| 00:01.16 | *** join/#asterisk scardinal (~supreme@0905ds1-rdo.0.fullrate.dk) |
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| 00:03.35 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 00:05.26 | nny | if I have dtmfmode=rfc2833 in my sip.conf is there any reason why the SIP provider would tell me that it is doing |
| 00:05.26 | nny | DTMF inband, even though I have stated otherwise in my sip.conf |
| 00:08.19 | p3nguin | You might not have specified the dtmfmode in the peer, or you might have specified something other than rfc2833 on the ITSP's side. |
| 00:08.56 | nny | p3nguin: yeah assuming as such, the peers all have dtmfmode directly in sip.conf |
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| 00:09.31 | nny | p3nguin: wouldn't RFC2833 show up as a packet in a tcpdump/wireshark output |
| 00:10.37 | nny | p3nguin: like this: http://www.atlantiatech.com/DTMF_1224.png |
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| 00:20.20 | nny | god damn this is frustrating |
| 00:20.29 | nny | ITSP says it's the system using inband |
| 00:20.54 | nny | nothing* (polycomm configs, sip show peer XXX, etc) nothing says it is, all are specifically stating dtmfmode=rfc2833 |
| 00:30.42 | p3nguin | What does sip show peer yourpeer say about it? |
| 00:33.22 | nny | p3nguin: rfc2833 |
| 00:33.36 | nny | p3nguin: for both the phones, the ITSP and in the sip.cfg for the phones themselves |
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| 01:16.47 | *** join/#asterisk andrecourchesne (~chatzilla@mail.net-forces.com) |
| 01:17.39 | andrecourchesne | Hi all, I need to write an asterisk module that will modify a call audio stream. Is the application the best approach or the function one? |
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| 01:18.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 01:22.20 | andrecourchesne | Anyone ever did an application that uses manipulate_callback ? |
| 01:28.10 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 01:43.08 | *** join/#asterisk keith4_ (~keith@unaffiliated/keith4) |
| 01:45.44 | keith4_ | is there a "proper" way to only allow an extension to ring during business hours? i found some half-assed posts on the digium forums, but this feels very unclean: http://forums.digium.com/viewtopic.php?f=1&t=10273 |
| 01:46.16 | keith4_ | maybe just using GotoIfTime? |
| 01:51.19 | Corydon76-dig | Yep, gotoiftime to route directly to voicemail during after hours |
| 01:55.43 | keith4_ | ok, thanks |
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| 02:08.50 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 02:11.01 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
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| 02:29.04 | Futnet_Jkenney | Good evening everyone! |
| 02:29.15 | Futnet_Jkenney | Has anyone got hints to work from realtime? |
| 02:30.15 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:31.29 | keith4_ | uh, be sure to read the wiki pages first? |
| 02:35.51 | Futnet_Jkenney | I looked at the wiki however i cannot seem to see where realtime is covered |
| 02:35.59 | Futnet_Jkenney | i have everything working out of realtime |
| 02:36.05 | Futnet_Jkenney | i am just unsure how to add the hint |
| 02:36.14 | Futnet_Jkenney | so asterisk can see it from the database |
| 02:36.41 | Futnet_Jkenney | I have hints working from text based dialplan |
| 02:36.54 | Futnet_Jkenney | i would like to get them to work from the realtime based dialplan |
| 02:39.17 | *** join/#asterisk M0rg0th (~myself@bl10-131-178.dsl.telepac.pt) |
| 02:43.12 | ChannelZ | quit now and cake will be served immediately |
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| 04:22.33 | *** join/#asterisk QubeZ (~nkasu@236.6.103.97.cfl.res.rr.com) |
| 04:22.34 | QubeZ | hello all |
| 04:22.50 | QubeZ | I launched Asterisk with the -T option but it doesn't timestamp the console output. How can I achieve this? |
| 04:26.39 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
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| 04:38.19 | ChannelZ | did you actually launch the primary asterisk task with it, or are you using -r ? |
| 04:42.32 | smooth_penguin | hi |
| 04:45.31 | QubeZ | using -r |
| 04:51.04 | joobie | guys got an issue where when i dial out, the other person comes across really quiet on my end |
| 04:51.26 | joobie | im using a polycom 321 phone, connected via SIP to asterisk.. then the call routes out via pennytel to the 3rd party |
| 04:51.54 | joobie | i setup Monitor() on the asterisk server and when i listen to me talk, it's clear / loud,etc.. but when i listen to the other person talk with Monitor(), it's very quiet |
| 04:52.16 | joobie | apart from trying to get pennytel to fix this, is there something ic an do to improve quality? or improve loudness?? |
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| 05:06.37 | smooth_penguin | hey ChannelZ: |
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| 05:33.20 | ChannelZ | yesh |
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| 07:27.36 | verywiseman | i want to transfer call to another extension when there is no answer, how can i do that? |
| 07:27.58 | ChannelZ | Use a timeout value on Dial |
| 07:28.12 | ChannelZ | Then dial someone else, send them to voicemail, whatever |
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| 07:29.54 | schmidts | morning |
| 07:31.22 | ChannelZ | yawns |
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| 08:54.15 | aliverius | anyone using misdn on ubuntu? |
| 08:55.22 | *** join/#asterisk henk (~henk@netwichtig.de) |
| 08:55.36 | henk | hi |
| 08:56.47 | henk | what's the status of ael? should i use that? is it stable? |
| 08:57.55 | *** join/#asterisk sgimeno (~chatzilla@163.117.211.10) |
| 09:03.05 | russ | henk, I use ael |
| 09:03.17 | russ | it works well, but most of the documentation and examples out there are not in ael |
| 09:03.19 | *** join/#asterisk vgster (~vgster@94-194-190-189.zone8.bethere.co.uk) |
| 09:03.34 | russ | I enjoy it more than the old format |
| 09:03.49 | henk | russ: since when? does the syntax make sense? is there a good reference for the syntax? |
| 09:03.58 | henk | russ: noticed any oddities? |
| 09:04.05 | russ | I've used it for a couple months |
| 09:04.18 | russ | several things about the syntax I learned from the parser |
| 09:04.39 | henk | russ: 'from the parser' means you looked at the code and figured out what it expects? |
| 09:05.10 | russ | no, I started asterisk and the parser complained |
| 09:05.34 | henk | ah, trial and error, i like that, as long as the error messages are helpful. are they? |
| 09:06.06 | *** join/#asterisk rocknfra (~rocknfra@host64-225-dynamic.10-87-r.retail.telecomitalia.it) |
| 09:06.56 | russ | they tend to be |
| 09:07.01 | russ | http://pastebin.com/HS1hypTg |
| 09:07.30 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 09:07.54 | henk | russ: oh that does look good... thank you for the information, i'll give it a try and see if it works for me :) |
| 09:08.01 | russ | http://pastebin.com/C9QqGV32 |
| 09:10.14 | russ | semicolons are annoyingly important and you generally don't get any warning or error if you forget them |
| 09:11.35 | henk | hm, i guess that's ok, unrealircd, dhcpd, puppet, and bind act almost the same... they just say that there is a syntax error somewhere. as long as i know, i'll usually remember semicolons might cause that... thanks :) |
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| 09:23.32 | *** join/#asterisk freckle (~viperdude@viperdudeuk.broker.freenet6.net) |
| 09:24.28 | freckle | what is the best way to captute the Channel ID's of calls in progress? |
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| 09:55.43 | anny__ | hi all |
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| 10:03.55 | kaldemar | freckle: ${CHANNEL} |
| 10:04.51 | ectospasm | kaldemar: isn't that a dialplan function? Shouldn't it be ${CHANNEL()}? |
| 10:20.12 | freckle | kaldemar: yeah, thanks |
| 10:21.11 | kaldemar | ectospasm: it's a channel variable |
| 10:22.43 | kaldemar | ectospasm: the CHANNEL function offers all kinds of information on channels, but not the call id that you can parse from the channel name. |
| 10:22.59 | ectospasm | Ah, OK. Good to know. |
| 10:24.30 | ectospasm | ...where exactly is that documented? There isn't a CLI command "core show variables" |
| 10:26.11 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
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| 10:31.10 | DrDigital | can i ask about astrisknow here? |
| 10:32.38 | kaldemar | ectospasm: for example in the source package in doc/tex/channelvariables.tex |
| 10:32.57 | ectospasm | DrDigital: try #asterisknow |
| 10:33.19 | DrDigital | ill just idle their till someone answers then |
| 10:33.45 | jkroon | hi guys, it was my understanding that signalling bri_net_ptmp has been implemented in recent libpri versions, now with ast 1.6.2.11, libpri 1.4.11.4 and dahdi-{linux,tools} 2.4.0 I still get the "sucks for you" message ... any pointers? |
| 10:34.21 | kaldemar | or #freepbx if the question is related to generated asterisk configs or the GUI. |
| 10:34.57 | *** join/#asterisk rsp2k (~rpm@unaffiliated/rsp2k) |
| 10:35.25 | rsp2k | is it possible to stream a file while collecting DTMF digits? |
| 10:35.33 | rsp2k | I don |
| 10:35.51 | rsp2k | I don´t want the sound file to stop playing when the user starts to enter digits |
| 10:40.25 | jkroon | Background? |
| 10:40.29 | *** join/#asterisk oryxtec (~asda@119.158.81.218) |
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| 10:40.42 | oryxtec | can we block all traffic coming to my server on port 5060 udp and we can only allow some specific IP to come and register on 5060 udp port?? through IPtables?? |
| 10:41.23 | *** join/#asterisk datacompboy (~opera@l49-3-84.cn.ru) |
| 10:41.45 | datacompboy | Hi! Anybody knows what "Multi Pilot Number" means? |
| 10:42.09 | *** part/#asterisk ketema (~ketema@ketema.net) |
| 10:48.19 | *** join/#asterisk asilva (~structz@gandalf.ai.unesp.br) |
| 10:48.38 | asilva | Morning all! |
| 10:48.50 | rsp2k | background stops playing when digits are received |
| 10:50.05 | rsp2k | I am collecting quite a few digits and I want to play music to the user while they enter them. |
| 10:50.18 | jkroon | oryxtec, yes you can. |
| 10:51.09 | oryxtec | jkroon : how .. plz guide me IPtables command |
| 10:51.56 | jkroon | depends on your existing iptables configuration ... but changes are a double-rule on INPUT will do: iptables -A INPUT -s good.ip -j ACCEPT; iptables -A INPUT -p udp --dport 5060 -j DROP |
| 10:52.31 | rsp2k | is it possible to collect digits during MOH? |
| 10:52.51 | oryxtec | and if i want only few ip to use 5060 port on my sever in that case? |
| 10:55.00 | jkroon | oryxtec - then multiple ACCEPT rules before the drop? |
| 10:56.12 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 10:56.12 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 10:56.16 | oryxtec | how plz give me a hint |
| 10:56.24 | oryxtec | of complete command |
| 10:57.31 | jkroon | oryxtec, i already gave you everything you need. i'm not going to hold your hand. the above is two commands, just repeat the first one for each IP before you do the DROP one. |
| 10:57.40 | jkroon | or go read an iptables howto. |
| 10:57.57 | oryxtec | rite thanks |
| 10:58.01 | oryxtec | one question |
| 10:58.25 | oryxtec | if i block all udp ports then i wont be able to send and recive any trafic to my voip provider? |
| 10:59.33 | *** part/#asterisk datacompboy (~opera@l49-3-84.cn.ru) |
| 10:59.33 | jkroon | read the iptables man page, look at the state module. |
| 11:00.53 | oryxtec | thanks |
| 11:01.44 | asilva | Can someone help me out with this: When i have a voicemail alert on my ip telephone and i restart asterisk, the alert goes off, anyone knows how to fix this? |
| 11:03.36 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 11:04.23 | henk | i am atm testing my setup with asterisk connected to a cisco voice router. i have a softphone (linphone) and a mobile phone. when i call my softphone from the mobile and hang up, the softphone hangs up too. when i call the mobile from the softphone and hang up on the mobile, the softphone stays connected. what could cause that? |
| 11:08.09 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:08.15 | ectospasm | henk: what technology (DAHDI, SIP, etc.) is Asterisk connecting to the Cisco router on? |
| 11:08.57 | henk | ectospasm: it's all sip. using tcp for cisco and udp for the softphone. |
| 11:09.28 | ectospasm | henk: does a SIP trace show a BYE coming from Cisco? |
| 11:09.52 | henk | i have full access to all devices involved, asterisk server, firewall (which does/should not filter anything on the path cisco<->asterisk) and cisco router. |
| 11:10.24 | henk | ectospasm: hm, sip trace? sorry, never heard that. a simple 'sip set debug' on the asterisk cli? |
| 11:10.33 | ectospasm | henk: yes |
| 11:10.43 | henk | i'll check |
| 11:10.46 | ectospasm | henk: you may also want to collect a packet capture. |
| 11:10.52 | ectospasm | (using tcpdump) |
| 11:11.13 | asilva | or wireshark |
| 11:11.18 | asilva | gui view |
| 11:11.26 | ectospasm | essentially, you should see a BYE coming from Cisco when the mobile phone hangs up |
| 11:11.40 | ectospasm | if you do not, it's something at the Cisco level (or beyond) |
| 11:11.42 | henk | ok, i do when the mobile phone initiates the call, let's check the other way round... |
| 11:14.45 | henk | yes, nothing coming from the cisco it seems... |
| 11:17.21 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:19.42 | henk | does anyone happen to know what might be causing that on the cisco? |
| 11:20.01 | jkroon | borked firmware?!? |
| 11:20.30 | henk | on a cisco? impossible! |
| 11:21.03 | henk | jkroon: i'll keep it in mind, but since i'm not keen on upgrading, i'll try to rule out other possibilities first ;) thanks |
| 11:21.16 | jkroon | you'd be surprised some of the crap I've found in some seriously high end SIP systems. |
| 11:22.07 | jkroon | right, has anybody managed to get bri_net_ptmp working yet? |
| 11:23.48 | henk | jkroon: perhaps i'd be, but i've seen quite some broken stuff, so i'd probably just facepalm... :-p |
| 11:24.54 | jkroon | i think the worst yet is a device that receives a 404 response and then sends 180 preceding the 404 to the initiating device. |
| 11:37.52 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
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| 11:57.47 | asilva | Can someone help me out with this: When i have a voicemail alert on my ip telephone and i restart asterisk, the alert goes off, anyone knows how to fix this? |
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| 12:04.16 | asilva | ? |
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| 12:24.28 | Chainsaw | Where is the language set for Local/ channels please? |
| 12:31.12 | russellb | Chainsaw: that's a complicated question :-) |
| 12:31.24 | Chainsaw | russellb: It must be, seeing how I'm getting no answers. |
| 12:31.36 | russellb | it depends which side of the local channel you're talking about |
| 12:31.40 | Chainsaw | russellb: It's defaulting to en, which is wrong. I could do with a global langauge=en_GB that just works on everything. |
| 12:31.47 | russellb | and it's interesting you ask because we _just_ put in a patch related to this ... |
