, the fact that it is trying to bring up the ISDN layer and seeing itself says the card is functioning. |
| 20:29.24 | p3nguin | teknomega: If the ITSP sends calls to you at various extensions, you'll have to create a pattern that matches all possible extensions they call. |
| 20:29.29 | jpsharp | fprior: So that's about as far as you can get with testing without either an actual telco line or a second E1 port. |
| 20:29.39 | teknomega | p3nguin, ahh ok... so i just create a context=foo inside the peer controls for the ITSP |
| 20:29.50 | teknomega | p3nguin, [foo] then routes calls for me |
| 20:30.07 | p3nguin | teknomega: Calls matching the defined peer would obey the context value. |
| 20:30.34 | teknomega | p3nguin, what would be the syntax for ALL DID's or ALL CALLS |
| 20:30.36 | teknomega | . |
| 20:31.06 | p3nguin | teknomega: Like I said: you would have to create a pattern which matches all the extensions that they send calls to. |
| 20:31.27 | fprior | jpsharp, I would like test dialplan, is possible ? |
| 20:31.38 | p3nguin | teknomega: A _possible_ pattern to match all extensions might be _X. for the pattern. |
| 20:31.41 | teknomega | doesn't . match everything ? |
| 20:31.49 | teknomega | p3nguin, ty |
| 20:31.59 | p3nguin | teknomega: Don't use a dot. It's bad and usually wrong. |
| 20:32.24 | p3nguin | teknomega: _X. will require at least one digit followed by one or more characters after. |
| 20:32.32 | jpsharp | fprior: Not with single ended ISDN. You have to be talking to something else other than yourself. |
| 20:33.17 | fprior | jpsharp, thank you |
| 20:33.37 | teknomega | p3nguin, exten => _X.,1,Dial(SIP/eZuce/${EXTEN}) |
| 20:33.40 | teknomega | does that look right ? |
| 20:34.16 | ZenMaster | p3nguin: So I have a quick question. |
| 20:34.27 | ZenMaster | I am reading on how to do what I needed earlier. |
| 20:34.41 | ZenMaster | told me to do secret=pwd_ivan |
| 20:35.02 | ZenMaster | Does that mean the password is "pwd_ivan" "ivan" or "_ivan". For instance. |
| 20:35.05 | p3nguin | teknomega: yes |
| 20:35.25 | p3nguin | zenmaster: pwd_ivan is the plaintext password. |
| 20:35.57 | jpsharp | b |
| 20:36.12 | ZenMaster | keeps telling me wrong password. |
| 20:36.20 | ZenMaster | Maybe I mistyped. |
| 20:36.23 | p3nguin | Then you supplied the wrong password! |
| 20:36.39 | teknomega | p3nguin, |
| 20:36.42 | teknomega | p3nguin, ty |
| 20:36.55 | ZenMaster | p3nguin: Do you have to restart the asterisk service anytime you make changes to sip.conf? |
| 20:37.04 | p3nguin | I guess there is a possibility that you gave the right password but the wrong user or auth ID. |
| 20:37.08 | p3nguin | zenmaster: sip reload |
| 20:37.21 | p3nguin | in the Asterisk CLI, of course |
| 20:39.20 | ZenMaster | p3nguin: I am doing this from a Terminal. :) |
| 20:39.30 | *** join/#asterisk acedia (~garland@unaffiliated/ffs) |
| 20:39.42 | p3nguin | asterisk -r |
| 20:39.55 | p3nguin | then: sip reload |
| 20:40.12 | p3nguin | Learn to use the asterisk CLI early -- you'll spend a lot of time there later. |
| 20:41.08 | teknomega | p3nguin, i do miss the asterisk CLI when i work in freeswitch |
| 20:41.16 | teknomega | even tho freeswitch has a CLI i like the other one better |
| 20:41.54 | ZenMaster | p3nguin: Roger roger. |
| 20:42.18 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 20:42.21 | p3nguin | When you change sip.conf, sip reload. When you change extensions.conf, dialplan reload. |
| 20:42.31 | ZenMaster | Roger. |
| 20:42.39 | ZenMaster | Now I am getting username authname mismatch. |
| 20:44.55 | ZenMaster | Hmm I am wondering is there an easy way to run a asterisk system test enviroment? |
| 20:47.20 | Chainsaw | You deploy an identical box that you don't connect to the phone system? |
| 20:51.14 | leifmadsen | use a virtual machine? |
| 20:51.30 | leifmadsen | (which you could easily deploy with chef or puppet, which woudl be the same system you use to deploy asterisk) |
| 20:51.37 | ZenMaster | Hmmm I think this is a virtual machine. it is at 1wilshire blvd. |
| 20:52.00 | leifmadsen | finally! I now know where the cloud lives! |
| 20:52.02 | ZenMaster | Well I love *nix. Just wondering if I could easily set something up in my test enviroment at home. |
| 20:52.03 | leifmadsen | attacks |
| 20:52.09 | leifmadsen | yes? |
| 20:52.15 | ZenMaster | Go for it... |
| 20:52.25 | ZenMaster | ;{) |
| 20:52.45 | leifmadsen | https://github.com/kickstandproject <-- relevant |
| 20:55.02 | ZenMaster | leifmadsen: I think that single website has just solved all my lifes problems. |
| 20:55.13 | *** join/#asterisk jsjc (~Adium@185.Red-83-59-179.dynamicIP.rima-tde.net) |
| 20:55.36 | leifmadsen | ZenMaster: pabelanger has been working really hard on it for the last 6 months and is now deploying into production to several customers |
| 20:55.37 | ZenMaster | The ultimate goal of this project I have walked into is also to get these 8x8 phones to work also. The brand is polycom. I have done some research into reflashing them to work with Asterisk. |
| 20:56.01 | ZenMaster | Well we have a GUI front end supposidly been in development for over two years. |
| 20:57.35 | teknomega | would an intel atom CPU suffice for a 10 call volume B2BUA |
| 20:57.48 | leifmadsen | without transcoding, I'd say likely |
| 21:01.47 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:02.58 | capt-rogers | Jim signed my Asterisk book the other day in TLUG Toronto Asterisk Presentation...we signed it Welcome to the Dark Side! Yes, I would much rather learn Asterisk from CLI as opposed to FreePBX |
| 21:06.10 | *** join/#asterisk timahvo1 (~rogue@197.177.64.64) |
| 21:07.00 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-196.mobile.uci.edu) |
| 21:18.54 | *** join/#asterisk kenshin (~kenshin@67.55.36.159) |
| 21:32.43 | *** part/#asterisk gavimobile1 (~user@bzq-84-108-109-1.cablep.bezeqint.net) |
| 21:38.04 | *** join/#asterisk dirty (~johnny@70.99.103.122) |
| 21:52.22 | *** part/#asterisk rhg412 (~harrisong@68-73-223-152.ded.ameritech.net) |
| 21:59.24 | *** join/#asterisk wonderworld (~ww@dsdf-4d0a151e.pool.mediaWays.net) |
| 22:04.50 | *** join/#asterisk talntid (~t@216.229.186.226) |
| 22:05.15 | talntid | http://pastebin.com/eH5aBKW3 |
| 22:05.43 | talntid | I know I'm doing something wrong. the recording I am playing to them, says "press 1,2,3" etc... but it just hangs up at the end of playback, as opposed to waiting for input.. |
| 22:08.39 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 22:10.20 | *** join/#asterisk Ice_Strike (~Ice_Black@94-192-112-241.zone6.bethere.co.uk) |
| 22:11.11 | Ice_Strike | hi |
| 22:13.51 | *** join/#asterisk Bullmoose (~Bullmoose@65-129-15-106.bois.qwest.net) |
| 22:15.54 | Ice_Strike | I have asterisk installed on VPS |
| 22:16.43 | Ice_Strike | *sometime* DTFS is not accepting properly and IVR is not recognizing the DTFS sound maybe |
| 22:17.18 | Ice_Strike | How can I find out if it problem with VOIP provider or VPS is not good enouh? |
| 22:19.04 | Ice_Strike | DTMF* |
| 22:25.36 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-wbtcsvddlndnbcjf) |
| 22:26.38 | ZenMaster | Ok so I seem to have learned quite a bit here about asterix. |
| 22:27.24 | ZenMaster | Now, I have an 8x8 phone that I want to connect to my asterix. I can clearly edit the config of this 6755i via web based admin. Just not sure what settings to fill in to make it connect to my asterix. |
| 22:27.28 | ZenMaster | server... |
| 22:34.09 | *** join/#asterisk nickgaw (~nick@70-139-55-30.lightspeed.hstntx.sbcglobal.net) |
| 22:35.19 | nickgaw | Hi, Where in sip.conf should I go to enter in my sip server information as I am trying to configure my asterisk server to be a sip phone for another extention I have on a system so I can dial that extention and get forwarded to my local asterisk system? |