| 12:32.43 | russellb | You can set it in the dialplan executed in the Local channel, but in existing versions that only sets it on one side of the local channel |
| 12:32.58 | russellb | we just made a (relatively complicated) change to make it so when you set it in the dialplan, it gets set on both sides of the local channel |
| 12:33.06 | russellb | it will be in the next bug fix updates for all versions |
| 12:33.16 | *** join/#asterisk fofware (~fabian@186.124.148.94) |
| 12:34.02 | redax | hi |
| 12:34.56 | Chainsaw | russellb: Right, thanks. It's getting me on a divert. |
| 12:35.07 | redax | is the deny/permit lines in the sip.conf for a peer should work well? |
| 12:35.18 | Chainsaw | russellb: This "global language" setting that I'm referring to doesn't exist then? |
| 12:35.25 | redax | seems like I can register/originate a call even if I'm out of the permitted ip range... |
| 12:35.53 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 12:35.56 | [TK]D-Fender | redax: Wow, it's almost like it's RESTRICTING you to things you are permitted to do!@ |
| 12:36.51 | ectospasm | redax: what's your deny line look like? |
| 12:37.07 | redax | D-Fender, hm? am I wrong? deny=0.0.0.0/0.0.0.0 ; permit=192.168.10.0/255.255.255.0 |
| 12:37.36 | ectospasm | redax: and your insecure line? |
| 12:38.06 | [TK]D-Fender | redax: And your device is where? |
| 12:38.27 | *** join/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 12:38.38 | redax | no, I'm not on the lan |
| 12:38.42 | [TK]D-Fender | redax: and SHOW us your config |
| 12:38.44 | redax | ie. some internet address |
| 12:39.02 | [TK]D-Fender | redax: And the SIP DEBUG to match |
| 12:39.48 | redax | geez. |
| 12:40.14 | redax | forget me... I think I gonna kill my collegue... |
| 12:40.27 | redax | permit=192.168.10.0/0.0.0.0 |
| 12:40.47 | redax | sorry |
| 12:40.48 | [TK]D-Fender | SMRT |
| 12:41.04 | tzanger | haha |
| 12:42.23 | redax | not that funny... |
| 12:44.19 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 12:44.46 | hurdman | hi ! what's the better way to be sure that asterisk is running from an externel script ? |
| 12:45.03 | hurdman | to see if it's blocked |
| 12:45.08 | hurdman | or something like that |
| 12:45.36 | ectospasm | hurdman: normal usage has asterisk started by safe_asterisk |
| 12:46.11 | hurdman | ectospasm: and ? |
| 12:46.20 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 12:47.08 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:47.33 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:47.56 | t_dot_zilla | does anybody know how many phones asterisk can page simultaneously? ver. 1.6.1.20 |
| 12:48.37 | hurdman | hurdman: if it's dead it restart, but not if it"s blocked |
| 12:49.50 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:50.05 | *** join/#asterisk Zeeek (~anonymous@pdpc/supporter/active/zeeek) |
| 12:52.47 | [TK]D-Fender | hurdman: It'd be nice if you read his question. |
| 12:53.23 | [TK]D-Fender | t_dot_zilla: how fast can a car go? |
| 12:53.40 | hurdman | [TK]D-Fender: it's my question :) hurdman> hi ! what's the better way to be sure that asterisk is running from an externel script ? |
| 12:54.04 | hurdman | :) |
| 12:54.09 | [TK]D-Fender | hurdman: Listen to yourself! Can't you hear the voices? :p |
| 12:54.24 | t_dot_zilla | im just wondering if asterisk has a set limit for call paging |
| 12:54.26 | [TK]D-Fender | hurdman: And better than "what"? |
| 12:54.46 | [TK]D-Fender | t_dot_zilla: well a DIALPLAN LINE has a length limit.. and it isn't that big |
| 12:55.10 | [TK]D-Fender | t_dot_zilla: that could be considered a "limit" depending on your pOV |
| 12:55.11 | hurdman | [TK]D-Fender: i wan't to have an alert if my asterisk freeze |
| 12:55.35 | hurdman | [TK]D-Fender: i have tried to make a asterisk -rx "something" |
| 12:55.45 | hurdman | [TK]D-Fender: but it seems to be ugly |
| 12:55.47 | [TK]D-Fender | hurdman: thats a good start |
| 12:56.27 | hurdman | [TK]D-Fender: i'm looking for a better way :) |
| 12:56.37 | [TK]D-Fender | hurdman: You need to confirm that it responds. You can do taht. or an AMI call, or use a CLI client to place a SIP call etc and log the outcome. |
| 12:57.08 | [TK]D-Fender | hurdman: You need to confirm not just that a socket (etc) starts, but that * responds in a meaningful way. your current way sounds good |
| 12:57.17 | hurdman | ok |
| 12:57.22 | hurdman | thanks |
| 12:57.57 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-187-148.pools.arcor-ip.net) |
| 12:58.26 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-187-148.pools.arcor-ip.net) |
| 12:58.56 | *** join/#asterisk BANSAL (~bansal@117.207.83.20) |
| 13:03.22 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:04.18 | *** join/#asterisk nitram (foo@superblob.com) |
| 13:07.58 | *** join/#asterisk l2trace99 (~jr@74.118.40.1) |
| 13:13.46 | *** join/#asterisk vader-- (~me@c-71-225-201-226.hsd1.nj.comcast.net) |
| 13:15.35 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 13:15.53 | hurdman | [TK]D-Fender: ok, i put a call file into my /var/spool/asterisk who call an iax extension who create a file with a test number who is read by my monitoring daemon |
| 13:16.09 | hurdman | it seems to be ok :) |
| 13:18.43 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:20.55 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:27.26 | *** join/#asterisk Delido1983 (~Delido198@web.its-medienservice.de) |
| 13:35.05 | fors1 | hi. I'm trying to locate some "lost" phones of former emplyoees. I'm planning on letting the phone ring until someone picks it up, and then play a prerecorded message about handing the phone back. Problem is that after ~2 minutes, the call attempt is dropped with this error in cli: chan_sip.c: Re-invite to non-existing call leg on other UA. SIP dialog |
| 13:35.47 | *** join/#asterisk Abhishek_Singh (~root@210.212.20.75) |
| 13:36.47 | henk | my asterisk is connected to a cisco voice gateway via sip over tcp. when i place a call from my softphone via asterisk via cisco (everything sip) to a cellphone via a PRI line and hangup on the cell, the voice gateway sends no "BYE" afaict. does anyone happen to know how to fix that? |
| 13:37.37 | henk | [TK]D-Fender: i would still like to clarify the communication we've been having a few weeks ago. do you care? may i query you? |
| 13:37.54 | *** join/#asterisk windback (~quassel@200.68.97.73) |
| 13:39.15 | [TK]D-Fender | henk: does this need to be private? |
| 13:40.08 | Zeeek | rubs his hands together |
| 13:40.08 | henk | [TK]D-Fender: it doesn't need to be, it's just OT here, so i thought it to be better... |
| 13:41.08 | [TK]D-Fender | henk: OK |
| 13:42.02 | henk | [TK]D-Fender: ok as in "you're right, a query is better suited" or as in "ok, if it doesn't need be, let's just talk here"? |
| 13:42.22 | [TK]D-Fender | henk: if its OT and you want to PM it, sure |
| 13:42.56 | henk | [TK]D-Fender: hey, you still remember the situation i am talking about? |
| 13:43.00 | henk | d'uh |
| 13:43.11 | [TK]D-Fender | henk: Not a clue |
| 13:43.18 | henk | forgot the /msg |
| 13:45.16 | *** join/#asterisk x-ip (~lain@unaffiliated/x-ip) |
| 13:45.27 | x-ip | hi all =) |
| 13:46.14 | asilva | hi |
| 13:46.22 | asilva | Can someone help me out with this: When i have a voicemail alert on my ip telephone and i restart asterisk, the alert goes off, anyone knows how to fix this? |
| 13:47.17 | *** join/#asterisk oryxtec (~oryxtec@119.152.114.249) |
| 13:48.00 | oryxtec | hi all .. im using asterisk 1.4 .. and i am having dtmf issue on it.. please can any one guide me how to fix this issue? |
| 13:49.58 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 13:49.59 | [TK]D-Fender | oryxtec: Perhaps you could provide actual details... |
| 13:51.57 | oryxtec | for example if i call on any number on which there is IVR like press 1 for sale and press 0 for operator and if i press 1 or 0 that DTMF tone does not work |
| 13:52.41 | *** join/#asterisk jks (~jks@193.189.93.254) |
| 13:53.18 | [TK]D-Fender | oxYou have not sold us what you are CONNECTING to and how it talks to *, or what the caller is using, or anything at all... |
| 13:53.25 | [TK]D-Fender | oryxtec: ^ |
| 13:54.26 | oryxtec | i didn't get ur question sorry |
| 13:54.34 | oryxtec | can u repeat |
| 13:54.35 | oryxtec | plz |
| 13:55.56 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-yuubckylouesdkvx) |
| 13:56.14 | [TK]D-Fender | oryxtec: You have not sold us what you are CONNECTING to and how it talks to *, or what the caller is using, or anything at all... |
| 13:56.52 | n3hxs | sold = told ;) |
| 13:56.56 | oryxtec | ooh |
| 13:57.01 | oryxtec | :) |
| 13:57.17 | oryxtec | told i was thinking sold .. wht to sold :) |
| 13:58.17 | oryxtec | im connecting to asterisk using SIP eyebeam and i m dialing to a canada number |
| 13:59.01 | drmessano | Not with, TO |
| 13:59.07 | [TK]D-Fender | oryxtec: does eyebeam > * work? You need to break EACH leg down. |
| 13:59.18 | [TK]D-Fender | oryxtec: Also you haven't said how you are dialing OUT |
| 14:01.06 | oryxtec | yes eyebeam works with * and i m dialing out through voip provider. |
| 14:02.11 | [TK]D-Fender | oryxtec: Go prove each leg independently |
| 14:03.24 | oryxtec | sorry to say but i gave u all details .. i don't know wht else to explain :S |
| 14:04.31 | [TK]D-Fender | oryxtec: No you haven't. Not by a long shot |
| 14:06.12 | oryxtec | [TK]D-Fender: i m new to asterisk sorry but please ask me what more information do you need in order to understand my issue. |
| 14:08.21 | [TK]D-Fender | oryxtec: You haven't said what PROTOCOL you are using with your provider. Or the NAME of the provider. Or the CODECS being used. You have not shown us CONFIGS (PASTEBIN!). You have not shown us CALL DEBUG (PASTEBIN!). You have not confirmed what independent testing you have done. In short, you've told and shown us virtually nothing. |
| 14:08.23 | [TK]D-Fender | ~pb |
| 14:08.23 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:08.25 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 14:09.38 | oryxtec | now i understand .. let me provide u all details. plz hold |
| 14:09.54 | [TK]D-Fender | [10:03]<oryxtec>sorry to say but i gave u all details .. i don't know wht else to explain :S |
| 14:10.00 | [TK]D-Fender | [10:09]<oryxtec>now i understand .. let me provide u all details. plz hold |
| 14:10.08 | *** join/#asterisk zamba (marius@flage.org) |
| 14:10.11 | [TK]D-Fender | sighs |
| 14:10.43 | *** join/#asterisk jamko (~Jim@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 14:11.14 | zamba | this is a bit off-topic.. but i try here anyway, since i really don't know where else to ask.. i want to forward a call to a number that's on cisco callmanager to an extension on my asterisk server.. that works just fine.. but the problem is that the originating number (msn?) isn't displayed on my phone when the call is put through - only the main switchboard's number.. |
| 14:11.21 | zamba | i'm guessing this is a setting on cisco callmanager? |
| 14:11.40 | fauxalliance | rpid? |
| 14:11.55 | zamba | fauxalliance: was that for me? |
| 14:12.22 | [TK]D-Fender | fauxalliance: Decent guess. |
| 14:12.25 | fauxalliance | zamba, Remote-Party ID? |
| 14:12.36 | [TK]D-Fender | zamba: Perhas you should actually look at the call. |
| 14:12.40 | [TK]D-Fender | perhaps* |
| 14:12.49 | zamba | [TK]D-Fender: at the asterisk endpoint, you mean? |
| 14:12.51 | drmessano | What's this "look" ? |
| 14:12.57 | fauxalliance | cisco call manager is a whole other can of worms... |
| 14:13.12 | [TK]D-Fender | zamba: You mean the thing sitting between what you think has the information, and what should get it? NOVEL! |
| 14:13.25 | zamba | [TK]D-Fender: problem is i have no control over the callmanager, and i'm trying to convince the one administrating that, that this is in fact possible |
| 14:13.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:13.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:13.30 | [TK]D-Fender | fauxalliance: worms leads to trolling... |
| 14:13.39 | fauxalliance | or an itchy arse. |
| 14:13.43 | drmessano | zamba: Look at the call on the Asterisk CLI.. SIP Debug.. Staring at the phones will only make you go blind |
| 14:13.56 | *** join/#asterisk _zoom_ (~user@41.218.36.149) |
| 14:14.01 | jamko | Morning... If the mysql DB is on a different host than *, what would be entered for "dbsock" directory in res_mysql.conf? |
| 14:14.26 | *** part/#asterisk Delido1983 (~Delido198@web.its-medienservice.de) |
| 14:14.27 | [TK]D-Fender | JamYou wouldn't <-- that is for UNIX sockets |
| 14:15.06 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 14:15.39 | jamko | So just remove the line all together? |
| 14:15.53 | zamba | drmessano: the from field is the switchboard's main number |
| 14:16.08 | zamba | drmessano: so the culprit is the callmanager (which i basically knew all along) |
| 14:16.48 | fauxalliance | \o/ |
| 14:16.51 | TobSnyder | anyone tried asterisk and outgoing calls via ISDN to UMTS phones? |
| 14:16.57 | _zoom_ | hi, to what extent do think asterisk is good for voip wholesaler? |
| 14:16.59 | drmessano | I guess you're asking in the wrong place then |
| 14:17.27 | fauxalliance | _zoom_, depends on how badass your multi-tenancy skills are. |
| 14:17.34 | zamba | drmessano: yeah, that's why i labeled this as a bit off topic :p |
| 14:17.46 | russ | _zoom_, you could ask teliax since they have used asterisk and another pbx |
| 14:17.46 | zamba | drmessano: but hopefully there was some callmanager administrators lurking in here |
| 14:17.59 | drmessano | zamba: None that would ever admit it |
| 14:18.11 | zamba | hehe |
| 14:18.22 | zamba | but callmanger calls this rpid? |
| 14:18.48 | drmessano | zamba: You MAY stand a better chance asking if someone is using Microsoft OCS for their Enterprise VoIP |
| 14:18.53 | fauxalliance | asterisk does zamba.. google trustrpid |
| 14:19.04 | drmessano | zamba: or "does anyone use Windows here?" |
| 14:19.07 | fauxalliance | drmessano, are you cereal? |
| 14:19.54 | drmessano | Cisco is the devil.. D E V I L.. makers of the Crappy Call Manager and the U Can't 500 series systems |
| 14:20.24 | drmessano | "How can I ____ on a UC520?" "U Can't!" |
| 14:21.00 | zamba | hehe |
| 14:21.03 | *** join/#asterisk smooth_penguin (~root@59.95.0.250) |
| 14:21.13 | drmessano | You may be better off just replacing the entire thing with Asterisl |
| 14:21.16 | fauxalliance | step 1, remove anything with 'Monopoly of Cisco' on it. step 2) replace everything with FOSS. |
| 14:21.16 | drmessano | or ASTERISK |
| 14:21.41 | fauxalliance | or call Cisco... |
| 14:21.45 | drmessano | Yes, remove all MCOSH from the network |
| 14:21.56 | drmessano | Microsoft Cisco Or Some Shit |
| 14:22.00 | fauxalliance | dont forget the PIX |
| 14:22.13 | fauxalliance | and the SonicWall |
| 14:22.24 | drmessano | gah, meant to caps the H |
| 14:22.35 | fauxalliance | and last but not least, Avaya and Grandstream. |
| 14:22.39 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 14:22.39 | fauxalliance | gotta go. |
| 14:22.54 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 14:23.22 | drmessano | There was a guy recently giving away an almost new Avaya system because he couldn't afford the support contract. Switched to Asterisk. |
| 14:23.37 | drmessano | The End Is Nigh |
| 14:23.46 | [TK]D-Fender | [10:18]<zamba>but callmanger calls this rpid? <- Look. At. The. CALL |