| 22:40.27 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:54.06 | p3nguin | zenmaster: You learned so much about asterisk that you think it is "asterix" I guess. |
| 22:54.33 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 22:56.45 | citywok | p3nguin: he learned enough to join the #asterisk channel to ask about #asterix |
| 22:57.39 | citywok | and he got a packet8 phone with a different firmware and came here to ask about how to hack it to work with asteri"x" |
| 22:57.50 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 22:58.03 | Ice_Strike | I have asterisk installed on VPS |
| 22:58.06 | Ice_Strike | *sometime* DTFS is not accepting properly and IVR is not recognizing the DTFS sound maybe |
| 22:58.14 | dirty | wonders where obelix would be? |
| 22:58.17 | Ice_Strike | How can I find out if it problem with VOIP provider or VPS? |
| 22:58.19 | citywok | Ice_Strike: DTMF? |
| 22:58.27 | Ice_Strike | Yea soz DTMF* |
| 22:59.42 | p3nguin | What dtmfmode are you using for the peer where the call comes from? |
| 23:00.25 | Ice_Strike | dtmfmode=inband |
| 23:00.37 | Ice_Strike | I have tried rfc2833 and it didnt work for some reason |
| 23:01.32 | Tim_Toady | inband is going to work only with g711, and still might have troubles if you have jitter or packet loss |
| 23:02.36 | michael-i | I'm having some trouble getting Playtones() in 10.2.1 to playback 4000-5000hz tones correctly to a g722 connected phone. It seems that ast_playtones_start() in indications.c forces a sample rate of 8000. |
| 23:02.37 | citywok | packetloss/jitter can be an issue with a VPS as well |
| 23:02.41 | Tim_Toady | rfc2833 is your best bet, maybe u didnt got your phones configured to use it |
| 23:02.47 | michael-i | Could that be the issue? |
| 23:03.13 | WIMPy | michael-i: Quite possible. What Asterisk version? |
| 23:03.19 | p3nguin | I always use rfc2833 for everything I configure. |
| 23:03.24 | michael-i | WIMPy: 10.2.1 |
| 23:04.28 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 23:04.56 | michael-i | I recompiled indications.c with a sample rate of 16000 to test. Things don't breakdown as before but the tones are wrong: 4000 sounds like 1000, etc. |
| 23:11.10 | Ice_Strike | Tim_Toady I am using ulaw with DTMF - yes I do have problem with jitter or packet loss I think |
| 23:11.50 | Tim_Toady | Ice_Strike: i would give rfc2833 another try |
| 23:11.50 | Ice_Strike | sometime it work and sometime it dont |
| 23:11.57 | Ice_Strike | I will do |
| 23:13.02 | Ice_Strike | Is it possible to monitor jitter or packet loss ? |
| 23:13.13 | Ice_Strike | while DTMF are sending |
| 23:14.08 | Tim_Toady | you can monitor it for each call |
| 23:14.41 | Ice_Strike | listening you mean? |
| 23:15.40 | Tim_Toady | you can get that info from the ${CHANNEL(rtpqos,audio,all)} variable |
| 23:17.03 | Ice_Strike | Tim_Toady My VOIP provider in USA and asterisk making a call to UK (IVR) |
| 23:17.11 | Ice_Strike | Do you think that might be an issue? |
| 23:18.05 | Tim_Toady | i dont know |
| 23:18.13 | Ice_Strike | Like this? Noop(${CHANNEL(rtpqos,audio,all)}) |
| 23:18.32 | Tim_Toady | Ice_Strike: https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL |
| 23:19.29 | Ice_Strike | ok cool |
| 23:21.04 | Ice_Strike | Tim_Toady NoOp("SIP/test-00000539", "ssrc=1309628886;themssrc=0;rxjitter=0.000000;rxcount=0;txcount=0;") |
| 23:21.36 | Ice_Strike | rfc2833 dont work properly |
| 23:24.00 | wtf911 | Connected to Asterisk 1.6.2.22 currently running on unknown (pid = 1501) |
| 23:24.00 | wtf911 | [May 14 23:22:55] NOTICE[1509]: chan_sip.c:20792 handle_request_invite: Call from 'xlite1' to extension '16071111111' rejected because extension not found in context 'default'. |
| 23:24.07 | wtf911 | can anyone help me fix that error? |
| 23:24.46 | Ice_Strike | wtf911 you don't have 16071111111 extension in context 'default' |
| 23:25.26 | wtf911 | i have this... |
| 23:25.27 | wtf911 | [default] |