| 14:23.53 | *** join/#asterisk dailylinux (~test@s21-06205.dsl.no.powertech.net) |
| 14:23.54 | *** join/#asterisk murdock_ut (~chatzilla@mail.kimballequipment.com) |
| 14:23.58 | fauxalliance | The RedHad install powering the Avaya ACD is free though drmessano |
| 14:24.02 | fauxalliance | s/d/t |
| 14:24.31 | *** join/#asterisk tgreer (~tso@pdpc/supporter/student/tgreer) |
| 14:24.41 | fauxalliance | The goon that SSH's to it from avaya... with our without your consent... thats priceless. |
| 14:24.50 | smooth_penguin | hey drmessano do you happen to have any syslogs for the SPA3102 :\ |
| 14:24.59 | tgreer | hi all... recommendations on some voip phones to go with my asterisk server? |
| 14:25.13 | drmessano | fauxalliance: I can manage a UC520 using Firefox on a *nix box.. is that FOSS? |
| 14:25.41 | fauxalliance | drmessano, only if it works ;-) |
| 14:25.45 | drmessano | smooth_penguin: No, i've never needed to debug an SPA-3102 to the point of Ad Nauseum. They all just WORK. |
| 14:26.12 | smooth_penguin | hahah |
| 14:26.30 | drmessano | smooth_penguin: But in your case, I pointed out several times that the SPA-3102 is KNOWN to NOT WORK with CID in India. |
| 14:26.36 | drmessano | So what do you need to debug? |
| 14:27.10 | smooth_penguin | drmessano: I want to see the debug log to check if it detects the caller id, like mine does |
| 14:27.20 | drmessano | of course mine does |
| 14:27.33 | drmessano | You could have just ASKED me if caller id works |
| 14:27.39 | drmessano | Yes, yes it does |
| 14:27.39 | smooth_penguin | nono |
| 14:28.13 | smooth_penguin | http://pastebin.com/Am0LZV5f |
| 14:28.15 | fauxalliance | smooth_penguin, India? DTMF CID! |
| 14:28.41 | smooth_penguin | Ive seen some logs where they had - "FXO:Digit=98xxxxxxx" |
| 14:28.44 | smooth_penguin | the entire number |
| 14:28.55 | smooth_penguin | not split up in different lines |
| 14:28.55 | *** join/#asterisk JimVanM (~jimvanm@bas1-toronto06-2925209635.dsl.bell.ca) |
| 14:29.00 | smooth_penguin | hence was wondering |
| 14:29.01 | drmessano | Yes, and his box is detecting the DTMF, but not processing it as being the callerID.. Which is the problem with the SPA-3102 in India |
| 14:29.10 | smooth_penguin | ok |
| 14:29.36 | drmessano | smooth_penguin: The DTMF to FSK box is $18 USD.. not sure what that was in Rupees |
| 14:29.50 | smooth_penguin | drmessano: yep ill be getting my hands on one this week |
| 14:30.01 | smooth_penguin | Im just annoyingly impatient :P |
| 14:30.13 | drmessano | You got part of that correct |
| 14:30.40 | smooth_penguin | :p |
| 14:30.52 | drmessano | We've already established that it doesn't work |
| 14:31.05 | drmessano | If you're bored, you can come wash my car |
| 14:31.06 | *** join/#asterisk veryhappy (~no@port-92-195-245-145.dynamic.qsc.de) |
| 14:31.12 | smooth_penguin | lol |
| 14:31.26 | tgreer | anyone used grandstream phones? |
| 14:31.30 | tgreer | looking at GXP2000 |
| 14:31.33 | fauxalliance | tries not to. |
| 14:31.48 | veryhappy | hi guys have a big problem im a newbie to asterisk and i dont know which username i have to insert into sip.conf to register my sip provider and call out |
| 14:32.07 | fauxalliance | veryhappy, the one the providor gave you |
| 14:32.17 | veryhappy | yeah its funny |
| 14:32.22 | veryhappy | i have 2 usernames |
| 14:32.31 | [TK]D-Fender | ~gs |
| 14:32.31 | infobot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 14:32.35 | [TK]D-Fender | ~grandstream |
| 14:32.35 | infobot | grandstream is probably the Yugo of VoIP hardware. Run... Run away now. Though, therealcircut says that they're not that bad. |
| 14:32.38 | [TK]D-Fender | tgreer: ^^ |
| 14:32.46 | fauxalliance | ^^^agreed |
| 14:32.54 | tgreer | [TK]D-Fender: recommendation for phones then? |
| 14:32.58 | fauxalliance | aastara |
| 14:33.00 | fauxalliance | sp? |
| 14:33.02 | [TK]D-Fender | tgreer: Polycom > All |
| 14:33.06 | *** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:33.10 | [TK]D-Fender | tgreer: Where are you located? |
| 14:33.42 | tgreer | [TK]D-Fender: UK |
| 14:33.48 | veryhappy | ah is it possible that i have to insert also 1und1/username@provider.de/username? |
| 14:33.50 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 14:34.44 | tgreer | why polycom > snom? |
| 14:34.53 | tgreer | veryhappy: yes |
| 14:34.56 | [TK]D-Fender | tgreer: In UK Linksys is the ebst bang/buck, and they are "decent" |
| 14:35.21 | veryhappy | ah |
| 14:35.28 | [Outcast] | anyone here working on the gtalk stuff |
| 14:35.31 | [TK]D-Fender | tgreer: Polycom is top-notch audio and build quality, and amongst the most stable. |
| 14:35.49 | veryhappy | i just wrote it without 1und1/ |
| 14:35.54 | veryhappy | ok thank you i will test it |
| 14:36.11 | nayfan | http://theoatmeal.com/comics/email_address |
| 14:36.12 | drmessano | [Outcast], Which "stuff"? |
| 14:36.26 | nayfan | sorry wrong room |
| 14:37.23 | [Outcast] | well I have got gtalk working from IM client, but I can't recieve GV calls that have been forward to my gtalk account. |
| 14:37.31 | [Outcast] | just if anyone else has tried |
| 14:37.32 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 14:38.00 | drmessano | [Outcast], I don't think that works or is going to work |
| 14:38.08 | tgreer | [TK]D-Fender: so i've been looking at snom all these years wrongly? :( |
| 14:38.08 | [Outcast] | why? |
| 14:38.28 | drmessano | [Outcast] I don't think the call setup is going to work over Jingle. |
| 14:39.14 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:39.15 | TobSnyder | when calling from SIP phone to a mobile phone (that is using UMTS), we don't hear each other. When calling via an analog telephone set to this mobile phone, everything is okay. Could this be a codec problem? |
| 14:39.15 | [Outcast] | if can get calls from the IM client, i should be that hard to modify the singling to make it work. I am not using jingle, I am using chan_gtalk and res_jabber |
| 14:39.20 | drmessano | [Outcast], Calls sent to the web gtalk client are not the same as being sent to xmpp/jingle |
| 14:39.49 | [Outcast] | agreed but there should be away to detech them and answer them |
| 14:39.58 | [TK]D-Fender | tgreer: Snom tends to be a poor value. In North America Polycom is very well priced and more reliable. Elswhere Snom is way more expensive than Linksys. It depends how your budget & needs relate to that |
| 14:40.13 | drmessano | [Outcast]: Maybe you don't understand.. They are not SENDING the calls to non-web clients |
| 14:40.21 | Naikrovek | cites the often stated: Polycom > * |
| 14:40.26 | murdock_ut | I like our Snoms. |
| 14:40.27 | drmessano | [Outcast]: Nothing to DETECT |
| 14:40.38 | [Outcast] | drmessano the call is being sent to my asterisk box |
| 14:40.52 | *** part/#asterisk wwalker (~wwalker@208.92.232.27) |
| 14:41.23 | [Outcast] | just have a little trouble with DTMF ATM |
| 14:41.23 | drmessano | and you see this where? |
| 14:41.27 | tgreer | [TK]D-Fender: ok i'm looking for an polycom comparison to the snom 320... suggestions? |
| 14:41.38 | [Outcast] | on my asterisk CLI |
| 14:41.47 | [TK]D-Fender | tgreer: What are your needs? |
| 14:41.52 | tgreer | [TK]D-Fender: pm |
| 14:42.02 | Katty | man i'm just havin a bum day |
| 14:42.15 | Naikrovek | so have a better day |
| 14:42.22 | Naikrovek | i give you permission to have a better day |
| 14:42.23 | drmessano | [Outcast]: I get call notifications, but there is no call setup there |
| 14:42.27 | Naikrovek | (sometimes that works...) |
| 14:43.34 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:43.35 | [Outcast] | I got I am using a someone's modified branch which get the RTP to stand up.....the singling is just a little off, so it try to screen the call everytime.... |
| 14:43.51 | smooth_penguin | hugs Katty |
| 14:44.10 | Zeeek | {{{Katty}}} |
| 14:44.12 | [Outcast] | need modify the singling slightly or need to get DTMF working |
| 14:44.54 | [Outcast] | doing a packet trace to see if I can tell the difference |
| 14:46.14 | TobSnyder | when calling from SIP phone to a mobile phone (that is using UMTS), we don't hear each other. When calling via an analog telephone set to this mobile phone, everything is okay. Could this be a codec problem? |
| 14:54.54 | *** join/#asterisk bons (~bons@office.nessus.at) |
| 14:55.14 | bons | hy guys |
| 14:56.24 | bons | i would like to create an additional small log file... |
| 14:56.33 | bons | can i do this natively or should i run an external script? |
| 14:58.00 | [TK]D-Fender | bons: To log WHAT? |
| 14:58.19 | oryxtec | [TK]D-Fender: sorry for delay.... regarding DTMF issue.. here are details i m using SIP protocol with the provider and my provider name is voippro. i m only using G711 codec. here is my config ... http://pastebin.com/ii6NV0Ts |
| 14:58.46 | *** part/#asterisk gego (~gego@b238085.customer.hansenet.de) |
| 14:58.54 | [TK]D-Fender | oryxtec: You didn't even set the mode and I don't see your failed call with SIP DEBUG either |
| 15:00.10 | bons | to log simple info about call like only: date, src, dest |
| 15:00.12 | oryxtec | ok 1 sec plz |
| 15:00.44 | [TK]D-Fender | bons: You can output a line to your own logging method in the dialplan before you dial out... |
| 15:01.04 | bons | [TK]D-Fender, nice this is what i was interested in |
| 15:01.09 | bons | so i can create my logfile |
| 15:01.14 | bons | and then output there from the dialplan |
| 15:01.15 | bons | ? |
| 15:01.53 | [TK]D-Fender | bons: Yes. Its yoru dialplan, you can do whatever you want |
| 15:01.56 | [TK]D-Fender | your* |
| 15:02.05 | *** join/#asterisk jamko (~Jim@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 15:04.37 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.141.156.dsl.dyn.forthnet.gr) |
| 15:05.02 | *** join/#asterisk scruz (3ead2990@gateway/web/freenode/ip.62.173.41.144) |
| 15:05.04 | bons | [TK]D-Fender, just to understand better, i can configure my own logfile in the log conf and then output to it thourgh the dialplan... which one is the log function to call in the dialplan? |
| 15:05.09 | scruz | hey all |
| 15:05.53 | [TK]D-Fender | bons: there is no "log function" per se. You code how you want to log. |
| 15:06.29 | [TK]D-Fender | bons: doing a System() call to a shell command. Or using ODBC/MySQL, etc to insert a record. Or whatever. |
| 15:06.48 | bons | eh |
| 15:06.59 | scruz | i'm having an issue with call files. i have MaxRetries specifed in my call file, but it isn't retrying after the subscriber doesn't pick up |
| 15:07.07 | bons | [TK]D-Fender, so this is calling external scripting... its not native logginf function on the diaplan |
| 15:07.23 | [TK]D-Fender | bons: * already has call logging. its called **CDR** |
| 15:07.35 | [TK]D-Fender | bons: And there is an entire chaper in THE BOOK dedicated to it |
| 15:07.38 | bons | [TK]D-Fender, I KNOW, i just want to create my own log |
| 15:07.39 | [TK]D-Fender | ~book |
| 15:07.39 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:07.52 | [TK]D-Fender | bons: Well if its YOUR log then it is CUSTOM and YOU have to do it |
| 15:08.07 | [TK]D-Fender | bons: Time to take ownership of yoru system |
| 15:08.11 | bons | [TK]D-Fender, and i was asking if i can do it natively in the dialplan or i have to write something external like i always did |
| 15:08.14 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:08.32 | [Outcast] | drmessano: can't trace forgot it is encrypted!!! |
| 15:08.51 | [TK]D-Fender | bons: As I asid you can use System() as a means of doing this. and as I said its a question of what you are logging TO |
| 15:08.57 | scruz | also, i specified that asterisk should wait 60 seconds before dropping the call, but again i'm being ignored on that point. might the issue be with my upstream provider? |
| 15:08.57 | [Outcast] | need to work on the dtmf I think |
| 15:09.11 | *** join/#asterisk javierj (~quassel@190.146.192.22) |
| 15:09.20 | bons | WHY asking informations on irc channels alway ends to say, there is a book, you have to take ownership bla bla |
| 15:09.30 | bons | i was just asking information of a possible solution |
| 15:09.36 | bons | not asking how to dial an extension |
| 15:09.40 | *** part/#asterisk tgreer (~tso@pdpc/supporter/student/tgreer) |
| 15:09.49 | bons | and if im here it means i didnt find anything useful related on the web |
| 15:09.58 | bons | [TK]D-Fender, thanks anyway |
| 15:11.44 | [TK]D-Fender | bons: Well you hacven't told us what you want to log TO |
| 15:12.05 | [TK]D-Fender | bonAm I going to show you 10 formats you don't care about hoping to get it right? |
| 15:12.16 | bons | ... |
| 15:12.30 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 15:13.05 | *** join/#asterisk luckyaba (~lucky@ip72-194-218-169.sb.sd.cox.net) |
| 15:14.29 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 15:14.43 | [TK]D-Fender | bons: well? |
| 15:15.15 | bons | nc |
| 15:15.26 | [TK]D-Fender | bons: ? |
| 15:16.46 | b11d` | TK..you should write a comic strip or something |
| 15:17.00 | *** join/#asterisk oryxtec (~oryxtec@119.152.45.250) |
| 15:17.13 | *** part/#asterisk bons (~bons@office.nessus.at) |
| 15:17.15 | *** join/#asterisk talcite (~matthew@69-165-129-78.dsl.teksavvy.com) |
| 15:17.45 | *** join/#asterisk hfb (~hfb@pool-98-119-37-144.lsanca.dsl-w.verizon.net) |
| 15:17.54 | oryxtec | [TK]D-fender; sorry my internet got disconnected i ve also placed asterisk cli logs please check http://pastebin.com/iRBLMjE4 |
| 15:21.22 | scruz | sorry to ask again: can an upstream provider override settings in a call file? |
| 15:21.42 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 15:21.45 | scruz | like MaxRetries and WaitTime, specifically |
| 15:23.55 | rubberneck | I am running Asterisk 1.6.2.1, when I use the r option with the dial application (provide ring to calling party) I have a problem on calls where there is a "this number has been changed" message, the caller will never get the message. if the r option is removed it will work normally. Have I found a bug? |
| 15:25.17 | rubberneck | Sorry forgot to add that this is an outgoing SIP channel. ex: Dial(SIP/${EXTEN}@gateway,,r) |
| 15:25.23 | *** part/#asterisk Zeeek___________ (~anonymous@pdpc/supporter/active/zeeek) |
| 15:27.23 | [TK]D-Fender | oryxtec: I do not see you testing SIP/CC100 alone. Or SIP/sipvoip. AND you didn't even SPECIFY the mode |
| 15:28.14 | [TK]D-Fender | b11d`: This guy.. not to fucking bright. You'd think the idea of "what fucking storage format should this 'log' be in" would have made its way across. |
| 15:28.40 | [TK]D-Fender | b11d`: Before I show him 20 SQL and 3 BDB storage methods that are meaningless trash as well |
| 15:28.43 | *** join/#asterisk nickzxcv (~nick@schmalenberger.us) |
| 15:29.18 | [TK]D-Fender | scruz: No. If they don't respond then * will do waht yuo tell it |
| 15:29.51 | oryxtec | [TK]D-Fender: which mode you are talking about? |
| 15:30.01 | [TK]D-Fender | oryxtec: oryxtec DTMFMODE |
| 15:30.24 | scruz | [ TK ]D-Fender: * where? |
| 15:30.25 | oryxtec | let me try that |
| 15:31.02 | oryxtec | i m going to try dtmfmode = rfc2833 |
| 15:31.04 | oryxtec | ok? |
| 15:32.04 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta5 (2010/09/05), 1.6.2.12 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 15:32.19 | leifmadsen | Asterisk 1.4.36 and 1.6.2.12 are now available! See http://www.asterisk.org for the release announcements. |
| 15:32.49 | Qwell | no u |
| 15:33.02 | Katty | glomps Qwell |
| 15:33.07 | Qwell | is glomped |
| 15:33.23 | scruz | [TK]D-Fender: here's the code that generates the call file: http://pastie.org/1160747 but the calls are not retried, nor does the system wait till 60s elapse before dropping the attempt |
| 15:33.33 | Qwell | Katty: how are we? |
| 15:34.20 | Katty | Qwell: last night, probably a 4. |
| 15:34.23 | Katty | Qwell: this morning probably a 6 |
| 15:34.29 | Katty | Qwell: feeling about a 7 or 8 now tho |
| 15:34.33 | Qwell | excellentish |
| 15:35.21 | scruz | [TK]D-Fender: sorry. forgot that * is short for ... |
| 15:35.34 | scruz | smacks his head |
| 15:35.44 | p3nguin | Can anyone successfully establish a call to 44 0845 072 7227? I think my ITSP still hasn't fixed the problem, but I can't check to see if the number is even working. |
| 15:35.56 | scruz | that's what i get for trying to multitask |
| 15:36.03 | Katty | i don't think uk is included in our pri |
| 15:36.08 | Katty | as free long distance |
| 15:36.14 | [TK]D-Fender | scruz: Your pastebin proves nothing.... |
| 15:36.28 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:36.35 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:36.44 | p3nguin | If I can verify that the number works, I can bitch at the ITSP again for not fixing their shit. |
| 15:36.56 | p3nguin | It's annoying. I need to call that company, but I can't. |
| 15:37.02 | russ | p3nguin, I'm not sure my international calling is working |
| 15:37.09 | russ | can you give me a working +44 number/ |
| 15:37.16 | scruz | i'm not trying to prove anything. i'm just saying that that is what the call file contains, but what i get on the phones is relatively different. |
| 15:37.18 | p3nguin | russ: Who is your ITSP? |
| 15:37.25 | javierj | Why when I call from a sip extension to a gsm adaptor through a dahdi channel and I hangup first my sip extension the call continues until far end hungup? That is, asterisk does not hangup the dahdi channel and it keeps busy even when I have finish the call from the sip end. |
| 15:37.29 | russ | teliax |
| 15:37.42 | p3nguin | russ: If you can give me a known-good UK number that I can try, that could help at least a bit. |
| 15:39.04 | russ | maybe +44 (0)20 7323 8299? |
| 15:39.19 | russ | don't forget about the tricky 0 |
| 15:40.07 | russ | I think my international calling rider request is stalled :( |
| 15:40.14 | oryxtec | [TK]D-Fender : i have tried with dtmfmode as well but still same |
| 15:41.34 | oryxtec | :( |
| 15:43.01 | oryxtec | ? |
| 15:44.31 | [TK]D-Fender | oryxtec: I see NOTHING. and " i have tried with dtmfmode" tells me nothing |
| 15:44.49 | [TK]D-Fender | oryxtec: And still no confirmation and proof of independent testing |
| 15:44.55 | p3nguin | russ: I dialed 00 44 020 7323 8299. It isn't working for me. If you can confirm I am using the correct number, I'll jump on the ITSP again. |
| 15:45.56 | p3nguin | I could actually use as many known-good UK land line numbers as I can get to list for them as NON-working in their system. |
| 15:47.19 | [TK]D-Fender | p3nguin: Who are you dialing that # through? |
| 15:47.26 | p3nguin | VoIP.ms |
| 15:47.42 | p3nguin | They aren't very helpful in the ticket I opened a month ago. |
| 15:48.00 | p3nguin | Like it's MY responsibility to tell them why their shit won't dial through to the UK. |
| 15:48.00 | [TK]D-Fender | p3nguin: then you're doing ti wrong... voip.ms is a North American provider and you need the normal north American # format for an international call |
| 15:48.18 | [TK]D-Fender | p3nguin: Its your responsibility to know how to format a number properly... |
| 15:48.20 | oryxtec | [TK]D-Fender: http://pastebin.com/edKpqf9U |
| 15:48.42 | p3nguin | I don't know what that meant. They accept both 011 and 00 as the international dialing prefix. How do YOU want me to dial the UK numbers? |
| 15:48.56 | [TK]D-Fender | p3nguin: 01144XXX |
| 15:49.04 | p3nguin | Tried. It fails. |
| 15:49.15 | p3nguin | Already described all that. |
| 15:49.22 | [TK]D-Fender | p3nguin: And use Microsoft's UK office as a test # |
| 15:49.32 | [TK]D-Fender | p3nguin: You could like... SHOW us or something :p |
| 15:49.40 | *** join/#asterisk jamko (~Jim@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 15:50.33 | p3nguin | There's nothing more to show than I already showed. I'll do it again and paste as soon as I can find that MS phone number. |
| 15:51.01 | [TK]D-Fender | oryxtec: oryxtec try dtmfmode=inband |
| 15:51.14 | oryxtec | ok |
| 15:51.16 | *** join/#asterisk espiceland (~erin@nat/digium/x-tfwxxjcylhqwronx) |
| 15:52.16 | [TK]D-Fender | oryxtec: And I still have no independent test. Go do it <- |
| 15:53.33 | p3nguin | 0844 800 2400 ... does this look like the right number for MS UK? |
| 15:55.06 | Chainsaw | p3nguin: You can't call those internationally. |
| 15:55.35 | Chainsaw | p3nguin: You'd have to use the saynoto870.com search to find the non-geographic equivalent. |
| 15:55.48 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 15:55.53 | p3nguin | it's ringing, but that's all it's doing. |
| 15:56.26 | oryxtec | [TK]D-Fender: thanks its working fine now.. :) |
| 15:57.03 | p3nguin | chainsaw: Do you have any suggestion of a WORKING UK landline phone number? |
| 15:57.16 | Chainsaw | p3nguin: Yes, I've already PM'd you one. |
| 15:57.30 | p3nguin | That would be why I didn't see it. |
| 15:57.41 | Chainsaw | p3nguin: Since you didn't reply, I didn't feel inclined to give you more of them. |
| 15:57.44 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 15:58.11 | p3nguin | I don't receive PMs, so I didn't know you said anything to me outside of this channel. |
| 15:58.44 | oryxtec | one last thing... please have http://pastebin.com/mDZW8kzq and tell me with my hardware which g729 codec is compatible?? |
| 15:59.05 | Chainsaw | p3nguin: Well you'd better fix that up, I'm not going to give out phone numbers in a channel with >250 users in it. |
| 15:59.09 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-187-148.pools.arcor-ip.net) |
| 15:59.12 | oryxtec | <PROTECTED> |
| 15:59.16 | Qwell | oryxtec: grab the benchg729 tool from downloads.digium.com |
| 15:59.25 | p3nguin | chainsaw: Try now. |
| 15:59.45 | oryxtec | http://asterisk.hosting.lv/ |
| 15:59.48 | oryxtec | fromthis website |
| 15:59.50 | p3nguin | I don't see why I wouldn't be able to dial UK numbers, though. |
| 15:59.59 | Qwell | oryxtec: This conversation is done. |
| 16:00.05 | oryxtec | ok |
| 16:00.12 | oryxtec | thanks |
| 16:00.19 | Qwell | You are not legally allowed to use that codec. |
| 16:00.32 | Qwell | NOBODY here is going to help you. (Yes, that is a mandate.) |
| 16:00.36 | Chainsaw | Qwell will come to your house. With his lawyer. |
| 16:00.38 | Chainsaw | Seriously, don't do it. |
| 16:01.53 | p3nguin | So if the UK fax number works, then all these other numbers are not completing for what reason? |
| 16:03.15 | Katty | illegally uses the codec to be a rebel. |
| 16:03.21 | Katty | ^- not really. |
| 16:03.24 | p3nguin | I do not understand what prevents me from connecting to certain UK numbers if _some_ numbers work correctly. |
| 16:03.27 | Katty | pretends to be a rebel, regardless. |
| 16:03.36 | Chainsaw | Hi Katty :) |
| 16:03.53 | p3nguin | chainsaw just gave me two that connected properly on the first try. And I can dial with 011 or 00, like I already mentioned. |
| 16:03.56 | Chainsaw | p3nguin: Do the numbers start with +44 8? |
| 16:04.08 | Katty | hey Chainsaw (= |
| 16:05.01 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-187-148.pools.arcor-ip.net) |
| 16:05.21 | p3nguin | I tried the Microsoft number (dialed 011 44 0844 800 2400), and the one I was trying to reach for a month (dialed 011 44 0845 072 7227). They both fail. |
| 16:05.38 | Chainsaw | p3nguin: Those are geographic numbers, not reachable internationally. |
| 16:05.48 | Chainsaw | p3nguin: Type them in here: www.saynoto0870.com/search.php |
| 16:05.52 | *** join/#asterisk Abhishek_Singh (~root@220.225.244.114) |
| 16:05.54 | Chainsaw | p3nguin: And use that to find the real underlying number. |
| 16:06.13 | p3nguin | include the 44 on the front? |
| 16:06.19 | Chainsaw | Replace it with a 0. |
| 16:06.29 | Chainsaw | You're trying to reach 0844 8002400. |
| 16:06.54 | Chainsaw | You should also leave out the 0 after the 44. |
| 16:07.24 | p3nguin | So what number, exactly, do you want me to search for on that site? |
| 16:07.40 | Chainsaw | p3nguin: 0844 8002400 |
| 16:07.49 | Chainsaw | p3nguin: 0845 0727227 |
| 16:08.13 | p3nguin | That search site does not appear to be what you think it is. |
| 16:08.22 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:08.28 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:08.47 | Chainsaw | p3nguin: First number, result: 01954 713950 |
| 16:09.01 | p3nguin | I search the number like you said, and I get things like Health O Meter and Samsung Pn50A550. |
| 16:09.02 | *** part/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:09.22 | p3nguin | Oh, www provides a different site. |
| 16:09.24 | Chainsaw | p3nguin: Second number has no results, so you can't call that. Ask them for a number that's reachable internationally, because 08xx and 09xx numbers simply aren't. |
| 16:09.25 | p3nguin | gah! |
| 16:09.35 | Katty | gives p3nguin boxing gloves |
| 16:09.52 | Chainsaw | p3nguin: You can pretty much write a pattern to deny them. +448 & +449 will never complete. Don't bother. |
| 16:10.18 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:11.18 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 16:11.30 | p3nguin | What if I connect up to my ITSP in London... would I then be able to dial the UK numbers as non-international numbers? |
| 16:11.41 | elliot98 | finds his place |
| 16:11.44 | Chainsaw | p3nguin: Possibly, yes. |
| 16:11.50 | Katty | gives elliot98 a cookie. |
| 16:12.05 | elliot98 | sniffs cookie |
| 16:12.14 | Katty | it's not electric. |
| 16:12.21 | p3nguin | 011 44 01954 713950 doesn't work for me, either. |
| 16:13.08 | *** join/#asterisk CunningPike (~CunningPi@204.239.12.183) |
| 16:13.13 | Katty | CunningPike: ohai |
| 16:13.23 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:13.33 | CunningPike | Katty: HAI! |
| 16:13.35 | p3nguin | I just see a re-order message and the call disconnects. |
| 16:13.49 | *** join/#asterisk Leddy (leddy-free@aqua.surgehost.net) |
| 16:13.54 | Chainsaw | p3nguin: As I said before, don't have a 0 after the 44. |
| 16:14.00 | CunningPike | Katty: How are you, my dear? |
| 16:14.01 | Katty | CunningPike: howrchutoday |
| 16:14.02 | elliot98 | where can I find a playback application can jump into the middle of a sound file? |
| 16:14.02 | Chainsaw | p3nguin: +44 1954 713950 |
| 16:14.08 | p3nguin | tries |
| 16:14.12 | elliot98 | nibbles on cookie |
| 16:14.22 | Katty | CunningPike: i have omnomnom nearly 32oz of caffeinated soda. |
| 16:14.41 | CunningPike | Katty: Pretty good thanks - just tidying up after our Asterisk upgrade |
| 16:14.43 | Katty | CunningPike: so my giddy levels are on par with about 3 shots of jager |
| 16:14.48 | CunningPike | I haz Americano |
| 16:15.06 | CunningPike | This should be fun :D |
| 16:15.06 | Qwell | extra watery! |
| 16:15.18 | Katty | what's americano? |
| 16:15.22 | tzanger | Katty: jeez |
| 16:15.27 | Qwell | Katty: coffee + water |
| 16:15.36 | Katty | that doesn't sound good. |
| 16:15.41 | elliot98 | or what's the chances that an playback application can be developed that has the feature to jump into the middle of a sound file? |
| 16:15.48 | Katty | tzanger: yes, terrible for my health |
| 16:15.51 | CunningPike | Katty: n x shots of espresso made up with water |
| 16:15.53 | Qwell | it's not regular coffee. it's like espresso or something |
| 16:15.56 | p3nguin | Heh, the guy at Microsoft didn't like that I called him without wanting support on a Microsoft product. |
| 16:16.04 | CunningPike | Au contraire - it's very good |
| 16:16.05 | Katty | CunningPike: ooooh, i see. still doesn't sound good tho :< |
| 16:16.16 | Katty | CunningPike: with all that pesky coffee stuff. |
| 16:16.21 | Chainsaw | p3nguin: See, it works if you format your number right :) |
| 16:16.49 | drmessano | p3nguin: Did he tell you to reboot anyway? |
| 16:16.50 | CunningPike | This is the most I have said in this channel in months! |
| 16:17.20 | Katty | CunningPike: well that simply will not do. |
| 16:17.21 | p3nguin | I can't be expected to know how to format every phone number... if they don't want me to dial a 0, they shouldn't put a 0 in the phone number. |
| 16:17.50 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:17.58 | p3nguin | drmessano: No, but he did tell me to go to their web site if I did need help with a product. |
| 16:18.35 | elliot98 | so is there any kind of application such as this? |
| 16:18.53 | drmessano | p3nguin: Did you tell him their website isn't very Lynx friendly and doesn't render well on your EUNICHS system? |
| 16:19.03 | Katty | raden_work: :>>>>> |
| 16:19.05 | Katty | glomps raden_work |
| 16:19.18 | Chainsaw | p3nguin: That's what the +44 (0) means. 0 inside the UK, no 0 outside the UK. |
| 16:19.41 | raden_work | glomps Katty :) |
| 16:19.45 | raden_work | how it going |
| 16:19.45 | p3nguin | If I switch over to the london server, I obviously don't need the 00/011, and I assume I don't need the 44... so I'd dial 0845 072 7227. |
| 16:19.51 | Chainsaw | p3nguin: Correct. |
| 16:19.51 | drmessano | Why do they prefix it with a + if you are supposed to REMOVE it? |
| 16:19.57 | Katty | raden_work: 's all good. |
| 16:20.06 | Katty | raden_work: what's blazin |
| 16:20.07 | Chainsaw | drmessano: The + is the international dialling code, so you'd replace that with 00 or 01. |
| 16:20.16 | Chainsaw | drmessano: The (0) is the one that goes away. |
| 16:20.18 | p3nguin | drmessano: That's my complaint. Provide the friggin' number as I should dial it to reach you! |
| 16:20.51 | drmessano | Chainsaw: I am well aware how it works.. Perhaps I should have added a "LOL, J/K :) ROFL" at the end |
| 16:21.14 | drmessano | But then again, I am not 14 |
| 16:21.16 | Katty | roflmao |
| 16:21.19 | raden_work | Katty, same ole working on Customer service software, and then need to get onto developing a light therapy box for a company, and then hopefully back to my asterisk billing platform |
| 16:21.40 | Katty | drmessano: hey what are you tryin to say ;) |
| 16:21.54 | Katty | raden_work: Light Therapy? |
| 16:21.56 | drmessano | Katty: That in 4 more years, I can hit on you? |
| 16:22.10 | Katty | drmessano: eww. |
| 16:22.37 | drmessano | I hope I am not taking that out of context, because my fragile ego is shattered completely |
| 16:22.46 | drmessano | :( <-- SAD FACE |
| 16:22.46 | Katty | i'm sure it is ;) |
| 16:23.41 | *** join/#asterisk smash- (smash@51.sub-75-196-69.myvzw.com) |
| 16:23.48 | raden_work | Katty, yea to speed up skin cell regeneration, specifically the dermal elastic fibers... |
| 16:24.00 | smash- | w0 |
| 16:24.09 | drmessano | wzero? |
| 16:24.13 | drmessano | wzed? |
| 16:24.24 | Katty | raden_work: hmm. |
| 16:24.36 | Katty | raden_work: what are you building it for, specifically? |
| 16:24.41 | drmessano | smash-, wuzzzzeeeed to you too! |
| 16:24.42 | Katty | raden_work: hatchery? |
| 16:24.47 | Katty | raden_work: children? |
| 16:24.53 | raden_work | have a few clinics funding it |