| 23:25.27 | wtf911 | Dial(SIP/voipstunt/{EXTEN}@voipstunt) |
| 23:25.35 | wtf911 | in my extensions.conf |
| 23:25.36 | Ice_Strike | no |
| 23:26.25 | Ice_Strike | exten => 16071111111,1, Dial(SIP/voipstunt/{EXTEN}@voipstunt) |
| 23:26.53 | wtf911 | ah so what would be a good way to set that to dial any number? |
| 23:27.04 | Ice_Strike | in fact shouldnt it be... Dial(SIP/{EXTEN}@voipstunt) |
| 23:27.22 | wtf911 | yes! you're good lol...i changed it to try making it work |
| 23:27.41 | Ice_Strike | exten => _XXXXXXXXXXX,1, Dial(SIP/{EXTEN}@voipstunt) |
| 23:27.48 | Ice_Strike | X is how many digits allowed |
| 23:28.06 | wtf911 | what's the _ in the beginning for if i may ask? |
| 23:29.07 | Ice_Strike | wtf911 see http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns |
| 23:30.47 | *** join/#asterisk BekarfelASUS (~here@173.85.204.173) |
| 23:31.02 | Ice_Strike | Tim_Toady any other suggestion I should do? |
| 23:31.58 | p3nguin | exten => 16071111111,1, Dial(SIP/voipstunt/{EXTEN}@voipstunt) <--------- This is wrong. |
| 23:32.20 | p3nguin | exten => 16071111111,1,Dial(SIP/voipstunt/${EXTEN}) <--------- This isn't wrong. |
| 23:32.31 | Ice_Strike | p3nguin I did correct him about that |
| 23:32.41 | p3nguin | But you corrected it incorrectly. |
| 23:32.47 | p3nguin | Therefore it wasn't corrected. |
| 23:33.12 | Ice_Strike | Ah |
| 23:33.31 | p3nguin | wtf911: THe underscore indicates an extension pattern. |
| 23:33.57 | Ice_Strike | I am getting really mad with dtmf :/ |
| 23:34.13 | wtf911 | thank you guys :) |
| 23:34.16 | Ice_Strike | Any suggestion what I should do to debug |
| 23:34.27 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:35.47 | Ice_Strike | Might try different voip provider or get better vps |
| 23:36.10 | Ice_Strike | im running out of option to get dtmf to work correctly |
| 23:36.42 | BekarfelASUS | Ice_Strike: what are you trying to do with DTMF? |
| 23:37.15 | BekarfelASUS | not that I have any clue what I'd be talking about |
| 23:37.16 | BekarfelASUS | tbh |
| 23:37.21 | Ice_Strike | BekarfelASUS I am making a call to UK, it get answered right away via IVR |
| 23:37.26 | Ice_Strike | I am sending DTMF |
| 23:37.54 | Ice_Strike | but *sometime* IVR is not reconisging the digit that been sent |
| 23:40.22 | p3nguin | Are you calling that system from a phone or with some type of automation? |
| 23:40.54 | Ice_Strike | automation |
| 23:41.20 | Ice_Strike | I have also tried from softphone |
| 23:41.22 | Ice_Strike | same issue |
| 23:45.47 | BekarfelASUS | Ice_Strike: does it work with another system? |
| 23:46.13 | Ice_Strike | BekarfelASUS I have tried on other vps and yes same problem |
| 23:46.51 | BekarfelASUS | but you don't have a problem dialing the same numbers when placing the call? |
| 23:47.57 | Ice_Strike | BekarfelASUS dialing the same numbers via Voip you mean? |
| 23:49.07 | BekarfelASUS | ummm, I mean like, in the number in UK that you're calling, say there's a 1 that the IVR doesn't recognize 50% of the time, the 1 is always recognized when you dial the IVR's number? |
| 23:51.25 | Ice_Strike | BekarfelASUS it is very random |
| 23:51.48 | Ice_Strike | i have tried calling other number and send DTMF |
| 23:51.51 | Ice_Strike | also same problem |
| 23:52.08 | Ice_Strike | when i dial from my mobile or BT - it work fine |
| 23:58.23 | wtf911 | i have outgoing calls working thanks to you guys :D ... |
| 23:58.51 | wtf911 | now i've tried to change it to this... |
| 23:59.00 | wtf911 | exten => _XXXXXXXXXXX,1,Set(CALLERID(num)=2095551214) |
| 23:59.00 | wtf911 | exten => _XXXXXXXXXXX,1,Dial(SIP/voipstunt/${EXTEN}) |
| 23:59.06 | wtf911 | and i get call failed |
| 23:59.32 | wtf911 | i'm running version 1.6 and i do have the module loaded...i have logging on and get no errors or warnings or anything |
| 23:59.39 | Ice_Strike | change second number 1 to n |
| 23:59.48 | Ice_Strike | exten => _XXXXXXXXXXX,n,Dial(SIP/voipstunt/${EXTEN}) |