| 16:24.54 | Kobaz | your epidermis is showing |
| 16:24.55 | smash- | morning |
| 16:25.01 | smash- | lol |
| 16:25.04 | Katty | raden_work: ahh, i see. |
| 16:25.05 | drmessano | Kobaz, Your socks are untied |
| 16:25.12 | Kobaz | oh no |
| 16:25.17 | smash- | dropped ur pocket |
| 16:25.19 | Katty | raden_work: might help with bruising, scrapes, etc |
| 16:25.22 | raden_work | The main benefit is people with scaring , skin graphs, stretch marks, etc.... |
| 16:25.32 | drmessano | "dropped your pocket" is one of my favs |
| 16:25.39 | Katty | raden_work: that's a lovely idea. kudos to you dear |
| 16:25.57 | raden_work | we also have a lamp at 630 nm that eliminates acne in about 8 sessions |
| 16:26.10 | smash- | what if u dont have any |
| 16:26.24 | Katty | then yo udon't need it. |
| 16:26.37 | smash- | what if u want some |
| 16:26.39 | smash- | got anything for that |
| 16:26.42 | p3nguin | They certainly didn't like when I tried to dial the number "locally." Call rejected by 78.129.153.20: No such context/extension |
| 16:26.50 | raden_work | Have a lamp specifically made to stimulate a the bodys vitamin d production in about 3 - 4 minutes |
| 16:26.50 | smash- | like a peanut butter rub? |
| 16:26.53 | drmessano | raden_work: I have a laser that can cure all ailments in 3 seconds |
| 16:27.07 | raden_work | drmessano, sweet :) ill buy the rights to it |
| 16:27.21 | Katty | smash-: peanut butter's a bad idea. |
| 16:27.32 | Katty | smash-: if you're going to do a rub, i'd recommend a sugar rub |
| 16:27.41 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 16:27.42 | raden_work | smash-, we don't want to know what you do with peanut butter |
| 16:27.53 | smash- | =( |
| 16:28.09 | Katty | honey is also good for the skin. |
| 16:28.20 | Katty | rice water. |
| 16:28.23 | Katty | tomato. |
| 16:28.30 | raden_work | smash-, get a massage if you want a rub lol |
| 16:28.33 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 16:28.46 | raden_work | Katty, interesting ... |
| 16:29.07 | raden_work | there also a extract from red wine thats good for the skin cant remember what it is |
| 16:29.18 | Katty | if you remember, lemme know. |
| 16:29.33 | p3nguin | I don't see why I would have to dial an international calling code to call a UK phone number from a London phone system. |
| 16:29.37 | raden_work | its in my research notes somewhere |
| 16:29.53 | Katty | oh. |
| 16:29.55 | raden_work | p3nguin, that seems slightly odd |
| 16:29.58 | Katty | is it hard to replace a head light? |
| 16:30.07 | Chainsaw | p3nguin: Do you not have a prefix on the ITSP for outbound calls? Like a 9? |
| 16:30.08 | raden_work | Katty, what kinda car ? |
| 16:30.12 | p3nguin | katty: Depends on several factors. |
| 16:30.18 | Katty | raden_work: pontiac sunfire |
| 16:30.20 | p3nguin | katty: Sometimes it's not hard at all. |
| 16:30.25 | raden_work | pretty simple |
| 16:30.43 | Katty | hmm. i might give it a shot on my own then. |
| 16:30.45 | drmessano | p3nguin: The same reason they look at you funny if you ask for a "Flashlight" |
| 16:30.46 | raden_work | Katty, there are 2 i believe 10 mm bolts on top of the headlight |
| 16:30.49 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:31.00 | raden_work | Katty, take them off entire light comes off car change bulb and replace light |
| 16:31.07 | raden_work | Katty, take about 5 min :) |
| 16:31.14 | Katty | excellent. |
| 16:31.24 | Katty | i'm assuming the people at autozone or whatever can tell me which light i need |
| 16:31.36 | Katty | oh. could i put blue lights in? |
| 16:31.44 | drmessano | Sometimes it's as easy as giving a quarter turn to the bulb socket |
| 16:31.51 | drmessano | If it's exposed |
| 16:31.54 | p3nguin | katty: What year is the car? |
| 16:31.58 | Katty | p3nguin: 2000 |
| 16:32.42 | Katty | my dad would know. |
| 16:32.54 | Katty | but then i wouldn't learn how to do it |
| 16:32.57 | drmessano | My fav auto lighting accessory was the 50w bulbs I got for the backup lights |
| 16:33.12 | drmessano | Complimented my 55w headlights nicely |
| 16:33.13 | Katty | drmessano: do you know if the blue headlight ones are very expensive? |
| 16:33.23 | Katty | i think they're pretty. |
| 16:33.27 | p3nguin | katty: Looks like you need 9007 bulbs. |
| 16:33.29 | drmessano | Katty: Premium price.. But not much anymore |
| 16:33.36 | raden_work | Katty, what year your sunfire ? |
| 16:33.41 | Katty | p3nguin: awesome, thanks. |
| 16:33.43 | Katty | raden_work: 2000 |
| 16:33.50 | drmessano | Like $8 for the standard bulb, maybe $14 for the blue or something |
| 16:33.56 | Katty | oh that's not much at all |
| 16:33.56 | drmessano | if they're even THAT much more anymore |
| 16:34.07 | raden_work | Katty, headlight 9007 |
| 16:34.40 | Katty | i'm guessing i need more than a screwdriver to change the lights tho |
| 16:34.46 | Katty | which might be an issue. i'll go look at my car. |
| 16:35.10 | raden_work | Katty, its either a 8 mm or 10 mm socket or wrench i cant rember been awhile since I have worked on a sunfire |
| 16:35.54 | p3nguin | I have Silver Star bulbs which I like a lot, and they are $40 for a 2-pack for your car. |
| 16:36.05 | raden_work | Katty, if you have a cresent wrench you good :) |
| 16:36.41 | Qwell | Katty: make sure they're legal in your area |
| 16:36.42 | p3nguin | That's at AZ... Walmart might be a little less. |
| 16:38.01 | raden_work | Qwell, all sylvania lights are DOT legal |
| 16:38.09 | Qwell | DOT != city :) |
| 16:38.12 | raden_work | p3nguin, silverstars rock |
| 16:38.35 | Qwell | my sister-in-law got a ticket for using blue-tint headlights in a city where they weren't legal (by driving THROUGH the city, on the freeway) |
| 16:38.36 | p3nguin | they're nice and bright. |
| 16:38.40 | raden_work | Qwell, city must obide by DOT regulations |
| 16:38.46 | Katty | well i went and looked at my car, but i think i'm going to have to pop the hood to actually see a way to ...uhh take that clear plastic thingy off from the front of the light |
| 16:38.56 | raden_work | p3nguin, on my diesel ram the vibration is to much and I keep having them burn out :( |
| 16:39.04 | p3nguin | that's not uncommon. |
| 16:39.33 | p3nguin | katty: You usually change the bulb from the backside of the headlight assembly. |
| 16:39.39 | Katty | oooh. |
| 16:39.46 | Katty | okay. that makes more sense i suppose. |
| 16:39.58 | p3nguin | There is usually a retaining ring that you twist and the bulb pulls out of the housing. |
| 16:40.01 | Qwell | Katty: there's usually 1 screw/bolt holding it in. |
| 16:40.08 | Qwell | well, plus the bulb itself |
| 16:40.12 | Kobaz | Qwell: it's just a clip on my car |
| 16:40.15 | p3nguin | zero screws/bolts |
| 16:40.15 | raden_work | Qwell, 2 on a sunfire entire light assembly comes out |
| 16:40.15 | Qwell | protip: DO NOT TOUCH THE BULB |
| 16:40.25 | Qwell | use *clean* gloves, and don't touch the glass. |
| 16:40.34 | Kobaz | Katty: yes... wear latex gloves or something, if you think you'll wind up touching the bulb |
| 16:40.40 | Katty | oh |
| 16:40.43 | Katty | that's good to know |
| 16:40.43 | Qwell | any oil on your hand will cause it to break easily |
| 16:40.44 | p3nguin | Yeah, never get body oil on a halogen bulb. |
| 16:40.48 | p3nguin | It ruins them. |
| 16:41.01 | Katty | found a youtube video |
| 16:41.02 | p3nguin | They'll go yellow in a few minutes if you get oil on them. |
| 16:41.07 | Kobaz | the heat reflects off the oils and the bulb will burn out |
| 16:41.10 | Katty | 13mm wrench and screwdriver |
| 16:41.10 | Qwell | p3nguin: I thought they just shattered? |
| 16:41.16 | Kobaz | Qwell: nope |
| 16:41.39 | Qwell | well, either way, it's bad :D |
| 16:41.44 | Kobaz | mmmkay |
| 16:41.49 | tzanger | Kobaz: well light is absorbed by the oil, creating heat and hot spots. depending on if they're under pressure or not they can fail spectacularly |
| 16:42.06 | *** join/#asterisk timahvo1 (~rogue@41.223.57.77) |
| 16:42.08 | Kobaz | jeebus... 50mb for a windows broadcom ethernet driver |
| 16:42.20 | Kobaz | what happened to 2kb drivers that fit on floppys |
| 16:42.24 | *** join/#asterisk rue_work (~rue_work@24.207.122.10) |
| 16:42.35 | tzanger | Kobaz: clearly you needed all the support files, icnluding a flash animation and web browser plugin |
| 16:42.39 | Kobaz | haha |
| 16:43.39 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:43.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 16:43.42 | p3nguin | kobaz: I usually download their stupid multi-megabyte packages, then extract the driver by using 7zip. |
| 16:43.42 | Naikrovek | weird... when i scratch the sole of my foot i get heartburn. every. single. time. |
| 16:43.52 | Naikrovek | then when i stop, it goes away |
| 16:43.54 | Naikrovek | ... |
| 16:43.58 | Qwell | Don't do that then? |
| 16:44.03 | Naikrovek | but qwell |
| 16:44.03 | rue_work | so I have a aastralink pro 160 with audio breaking up (not echo) really bad, aastra wants me to grab a network capture, have wireshark, do I HAVE to find a 100mbit hub to sample the network or is there a setting I cant find to make the passthru port on a set behave like a hub not a switch |
| 16:44.05 | Naikrovek | it ITCHES |
| 16:44.06 | p3nguin | naikrovek: Are you overweight? |
| 16:44.13 | Naikrovek | p3nguin: yes |
| 16:44.22 | Naikrovek | some would call me a fat fucker, sure |
| 16:44.29 | p3nguin | You may be experiencing the same problem I have... GERD. |
| 16:44.37 | Naikrovek | googles |
| 16:44.59 | p3nguin | My doctor gave me Protonix and I haven't had heartburn since. |
| 16:44.59 | Qwell | it's autoimmune! |
| 16:45.01 | *** join/#asterisk KavanS (~KavanS@unaffiliated/kavans) |
| 16:45.03 | Qwell | has watched too much House. |
| 16:45.16 | Naikrovek | no this isn't real heartburn, but heartburn is the best description for it |
| 16:45.24 | p3nguin | erm... oh. |
| 16:45.33 | Naikrovek | when i (or someone else) scratches the sole of my foot, i get a tingly sensation in my chest |
| 16:45.36 | Naikrovek | also |
| 16:45.44 | p3nguin | well that's just weird, then. |
| 16:45.46 | Naikrovek | when i scratch my elbow i feel it on my stomach and vice versa |
| 16:45.48 | Naikrovek | yeah it is weird |
| 16:45.55 | drmessano | Maybe you're just making your esophagus horny by scratching your foot? |
| 16:46.01 | Naikrovek | heh |
| 16:46.01 | p3nguin | I was trying to help with actual heartburn. |
| 16:46.11 | Naikrovek | actual heartburn is not a problem, but thank you kindly, sir |
| 16:46.34 | p3nguin | Protonix is some nice stuff if you ever experience chronic heartburn. |
| 16:46.45 | Naikrovek | before i had chemotherapy i had an ulcer |
| 16:46.56 | p3nguin | It's stronger than Prilosec. |
| 16:46.56 | drmessano | Yeah, don't let them shove any cameras in either end |
| 16:47.04 | Naikrovek | they gave me a 7 day prescription of prevacid and i haven't had heartburn of any significant strength since |
| 16:47.09 | Naikrovek | drmessano: noted. |
| 16:47.18 | Naikrovek | that was 15 years ago, btw |
| 16:47.23 | p3nguin | wow |
| 16:47.30 | p3nguin | YOu were healed that quickly? |
| 16:47.32 | Naikrovek | yeah that worked wonders for me |
| 16:47.46 | Naikrovek | yeah ulcer w/chronic 24/7 heartburn to zero heartburn at all in about 36 hours |
| 16:47.56 | p3nguin | I figure I'll take mine for a while before I heal. |
| 16:48.03 | Naikrovek | then chemo started, and all was well |
| 16:48.07 | Naikrovek | except the chemo part |
| 16:48.09 | Naikrovek | ugh |
| 16:48.10 | Kobaz | Protonix doesn't do anything for me |
| 16:48.12 | rue_work | ARG the only menion I can find is ME, HERE |
| 16:48.18 | Kobaz | only prilosec works |
| 16:48.27 | p3nguin | kobaz: Perhaps you were taking it incorrectly? Most people do. |
| 16:48.27 | drmessano | p3nguin: That's generally why it's a 7 or 14 day "treatment".. it's SUPPOSED to resolve the problem.. though many end up taking it long term |
| 16:48.33 | rue_work | wireshark, I'm gonna need a hub aren't I? |
| 16:48.38 | Kobaz | p3nguin: correctly as in? |
| 16:48.44 | rue_work | ?bla |
| 16:48.47 | rue_work | ??bla |
| 16:48.58 | rue_work | !lart rue_work |
| 16:49.06 | rue_work | dunno if this irc client works... |
| 16:49.11 | p3nguin | kobaz: You're supposed to take it FIRST THING when you wake up before the stomach cells get active. |
| 16:49.13 | drmessano | Kobaz: It's a suppository.. you stuck it in the wrong end |
| 16:49.18 | p3nguin | haha |
| 16:49.49 | Kobaz | p3nguin: yeah, i would take it before breakfast |
| 16:49.53 | Kobaz | and get acid my afternoon |
| 16:49.57 | Kobaz | by |
| 16:50.05 | p3nguin | And you're supposed to wait at least 30 minutes before eating. |
| 16:50.42 | p3nguin | Most people don't take PPIs correctly, so they think the meds don't work for them. |
| 16:50.47 | rue_work | hello? |
| 16:50.52 | Kobaz | never had any issues with prilosec and the knockoffs |
| 16:50.58 | Naikrovek | rue_work: it works fine |
| 16:51.11 | rue_work | ah thanks / sorry |
| 16:51.15 | Naikrovek | i'm trying to figure out what you're trying to do |
| 16:51.33 | rue_work | I have a office with a aastralink pro 160 'pbx' |
| 16:51.45 | rue_work | its a lan sysetm only, no wan traffic |
| 16:52.02 | rue_work | the audio on some calls starts stuttering |
| 16:52.20 | rue_work | I need to know how to diagnose it |
| 16:52.33 | rue_work | the aastralink pro 160 runs asterisk |
| 16:53.10 | rue_work | and they have nothing but the phonesystem and a laserprinter on the network |
| 16:53.32 | *** join/#asterisk Abhishek_Singh (~root@220.225.244.114) |
| 16:53.46 | Naikrovek | rue_work: at this point all we can do is guess |
| 16:53.48 | rue_work | its an ideal system I have no idea why there are problems |
| 16:53.49 | drmessano | Swap out the 10mb hubs for 100mb switches |
| 16:53.51 | Naikrovek | network trouble? |
| 16:53.55 | rue_work | guesses are good |
| 16:53.55 | Naikrovek | yes what drmessano said |
| 16:53.56 | Naikrovek | heh |
| 16:54.20 | Naikrovek | make sure they're switches and not hubs |
| 16:54.32 | Naikrovek | check for congestion though i don't know how there could be any |
| 16:54.34 | drmessano | If they're gray, throw them away |
| 16:54.48 | drmessano | If they're blue, you paid more too |
| 16:54.49 | rue_work | its a switch I supplied it |
| 16:54.57 | Naikrovek | make/model? |
| 16:55.09 | rue_work | milan 800 |
| 16:55.09 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:55.26 | drmessano | Uh |
| 16:55.29 | Naikrovek | yeah uh |
| 16:55.33 | Naikrovek | wtf is a milan |
| 16:55.44 | rue_work | well its not expensive |
| 16:55.57 | Naikrovek | google shows nothing for "milan 800" |
| 16:55.58 | *** join/#asterisk riscphree (~riscphree@h216.115.90.75.dynamic.ip.windstream.net) |
| 16:56.02 | rue_work | really... |
| 16:56.03 | rue_work | hmm |
| 16:56.26 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 16:56.30 | rue_work | 'mil-s800' |
| 16:56.36 | drmessano | milan 800 switch got me there |
| 16:56.47 | drmessano | milan 800 did not, as you found |
| 16:57.00 | Naikrovek | k |
| 16:57.08 | drmessano | They make Trendnet look like ProCurves |
| 16:57.15 | Naikrovek | my thought exactly |
| 16:57.17 | p3nguin | Milan? Get rid of that Dollar General network hardware. |
| 16:57.27 | Naikrovek | spend $100 get a procurve or somethink |
| 16:57.29 | rue_work | we use a 24 port ver in out office WITH medium network traffic and never have voip problems |
| 16:57.30 | Naikrovek | even netgear |
| 16:57.44 | Naikrovek | netgear is fine actually, procurve if you wanna get fancy |
| 16:57.45 | drmessano | They probably have a 10mbit backplane |
| 16:58.07 | rue_work | remember this is a switch with only 5 3491 phones on it |
| 16:58.08 | drmessano | or maybe you just have a bad switch |
| 16:58.17 | rue_work | we already changed it, no difference |
| 16:58.27 | drmessano | Same make/model? |
| 16:58.28 | Naikrovek | yeah step 1 is to swap out that switch, even if only temporarily |
| 16:58.35 | rue_work | no 24port vs 8 port |
| 16:58.41 | Naikrovek | okay |
| 16:59.02 | Naikrovek | well i don't that switch but that is less likely to be the problem now, unless it's the same make or something |
| 16:59.08 | drmessano | Check your connections.. bad punch, bad plug causing retries out the ass |
| 16:59.18 | rue_work | aastra tech asked me for a network capture, wich is really useless seeing iirc they dont support hubbed connections |
| 16:59.19 | Naikrovek | go to best buy, get a netgear, try that, if it still does it, return the switch to best buy |
| 16:59.41 | Naikrovek | rue_work: REAL switches can barf everything they see out a port that you put wireshark on |
| 16:59.55 | rue_work | the pbx machine is right beside the switch, I can change out its cable... |
| 16:59.57 | Naikrovek | i forget the term though |
| 17:00.06 | Naikrovek | steps out for a few minutes |
| 17:00.07 | drmessano | Naikrovek: You're right.. Best Buy would have to sell them. Wal-mart has better junk |
| 17:00.08 | rue_work | yea managed switches cost a lot more than $50 |
| 17:00.13 | Naikrovek | lol yeah |
| 17:00.41 | rue_work | its an intermittent problem too |
| 17:00.57 | rue_work | ok well, that 1 cable is really the only solid idea here |
| 17:01.15 | rue_work | but for the record, to get a network capture I'd need to find a 100mbit hub, yes? |
| 17:01.21 | drmessano | Does the switch light up like a christmas tree when the phones are barfing? |
| 17:01.24 | rue_work | or a managed switch |
| 17:01.32 | rue_work | I can find that out |
| 17:01.47 | rue_work | your thinking that its beign killed by broadcast traffic? |
| 17:01.54 | drmessano | Perhaps |
| 17:01.58 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:01.59 | p3nguin | Netgear has some managed switches. |
| 17:02.17 | raden_work | runs |
| 17:02.17 | rue_work | aka cheap backplane (which I'd expect) BUT agian, larger office, heavier traffic, and no problems |
| 17:02.25 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 17:03.58 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 17:04.18 | drmessano | rue_work: I've seen a Linksys 4 port router/wireless box used as an AP shut down a network with 30 workstations on 2 $600 Cisco switches before.. Network issue is a network issue.. |
| 17:04.56 | drmessano | You need to isolate. Look for blinky lights on the switch. If something is MORE blinky when the jitter starts, unplug it |
| 17:05.10 | rue_work | fair enough |
| 17:05.16 | drmessano | Could be as simple as a bad endpoint |
| 17:06.02 | rue_work | the router that feeds the switch is a wireless thing, maybe I'll see about putting the voip gear on a seperate subnet with fixed ip's |
| 17:06.15 | drmessano | Packet capture and pie charts are great, but sometimes its better to look for the obvious and have another bite of your sandwich |
| 17:06.18 | rue_work | then again, it kinda sounds like that wouldn't help |
| 17:06.56 | rue_work | packet capture is only gonna be usefull if I'm packing a 100m hub... |
| 17:07.04 | rue_work | else I only know broadcast... |
| 17:07.19 | rue_work | ok I have a few leads I'll go see what I can do with them |
| 17:07.31 | fauxalliance | hold the mayo. |
| 17:07.51 | rue_work | my batsenses are telling me this is somehow ultimitly related to the wireless I lothe so much |
| 17:08.08 | fauxalliance | 802.11 is not so misbehaved... |
| 17:08.39 | fauxalliance | just an invisible torment. |
| 17:09.28 | *** join/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 17:10.25 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 17:11.24 | rue_work | ok thanks, farewell |
| 17:11.46 | nny | ugh. I was asking about this yesterday. I have an ITSP stating that we are using inband for DTMF instead of RFC2833. I have every peer set to use RFC2833, sip show peer XXX (including the ITSP) show dmtfmode=rfc2833 and the polycom sip.cfg states rfc2833. Their IP Engineer states that we are sending inband regardless. The DTMF even shows up in packetdumps with wireshark.. |
| 17:12.12 | nny | I am 99% sure the ITSP is crazy, anything else I should check? |
| 17:13.33 | [TK]D-Fender | nny: Pass them SIP debug and tell them to fuck off :) |
| 17:13.53 | nny | [TK]D-Fender: mm good point |
| 17:14.08 | nny | [TK]D-Fender: that's not even the strongest word I want to use, but it suffices |
| 17:14.42 | nny | i have been dealign with this for a month and a half. Each time I do insance amounts of troubleshooting, each time they come back with the utmost stupidest reponses |
| 17:15.01 | b11d` | which means they have no clue and want to buy time |
| 17:15.13 | b11d` | if they continue to reject you.. check out the Dunning-Kruger Effect :p |
| 17:15.49 | nny | the auto attendant we were testing against would reset if you hit 0. They tried it and said "0 does nothing" Can we test again with X. I told them 0 should reset the AA, they said oh, it does, but we want to test it against an extension that works on the remote system |
| 17:16.00 | nny | ./facepalm |
| 17:16.41 | nny | b11d`: agreed |
| 17:16.51 | *** join/#asterisk Faithful (~Faithful@nat76.mia.three.co.uk) |
| 17:18.40 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 17:18.44 | drmessano | nny: Who is the ITSP? |
| 17:22.10 | drmessano | hmmm |
| 17:22.38 | nny | drmessano: called "Excel" |
| 17:22.58 | drmessano | Never heard of them |
| 17:28.14 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 17:31.14 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:32.01 | *** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net) |
| 17:41.25 | Katty | drags in |
| 17:41.37 | [TK]D-Fender | What did the Kat drag in? |
| 17:41.47 | Katty | my ass. |
| 17:41.49 | Katty | cause it's draggin |
| 17:42.33 | *** join/#asterisk Alatariel (~Alatariel@g230056098.adsl.alicedsl.de) |
| 17:42.38 | Alatariel | guten Abend |
| 17:42.56 | Katty | guten tag! |
| 17:43.07 | [TK]D-Fender | Gluten-free! |
| 17:43.09 | b11d` | wass was los |
| 17:43.12 | b11d` | was est los |
| 17:43.14 | Alatariel | Schick. Erste Mal hier und direkt eine Antwort :D |
| 17:43.14 | b11d` | whatever it is |
| 17:43.32 | Katty | wie geht es ihnen? |
| 17:43.55 | WIMPy | Alatariel: Ja, aber du solltest die Sprache oder den Channel wechseln (#asterisk.de) |
| 17:43.59 | *** join/#asterisk mace (~mace@debian/developer/mace) |
| 17:44.03 | Alatariel | sorry :D |
| 17:44.09 | Alatariel | thx WIMPy |
| 17:44.16 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:44.36 | Katty | Alatariel: gluck (= |
| 17:44.37 | Alatariel | I got a problem. Mainly a combination from "no idea" and "no time". |
| 17:44.49 | Katty | Alatariel: consultant? |
| 17:45.01 | Alatariel | Katty: technician |
| 17:45.09 | WIMPy | No time+no idea=lots of money |
| 17:45.18 | Katty | Alatariel: do you need a consultant? |
| 17:45.31 | Katty | ummumm |
| 17:45.35 | Katty | brauchen sie einen berater |
| 17:45.36 | Katty | i think |
| 17:45.48 | Alatariel | Katty: The concept is ready, but the realisation doesn't work |
| 17:45.56 | Katty | )= |
| 17:46.04 | Katty | i understand. |
| 17:46.24 | WIMPy | Good enough. Most people I meet don't know what thy want. |
| 17:46.34 | Katty | i want a cookie. |
| 17:46.40 | Katty | and a male. |
| 17:46.41 | *** part/#asterisk mace (~mace@debian/developer/mace) |
| 17:46.45 | Alatariel | Ticket System OTRS sends a mail to itself, which gets translated into a wav file. Now I have to create a call using asterisk and calling a swyx telephony platform |
| 17:46.46 | Katty | male optional. |
| 17:46.58 | Alatariel | only tofifee here ;) |
| 17:47.02 | Qwell | Katty: what about a male cookie? |
| 17:47.11 | Qwell | (can cookies have gender?) |
| 17:47.14 | Katty | Qwell: i guess cookies would have genders |
| 17:47.17 | Katty | Qwell: just not in english |
| 17:48.37 | Qwell | Yes, in German, das Plätzchen. |
| 17:48.39 | Alatariel | I created a call file which should playback the wav file and call the swyx which will do the call routing and has the isdn card. But in asterisk I always get "remote side is busy" with SIP code 600 as soon as I move the call file to the outgoing queue |
| 17:48.48 | Qwell | it's neuter though. |
| 17:48.56 | Katty | go figure, huh |
| 17:49.04 | Katty | languages are all messed up |
| 17:49.09 | WIMPy | "Der Keks" might be better |
| 17:49.13 | Katty | the sun is masculine in spanish el sol, but feminine in germ die sonne |
| 17:49.27 | Katty | and then the moon is opposite |
| 17:49.39 | Qwell | weird |
| 17:49.41 | Alatariel | I would need someone who could have a look into the call file and tell me if there is anything wrong with it, so I can get the swyx guy to fix his side |
| 17:49.43 | Katty | very. |
| 17:49.53 | KavanS | differences in polycom 321 vs. 330 |
| 17:50.01 | Katty | Alatariel: i would recommend a consultant |
| 17:50.03 | KavanS | any sizable ones? |
| 17:50.09 | Katty | Alatariel: someone who speaks german. |
| 17:50.23 | Katty | KavanS: i think the 330 has an extra network jack |
| 17:50.24 | [TK]D-Fender | KavanS: 330 is older revision and has ethernet passthrough |
| 17:50.30 | Alatariel | Katty: :-p |
| 17:50.43 | KavanS | ok, I don't need ethernet passthrough I do not think |
| 17:50.57 | Katty | Alatariel: you might try #asterisk.de |
| 17:51.09 | Alatariel | Katty: is my english this bad? :-( |
| 17:51.23 | Katty | Alatariel: no, it's not bad. |
| 17:51.31 | Alatariel | but? |
| 17:52.00 | Katty | Alatariel: it would be easier on you with a german consultant |
| 17:52.27 | Alatariel | Katty: #asterisk.de has 5 users in it (me included) ... |
| 17:52.40 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 17:52.51 | Katty | what's the asterisk consultant channel? |
| 17:53.05 | KavanS | sweet ok |
| 17:53.15 | Alatariel | why exactly would i need a consultant? |
| 17:53.27 | Katty | Alatariel: to get help. |
| 17:53.36 | Alatariel | Katty: Is this no help channel? |
| 17:53.44 | Katty | Alatariel: sure it is, but it will take time. |
| 17:53.51 | Katty | Alatariel: do you have time? |
| 17:54.06 | Alatariel | Katty: for you, as much as you need ;) |
| 17:54.13 | Katty | that's what he said. |
| 17:54.26 | WIMPy | LOL |
| 17:55.09 | Alatariel | and my last toffifee |
| 17:55.18 | Katty | how cute. |
| 17:55.27 | Alatariel | desparate matches |
| 17:55.44 | Alatariel | -a +e |
| 17:57.00 | bmoraca_work | stupid server 2003 won't install |
| 17:57.01 | bmoraca_work | arg |
| 17:57.04 | KavanS | Alatariel, when people are mentioning you should get a consultant - it's a nice way of saying "you clearly are in over your head" |
| 17:57.25 | Katty | bmoraca_work: have you tried a hammer? |
| 17:57.30 | Alatariel | KavanS: never heard this one |
| 17:57.50 | WIMPy | bmoraca_work: Thank your hardware. |
| 17:57.53 | bmoraca_work | Katty: getting close...though it's a virtual server, so not much to smash on |
| 17:58.02 | bmoraca_work | WIMPy: it's a virtual server |
| 17:58.09 | ccomp5950 | Smash the hard drive platters, same difference. |
| 17:58.18 | bmoraca_work | i think my ISO or the server it's on is having troubles |
| 17:58.26 | WIMPy | Ok, then thank the virtualtisation thing. |
| 17:58.51 | Katty | bmoraca_work: bummer )= |
| 17:59.06 | Katty | bmoraca_work: i'd guess it's the ISO |
| 17:59.16 | Katty | bmoraca_work: unless you've seen a number of other funky squirks |
| 17:59.49 | bmoraca_work | it's weird because i just loaded another server on the same ESXi host with the same ISO yesterday |
| 17:59.59 | carrar | hammer time! |
| 18:00.00 | bmoraca_work | or, rather, monday |
| 18:00.07 | Katty | carrar: <3 |
| 18:00.12 | nny | seriously hate DTMF |
| 18:00.21 | carrar | DTMF is so last year |
| 18:00.25 | bmoraca_work | certainly getting close to hammer time...stupid piece of shit |
| 18:00.34 | Katty | bmoraca_work: piece of snickerdoodles. |
| 18:00.50 | nny | unrelated to my first issue, have a phone at my home office, server hosted on a public IP elsewhere. I cannot get dtmf to work sanely,no matter the method |
| 18:00.54 | bmoraca_work | mmmm....snickerdoodles are good. might have to make 4-5 batches of them this year |
| 18:01.09 | Katty | nny: have you tried a softphone? |
| 18:01.14 | b11d` | snickerdoodles are good.. snickerdoodle flavored coffee... not so good |
| 18:01.22 | Katty | nny: or generate them directly from the dial plan? |
| 18:01.38 | Katty | nny: dialdigits or whatever |
| 18:01.50 | carrar | using rfc2833? |
| 18:02.00 | nny | Katty: have not tried a softphone, have tried multiple ip phones |
| 18:02.08 | nny | yeah rfc2833. Just doesn't always get each press |
| 18:02.11 | Katty | nny: i ran into a dtmf problem which seemed to fail 75% of the time on a polycom phone, 50% of the time on a softphone, but never from the dialplan |
| 18:02.38 | Katty | nny: turned out to be my motherboard |
| 18:02.44 | Katty | nny: but still worth a shot. |
| 18:03.10 | carrar | did you toss that 386? |
| 18:03.20 | nny | well got it to duplicate them now |
| 18:03.27 | nny | so moving.. somewhere |
| 18:03.31 | Alatariel | Katty: could you have a look into my call file and tell me if there are some eclatant errors? |
| 18:03.33 | Katty | nny: it will help if you can get your telco or itsp to setup a number for you to dial into and spit back whatever dtmf you're sending out |
| 18:04.18 | nny | omgwtf |
| 18:04.20 | Katty | nny: you oculd also setup a similiar test on your server to just spit back wahtever it's recieving from the phones. that could isolate it to eternal/external at least |
| 18:04.31 | Katty | erm, internal. |
| 18:04.37 | Katty | wonders where eternal came from |
| 18:04.46 | nny | went from not getting evey digit to duplicating them |
| 18:04.51 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:04.53 | carrar | The Bible? |
| 18:04.54 | Katty | yeah we had that problem as well. |
| 18:04.56 | nny | rfc2833 is packet based |
| 18:05.05 | Katty | nny: sometimes it would just skip the number, and then spit a whole bunch back later |
| 18:05.08 | Katty | nny: like a major delay |
| 18:05.15 | nny | so i send a packet to my asterisk box and it goes to my ITSP |
| 18:05.21 | nny | how does that get duplicated on my end? |
| 18:05.34 | Katty | nny: but again, it may not apply to your situation. you can also check your logs. i believe they show dtmf |
| 18:05.41 | Katty | nny: if you enable logging at the cli |
| 18:05.46 | nny | yeah gonna do a debug on it next |
| 18:05.50 | Katty | nny: sip debug i think shows it |
| 18:05.56 | Katty | nny: don't quote me on that, it's been ahwile |
| 18:05.59 | bmoraca_work | nny: try rfc2833 from the phone to asterisk and inband from asterisk to the ITSP |
| 18:06.16 | Katty | nny: you can do something like read digits |
| 18:06.20 | Katty | nny: and then playback |
| 18:06.26 | Katty | nny: that will show you if it's just external or not |
| 18:06.42 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 18:07.21 | nny | yeah |
| 18:07.55 | Katty | the log will show the dtmf |
| 18:08.01 | Katty | and if it's botched up or not |
| 18:08.52 | nny | hmm now it works fine heh |
| 18:09.12 | Katty | talk sweetly and carry a bit stick. |
| 18:09.15 | Katty | s/bit/big/ |
| 18:09.23 | Katty | i really must check my internal wiring. |
| 18:12.07 | asilva | Hi there..is there a limit for callgroup and pickupgroup when using realtime(mysql) ?? |
| 18:17.11 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
| 18:18.04 | asilva | GOT THIS MESSAGE - [Sep 15 14:48:30] WARNING[11144]: channel.c:5713 ast_get_group: Ignoring invalid group 83 (maximum group is 63) |
| 18:18.10 | asilva | is there a way to increase that value? |
| 18:18.24 | Corydon76-dig | You need more than 64 groups? |
| 18:18.46 | asilva | yes |
| 18:19.01 | Corydon76-dig | There is no easy way to increase that value. It would be a very invasive change |
| 18:19.20 | Corydon76-dig | The previous increase, from 32 to 64, was fairly easy |
| 18:19.56 | asilva | ain't no possibility to increase that? |
| 18:21.15 | Corydon76-dig | Possibility, sure. Easy, no. |
| 18:21.47 | Corydon76-dig | The gcc compiler doesn't have a native 128-bit integer type as of yet |
| 18:22.52 | Corydon76-dig | So any such increase would involve changing the code from dealing with an integer to dealing with a pointer to a structure. |
| 18:24.04 | Corydon76-dig | Not easy, not simple. Have fun with that. |
| 18:24.52 | Corydon76-dig | Why exactly do you need more than 64 callgroups, anyway? |
| 18:25.31 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 18:25.57 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 18:26.30 | EmleyMoor | Is there anything I can do at the asterisk side to block abusers trying to SIP REGISTER? |
| 18:26.33 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 18:27.41 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 18:30.22 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 18:32.51 | [TK]D-Fender | EmleyMoor: Yes. Its called "firewall" |
| 18:33.17 | EmleyMoor | My firewall does not appear to be effective |
| 18:33.51 | [TK]D-Fender | EmleyMoor: iptables <--------- |
| 18:36.02 | afink | Hello all. I'm having an issue where my voice directory isn't working as expected. I have three users with the same last name that starts with a "J" but I get the first "J" then a user with an "L" last name then the rest of the "J" last names. One thing I noticed is that the user's extensions go in sequential order. Any ideas? |
| 18:37.33 | Katty | russellb: congrats on the 356 pages. |
| 18:37.43 | Katty | russellb: well done, sir. |
| 18:37.56 | russellb | <3 |
| 18:38.23 | russellb | it's just cat /var/log/asterisk/full >> book.txt |
| 18:38.28 | russellb | makes it get long fast |
| 18:38.46 | russellb | I have to give credit to leifmadsen for the idea, though |
| 18:39.15 | Katty | leifmadsen: kings to you also. |
| 18:43.27 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:43.52 | Katty | hi tony |
| 18:44.46 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:44.54 | *** join/#asterisk Alagar (~Administr@122.164.172.12) |
| 18:49.26 | afink | wow...I must be retarded. Thanks for letting me figure that one out on my own... |
| 18:50.46 | *** join/#asterisk ritztech (~ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 18:51.07 | ritztech | anyone know where a good Support is for Custom Dialplans/ AGI |
| 18:52.07 | bmoraca_work | ~book |
| 18:52.07 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:52.11 | bmoraca_work | that might be a good one |
| 18:53.18 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 18:53.32 | ritztech | i actually bought that sucker a 2 years ago... i mean like over phone cause digium is like 1000 dollars a year for only a few cases and thats NUTS |
| 18:53.41 | ritztech | trying other companies but holy Shiz |
| 18:53.54 | ritztech | lod.com might but i cant seem to make sure hes not wasting my time |
| 18:54.31 | bmoraca_work | i do consulting work |
| 18:55.22 | bmoraca_work | if you have a specific question, however, people might just look at it here for free |
| 18:57.22 | ritztech | i have but i believe im 80% done |
| 18:59.40 | Katty | bmoraca_work: i could use a consultant. |
| 18:59.52 | Katty | bmoraca_work: the head light in my car needs changing. |
| 18:59.57 | bmoraca_work | lol |
| 19:00.04 | bmoraca_work | i'm not a very car-smart person |
| 19:00.07 | bmoraca_work | unfortunately |
| 19:00.08 | Katty | :< |
| 19:00.31 | bmoraca_work | only ever changed a headlight in a 1956 Chevy Belair...was way simple |
| 19:00.53 | Qwell | Katty: I'd change them for you. and I'd even pretend to know what I was doing! |
| 19:00.55 | Katty | there's a light in my house i can't reach. |
| 19:01.09 | Katty | Qwell: horay! |
| 19:01.37 | beardy | I'm hungry. |
| 19:01.46 | beardy | What does Katty suggest? |
| 19:01.49 | Katty | gives beardy the rest of her lunch. |
| 19:01.55 | tzanger | I should get some lunch |
| 19:01.57 | beardy | \o/ |
| 19:02.01 | Katty | i recommend tacos. |
| 19:02.05 | tzanger | maybe arby's, I haven't eaten there in forever |
| 19:02.08 | tzanger | I had tacos last night |
| 19:02.15 | Katty | have more. |
| 19:02.16 | tzanger | u.s. taco bell isn't the same as canadian |
| 19:02.20 | tzanger | it's too far away from where I work |
| 19:02.24 | Katty | :< |
| 19:02.34 | tzanger | I had mexican food at an "authentic" mexican place here but it was *awful* |
| 19:02.37 | Katty | fruit salad! |
| 19:02.45 | beardy | It's raining and I'll have to go to the store. |
| 19:02.47 | Katty | mmmm, mexican food. |
| 19:02.48 | tzanger | if that's authentic mexican then it explains why I've never seen a fat mexican |
| 19:02.57 | Katty | MmMMmmmm black beans and corn in burrito |
| 19:03.11 | tzanger | see that sounds tasty |
| 19:03.29 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:03.33 | Katty | why couldn't i have that for lunch )= |
| 19:03.40 | tzanger | why don't you? |
| 19:04.05 | Katty | idk, i was thinking i needed iron so i made something with meat in it. |
| 19:04.09 | bmoraca_work | tzanger: never seen a fat mexican? every mexican in California over the age of 23 is fat. you need to get out more |
| 19:04.22 | tzanger | bmoraca_work: but that's an americanized mexican |
| 19:04.38 | bmoraca_work | no, not really |
| 19:04.45 | Katty | well i don't think that's very fair. |
| 19:04.54 | Katty | the majority of the world is actually scrawny. |
| 19:05.09 | [TK]D-Fender | [15:04]<tzanger>bmoraca_work: but that's an americanized mexican <- Is that like an africanized honeybee? |
| 19:05.10 | bmoraca_work | most mexicans in California (especially around where I am) don't "americanize". |
| 19:05.40 | bmoraca_work | but, that's not a topic for this channel |
| 19:05.43 | bmoraca_work | :) |
| 19:05.43 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 19:06.49 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:08.24 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:08.32 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 19:09.05 | leifmadsen | russellb: \o/ |
| 19:09.14 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0-beta5 (2010/09/05), 1.6.2.13 (2010/09/15), 1.4.36 (2010/09/15), *-Addons 1.6.2.1, 1.4.11 (2010/04/15), dahdi-linux 2.3.1 (2010/05/25), dahdi-tools 2.3.0 (2010/04/13), libpri 1.4.11.3 (2010/06/29) -=- Related channels: #asterisknow,#switchvox,#freepbx,#asterisk-dev,#asterisk-bugs,#astricon |
| 19:09.30 | Katty | i should go hang out in astricon for awhile. |
| 19:09.34 | Qwell | 1.6.2.13 has been releastd! |
| 19:09.38 | leifmadsen | :D |
| 19:09.43 | Katty | applauds. |
| 19:09.49 | bmoraca_work | too many version updates. uhg. |
| 19:10.02 | leifmadsen | Qwell: type fail |
| 19:10.07 | Qwell | leifmadsen: :D |
| 19:10.09 | leifmadsen | bmoraca_work: 1.6.2.13 is just 1.6.2.12 |
| 19:10.16 | leifmadsen | bmoraca_work: and it's only 1 per month |
| 19:10.22 | Chainsaw | It's in the topic, so it's official :) |
| 19:10.24 | Chainsaw | packages |
| 19:10.24 | Qwell | trick question! what version was 1.6.2.12? |
| 19:10.30 | leifmadsen | 1.6.2.12-rc1 |
| 19:10.33 | Qwell | WRONG |
| 19:10.43 | *** kick/#asterisk [Qwell!~Leif@asterisk/documenteur-extraordinaire/blitzrage] by leifmadsen (you're wrong!) |
| 19:10.47 | bmoraca_work | lol |
| 19:10.50 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 19:10.50 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:10.53 | leifmadsen | <3 |
| 19:12.47 | thehar | lol |
| 19:17.56 | Katty | goshdangitanyhow |
| 19:17.58 | Katty | i just broke a nail |
| 19:18.05 | Katty | throws things |
| 19:18.13 | Kobaz | if you keep them short then they dont break :P |
| 19:18.21 | Katty | but then they're not pretty :< |
| 19:18.57 | Kobaz | sure they are |
| 19:22.25 | Kobaz | we'll love you just as much with your short nails |
| 19:22.31 | Katty | :> |
| 19:22.43 | Katty | i still like my nails tho. |
| 19:24.51 | ritztech | dam gerbils |
| 19:26.28 | Chainsaw | Katty: Long actual nails can be quite nice. Those clip-on things are horrible though. |
| 19:27.32 | Katty | yeah i agree. |
| 19:30.42 | p3nguin | If the :3 in ${EXTEN:3} will shave off the first three digits of EXTEN, what will shave off the last three digits? EXTEN is possibly variable length. |
| 19:31.02 | nny | hmm ok now I am confused |
| 19:31.06 | nny | is rfc2833 inband? |
| 19:31.31 | nny | From the ITSP: Our trace shows we are getting the calls with G729 and inband DTMF (see screen shot shown below). Please note in the screen shot that each time a DTMF digit is pressed the trace shows the digits in the RTP stream which is "inband". |
| 19:33.14 | nny | screenshot of session http://www.atlantiatech.com/DTMF_1224.png |
| 19:33.53 | Qwell | Tell him Qwell said he's wrong. |
| 19:34.07 | nny | heh |
| 19:34.12 | bmoraca_work | rfc2833 isn't technically inband or out-of-band. it's a SIP notify. so it's inband to the SIP channel, but it's not part of the media stream |
| 19:34.20 | Qwell | No, it's not. |
| 19:34.26 | Qwell | It's in the RTP. |
| 19:34.39 | bmoraca_work | ahh, ok |
| 19:34.45 | nny | i was told specifically inband was audio tone played over the stream and undetectable via packet inspection |
| 19:34.50 | nny | is that wrong? |
| 19:34.58 | Qwell | No. |
| 19:35.00 | Naikrovek | no |
| 19:35.19 | Naikrovek | inband audio just looks like audio - dtmf or voice , to a packet analyser, will look the same |
| 19:35.26 | Naikrovek | you'll have to analyze the sound to know what's what |
| 19:35.51 | nny | that's eaxactly what I thought |
| 19:35.56 | nny | IP engineer my foot |
| 19:36.04 | *** join/#asterisk jamko (~Jim@173-162-11-81-naples.hfc.comcastbusiness.net) |
| 19:36.16 | Naikrovek | this is why in-band DTMF is to be avoided if possible. in-band to the AUDIO i'm talking |
| 19:36.37 | nny | anyone have a good link of the definition or RFC to send this asshat? |
| 19:36.45 | Naikrovek | ... 2833? |
| 19:36.53 | nny | i guess, more about what inband is |
| 19:36.56 | frigidzephyr | with DTMF via RFC2833 you'll see RTP packets with SDP headers that describe the dtmf event IIRC |
| 19:36.57 | nny | since his definition is wrong |
| 19:36.58 | Naikrovek | asshat won't read it, btw |
| 19:37.02 | nny | lol |
| 19:40.53 | Naikrovek | saw a bot named "one-man cheezeburger apocolypse" on a TF2 server last night. i laughed for 20 minutes. |
| 19:41.59 | nny | http://wiki.siemens-enterprise.com/index.php/DTMF seems to work |
| 19:42.05 | nny | says inband is in the voice band |
| 19:42.34 | Naikrovek | yes, in DTMF terms, the voice band is what is referred to as "in-band" or "out of band" |
| 19:42.46 | Naikrovek | not the RTP stream, it's all in-band if you look at it that way |
| 19:43.13 | nny | he says that it's inband because the tone is placed "in" the rtp stream |
| 19:43.17 | nny | swear to god that's the logic |
| 19:43.22 | nny | instead of "int he voice band" |
| 19:43.24 | nny | lol |
| 19:43.49 | nny | so out of band is when the dtmf is sent via USPS apparently |
| 19:46.04 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 19:47.28 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 19:48.18 | nny | <PROTECTED> |
| 19:48.20 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:48.36 | KavanS | nny, oh you'd be surprised... |
| 19:48.55 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
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| 19:49.25 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 19:49.26 | riscphree | must have some awesome hiring policies wherever that is at |
| 19:50.11 | [TK]D-Fender | inband? Isn't that the competitor to Guitar Hero? |
| 19:50.29 | nny | "click click click" man I love that song |
| 19:51.30 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:52.46 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 19:52.48 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 19:53.34 | ixti | Hello, everybody! Can somebody help with fax receiving? |
| 19:53.43 | bmoraca_work | maybe |
| 19:53.54 | ixti | SendFax() works good with both app_fax and res_fax (digium) |
| 19:54.00 | ixti | but ReceiveFax() |
| 19:54.11 | ixti | doesn't work with app_fax at all |
| 19:59.32 | *** join/#asterisk ixti (~ixti@217.70.249.69) |
| 19:59.39 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 19:59.40 | *** join/#asterisk ManxPower (~manxpower@24.236.78.136) |
| 20:00.08 | ixti | sorry my ISP dropped connection and I didn't received any message since my last reply if any |
| 20:00.12 | ManxPower | I am using Manager Originate: Is there a way to set separate Caller*ID info for each leg of the call without using something like chan_local? |
| 20:02.14 | ixti | so now I have no clue what I'm doing wrong |
| 20:02.14 | [Outcast] | is there any conditions where asterisk would not ACK a 200 ok |
| 20:02.53 | ixti | and would appreciate any thoughts :)) as i ran out of ideas :)) |
| 20:04.29 | [TK]D-Fender | ManxPower: Nope. |
| 20:05.59 | *** join/#asterisk giovani (~giovani@unaffiliated/giovani) |
| 20:06.44 | giovani | so I see a TON of iax2 channels when I do "iax2 show channels" and nearly all of them have "(None)" under the "Channel" header -- what's going on here? how do I clear out these clearly zombie/dead channels? |
| 20:09.09 | ManxPower | giovani, a registration and qualify creates channels. Do they show up in "core show channels"? |
| 20:09.53 | *** part/#asterisk nny (~Scott@cpe-174-107-201-103.sc.res.rr.com) |
| 20:10.34 | giovani | ManxPower: the ones labeled "(None)" do not show up there, no -- however, a lot of "dead" channels do show up in "core show channels" |
| 20:15.43 | giovani | restarting asterisk clears these -- but that's not really an acceptable solution -- I'd like to find the root cause |
| 20:16.33 | [TK]D-Fender | giovani: What version of * (which you should ahve mentioned walking in) |
| 20:17.26 | leifmadsen | I really wish you would actually type "Asterisk" instead of "*" which is terribly ambiguous |
| 20:18.53 | [TK]D-Fender | leifmadsen: Yeah, it's not like it's a giant part of the corporate logo or anything.... |
| 20:19.17 | [TK]D-Fender | leifmadsen: And because people also don't know the name for keys on their keyboards either :p |
| 20:19.22 | leifmadsen | I still read your sentences and half the time miss what you're talking about |
| 20:19.29 | [TK]D-Fender | leifmadsen: Which... we both know DOES fail occasionally. |
| 20:19.33 | leifmadsen | ya whatever |
| 20:19.50 | leifmadsen | I should not talk to you when I'm in anything but an ecstatic mood |
| 20:20.41 | riscphree | you guys are here just for my entertainment, aren't you? |
| 20:20.54 | Naikrovek | yep that's it |
| 20:21.02 | riscphree | I knew it |
| 20:21.14 | *** join/#asterisk Faithful (~Faithful@nat65.mia.three.co.uk) |
| 20:21.14 | [TK]D-Fender | riscphree: You are a very poorly designed figment of my imagination :p |
| 20:21.25 | riscphree | lol |
| 20:22.12 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 20:24.38 | [TK]D-Fender | checkout time, BBIAB |
| 20:25.24 | *** join/#asterisk normaldotcom (~normaldot@pc14474.cedarville.edu) |
| 20:26.02 | normaldotcom | Is there a way to stop asterisk from saying the parking extension number when a call is parked? |
| 20:27.43 | Naikrovek | why would you want it to not tell you the number |
| 20:27.49 | Naikrovek | you NEED that number to get back to the call |
| 20:28.18 | bmoraca_work | you can get that number other ways |
| 20:28.27 | normaldotcom | well, my dialplan actually handles it, and knows the extension |
| 20:28.32 | normaldotcom | it's for my googlevoice ringback |
| 20:28.49 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 20:29.19 | Naikrovek | oh well |
| 20:29.21 | Naikrovek | i dunno then |
| 20:30.38 | bmoraca_work | you could remove the digits from your sounds folder |
| 20:31.03 | Naikrovek | he may need them elsewhere? |
| 20:31.08 | normaldotcom | yeah, I thought about that, but I use them other places |
| 20:31.25 | bmoraca_work | it was just one option |
| 20:31.35 | Naikrovek | surely you could build a quietPark application from the park app source |
| 20:36.03 | Naikrovek | oh yeah that should be easy |
| 20:36.10 | Naikrovek | peruses app_parkandannounce.c |
| 20:36.23 | normaldotcom | lol |
| 20:36.32 | Naikrovek | i'm not being sarcastic |
| 20:36.38 | Naikrovek | i can't even write C but I think I could do this |
| 20:36.42 | Katty | that's wha.... |
| 20:36.47 | Katty | nevermind. |
| 20:36.50 | Naikrovek | lol |
| 20:36.56 | normaldotcom | I've done some c work, haven't looked at asterisk's source though |
| 20:37.04 | Naikrovek | http://svn.digium.com/svn/asterisk/trunk/apps/app_parkandannounce.c |
| 20:37.19 | Katty | in soviet russia! |
| 20:37.34 | Katty | goes back to lurkmode. |
| 20:38.46 | bmoraca_work | comment out "ast_say_digits(dchan, lot, "", dchan->language);" that line and recompile |
| 20:38.51 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 20:39.47 | normaldotcom | looks like that would work... |
| 20:40.07 | normaldotcom | ...sadly my current install is binary |
| 20:40.15 | normaldotcom | downloads sources |
| 20:40.34 | p3nguin | Is there an online resource to determine who owns a phone number? I mean like a number is a Verizon number or an AT&T number... not who you are calling if you call the number. |
| 20:40.56 | Naikrovek | like whois for numbers |
| 20:41.01 | Naikrovek | good question |
| 20:41.05 | bmoraca_work | p3nguin: www.localcallingguide.com is pretty useful, though not exactly up to date (if a number is ported, etc) |
| 20:41.43 | p3nguin | I'm okay with pre-port ownership at this point. |
| 20:41.50 | Naikrovek | www.telcodata.us |
| 20:41.57 | bmoraca_work | p3nguin: use this: http://www.localcallingguide.com/lca_prefix.php |
| 20:42.03 | p3nguin | cool |
| 20:42.04 | Naikrovek | http://www.telcodata.us/telcodata/telco |
| 20:42.13 | p3nguin | I'll try 'em both! |
| 20:42.20 | Naikrovek | damn right you will |
| 20:42.23 | Naikrovek | :P |
| 20:42.27 | p3nguin | cowers |
| 20:42.32 | bmoraca_work | i think telcodata.us uses localcallingguide |
| 20:43.05 | Naikrovek | wow it even shows you the exact location of the exchange building |
| 20:43.08 | Naikrovek | neato |
| 20:43.42 | p3nguin | very useful, indeed. |
| 20:43.49 | p3nguin | This is pretty much exactly what I was hoping for. |
| 20:44.11 | bmoraca_work | also has an XML interface, so you can automate certain things |
| 20:45.35 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:46.47 | Katty | there there, don't scare the p3nguin |
| 20:46.51 | Katty | hugs on p3nguin |
| 20:47.21 | p3nguin | :p |
| 20:58.53 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:00.46 | Katty | fenderbender |
| 21:06.22 | giovani | [TK]D-Fender: 1.4.21.1 |
| 21:06.48 | *** join/#asterisk korcan (~johnynum5@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 21:08.06 | korcan | can someone tell me where CID info is cached? I changed the name on a phone as an employee was replaced. everyones CID is still showing the old persons name. I factory reset a phone (SPA 962) thinking it was the phone and it still shows the old persons name. |
| 21:08.17 | korcan | the old person is not listed anywhere in my config |
| 21:08.20 | korcan | any ideas |
| 21:09.15 | seanbright | astdb? |
| 21:09.37 | seanbright | try running 'database show' from the asterisk CLI |
| 21:11.45 | korcan | seanbright: u talking to me? |
| 21:13.30 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 21:13.33 | [sr] | hi friends :) |
| 21:13.42 | [sr] | hi WIMPy, all ok? |
| 21:14.17 | Katty | hungry :< |
| 21:14.38 | *** join/#asterisk VxJasonxV (~jason@xmms2/troll/VxJasonxV) |
| 21:15.23 | korcan | anyone have any ideas on why cid isn't updating when I rename a person? |
| 21:16.04 | [sr] | hi Katty, i already had dinner two hours ago |
| 21:16.04 | [sr] | :P |
| 21:17.01 | [TK]D-Fender | korcan: go show us the call |
| 21:17.32 | korcan | [TK]D-Fender: what do you mean? the debug? |
| 21:17.35 | [TK]D-Fender | korcan: and * does not cache info that comes from a phone |
| 21:21.41 | VxJasonxV | I have a CallerID question, we're using a TDM2400, and I'm trying to figure out if we're even getting cid from our trunk. They say they send it, period, but I'm not sure how to verify that. |
| 21:22.07 | VxJasonxV | I'm under the assumption that I should see it at the console with high verbosity (or perhaps high debug). But I'm not sure where. |
| 21:22.36 | VxJasonxV | For example, the Answer() line says "Answer("DAHDI/3-1", "")". Should the number be on the right side of that? |
| 21:23.07 | VxJasonxV | (Long and the short of it: how do I easily see the calling number in asterisk, the quickest way). |
| 21:25.02 | *** join/#asterisk mmlj4 (~jkelly@ip70-171-94-246.no.no.cox.net) |
| 21:25.11 | [TK]D-Fender | VxJasonxV: No. If you want it to appear in CLI then you are going to have to NoOp / Verbose it, etc |
| 21:25.42 | VxJasonxV | thought that'd be the case. |
| 21:25.48 | VxJasonxV | NoOp(CALLERID(number)) ? |
| 21:26.45 | *** part/#asterisk exothermc (~miles@174.127.153.10) |
| 21:26.59 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 21:27.27 | Kobaz | NoOp(${CALLERID(num)}) |
| 21:27.41 | VxJasonxV | so I was close |
| 21:27.47 | Kobaz | but no donut |
| 21:27.59 | ChannelZ | mmmmm doonnnuuutttt |
| 21:28.36 | VxJasonxV | everbody loves donuts... |
| 21:32.10 | korcan | can anyone recomend a good wifi phone? |
| 21:32.29 | [TK]D-Fender | korcan: No such thing :) |
| 21:32.44 | korcan | are they that bad? |
| 21:33.35 | Kobaz | yeah |
| 21:33.38 | [TK]D-Fender | korcan: what are your actual needs? |
| 21:33.39 | Kobaz | and expensive |
| 21:33.59 | korcan | linksys wip330? ~$230 |
| 21:34.01 | Kobaz | the new polycom dect phones look nice... i haven't used them yet |
| 21:34.18 | Kobaz | dect -> sip |
| 21:34.48 | Kobaz | http://www.polycom.com/products/voice/wireless_solutions/dect_communications/index.html |
| 21:34.49 | [TK]D-Fender | [17:33]<korcan>linksys wip330? ~$230 <- specifically horrible garbage |
| 21:35.13 | korcan | [TK]D-Fender: :) |
| 21:36.48 | VxJasonxV | uhh...huh |
| 21:37.18 | korcan | so really none? cisco 7925G? |
| 21:37.33 | VxJasonxV | http://pastie.org/private/5qev854ozyqsgi7xr4a |
| 21:37.48 | VxJasonxV | Does this mean that we're not getting caller id? |
| 21:38.14 | [TK]D-Fender | VxJasonxV: it means ASTERISK isn't yet |
| 21:38.33 | VxJasonxV | your use of the word "yet" interests me |
| 21:38.50 | [TK]D-Fender | VxJasonxV: considered plugging a normal phone on the line and testing it? |
| 21:38.58 | VxJasonxV | I don't have a normal phone |
| 21:39.03 | VxJasonxV | Well |
| 21:39.14 | VxJasonxV | I have a phone with no screen, so I can't read callerid. |
| 21:39.40 | VxJasonxV | the phone number shouldn't come in after the fact, should it? I understand name and what not, but the trunk should be passing the number along nearly immediately. |
| 21:39.44 | VxJasonxV | ...right? |
| 21:39.47 | [TK]D-Fender | [17:38]<VxJasonxV>your use of the word "yet" interests me <- Good. Intentional grammar has not eluded you. I don't know if you configured your channels to support it in the fist place. Maybe, just maybe you should show us... |
| 21:40.37 | VxJasonxV | I'm trying to remember what file it's in ~_~ |
| 21:40.51 | VxJasonxV | I thought it was outside /etc/asterisk, in /etc/dahdi, but I'm obviously wrong |
| 21:40.54 | VxJasonxV | dahdi channels in /etc/asterisk it is |
| 21:41.50 | VxJasonxV | http://pastie.org/private/zica7ljjqaxe0lvxvo9yaa all the channels look like that, with the channel number difering. |
| 21:41.55 | VxJasonxV | differing* |
| 21:42.27 | VxJasonxV | hmm. the double definition of group |
| 21:42.43 | ixti | does limiting of maxbaudrate may help in solving "can't ReceiveFax()" issue? |
| 21:42.52 | VxJasonxV | don't think that should be relevant at the moment |
| 21:43.50 | ixti | because i just remembered that when i was testing it with russia - baudrate was selected 2400 |
| 21:48.29 | Kobaz | in soviet russia, baud rate selects YOU! |
| 21:49.14 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 21:49.19 | [TK]D-Fender | VxJasonxV: usecallerid=yes |
| 22:00.31 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 22:02.00 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 22:04.43 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 22:09.30 | *** part/#asterisk ManxPower (~manxpower@24.236.78.136) |
| 22:10.11 | ixti | Kobaz: seems like yes ;)) |
| 22:13.03 | voxter | Question. I have a SIP Peer who's context is set to "outgoing" (my outgoing context, of course) - when this SIP peer does call forwarding it comes back to Local/<fwdednumber>@inbound (my default context) - Is there a way to make it so that redirects end up in the context of the originating peer's context= definition? |
| 22:15.08 | jamko | With canreinvite=no , shouldn't asterisk still allow T.38 reinvite? |
| 22:18.08 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 22:19.43 | Qwell | jamko: That option is misleading. It doesn't disable reinvites. It disables direct RTP bridging |
| 22:19.56 | Qwell | (which is why it's been renamed in 1.8) |
| 22:21.28 | jamko | @Qwell: Thank you.. |
| 22:22.30 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 22:24.23 | jamko | What if you had canreinvite=no, but directrtpsetup=yes .. which would take precedent? Just for discussion sake, not that one would do that. |
| 22:25.55 | Qwell | depends on the order |
| 22:26.39 | VxJasonxV | [TK]D-Fender, what is 'yes' vs. 'asreceived' ? |
| 22:26.44 | Qwell | same thing that would happen if you had "canreinvite=no"/"canreinvite=yes" |
| 22:28.20 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 22:30.21 | [TK]D-Fender | VxJasonxV: completely separate parameter also required |
| 22:35.02 | jamko | ok .. logical enough. |
| 22:38.55 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:57.09 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 23:02.22 | *** join/#asterisk myster (~myster@d161-184-142-34.abhsia.telus.net) |
| 23:08.29 | *** join/#asterisk ShadowApex (~william@adsl-99-107-163-198.dsl.lsan03.sbcglobal.net) |
| 23:20.15 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 23:21.35 | joobie | guys I got an issue where users are complaining that some calls they make they find it very hard to hear the other person.. I have since setup Monitor() so that I can record the calls they make and when i listen to the call, they sound find but the remote party sounds very very quiet. The users are connected to asterisk via SIP and when they dial the remote party, it's going via an asterisk SIP peer via pennytel |
| 23:21.52 | joobie | just wondering what i can do on my end to try improve the volume / call quality - pennytel dont really help much |
| 23:24.00 | riscphree | google gives me About 56,500 results for "asterisk improve call quality" |
| 23:25.14 | joobie | that's nice |
| 23:25.45 | joobie | google gives me about 1,830 results for "irc ass bandits" |
| 23:25.46 | pabelanger | joobie: sounds like an issue with your phone |
| 23:26.09 | joobie | pabelanger, hmm.. how come? |
| 23:26.36 | joobie | pabelanger, i did the recording on asterisk itself.. if i listen to the user they are normal / loud.. if i listen to the remote party, they are soft |
| 23:26.43 | joobie | but that recording is before it reaches the phone speaker i thought |
| 23:27.24 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 23:27.29 | pabelanger | joobie: If your remote party is _soft_ in your recording, that just confirms to problem. |
| 23:27.39 | pabelanger | Are you doing an transcoding of audio within asterisk? |
| 23:28.03 | joobie | how does it confirm the problem though pabelanger ? interested in getting my head around the logic why |
| 23:28.16 | joobie | i am.. the user is g729 and pennytel is not |
| 23:28.20 | joobie | i think they are alaw from memory |
| 23:28.34 | joobie | sec ill check |
| 23:29.20 | joobie | hmm |
| 23:29.24 | joobie | how can i see which codec the user is using? |
| 23:29.36 | joobie | tried 'sip show peer <ext>' and it just shows the codec options there |
| 23:29.49 | pabelanger | joobie: I would stop transcoding audio, and see if the problem still exists. If it still happens, I would suspect a phone problem |
| 23:30.00 | *** join/#asterisk n0tk (~n0tk@216.160.42.30) |
| 23:30.30 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 23:30.54 | joobie | pabelanger, how can i see what codec a peer is using so that i can confirm there is no transcoding between the user / peer |
| 23:30.58 | *** part/#asterisk normaldotcom (~normaldot@pc14474.cedarville.edu) |
| 23:31.30 | pabelanger | *CLI> sip show peer <id> |
| 23:32.11 | joobie | i tried that.. what i get back of relevance is "Codecs : 0x108 (alaw|g729)" "Codec Order : (g729:20,alaw:20)" |
| 23:32.27 | joobie | .. but it doesnt really confirm the user is using g729 does it? |
| 23:33.33 | pabelanger | no, the only real way would be to enable a SIP debug an look at the INVITE message. Or you can configure both disallow=all; allow=alaw |
| 23:33.40 | joobie | ahh |
| 23:34.21 | joobie | true.. i guess each call can use watever |
| 23:34.55 | pabelanger | I would remove g729 from the picture, and confirm what happens using alaw. |
| 23:35.34 | joobie | ive gone the other route |
| 23:35.36 | joobie | and removed alaw |
| 23:35.47 | *** join/#asterisk oryxtec (~asda@119.158.36.248) |
| 23:35.50 | joobie | becauase bandwidth is an issue here.. would prefer if i can nail this and keep the user on g729 |
| 23:36.54 | oryxtec | i having a really issue all settings seems to be fine to but i am unable to dial out.. please can some 1 have a look http://pastebin.com/VXJyUWYA |
| 23:37.03 | pabelanger | That's fine, I only suggested alaw, since g729 codec does not ship within asterisk. And if there is a problem, we couldn't do to much to fix it. |
| 23:37.13 | joobie | ahh |
| 23:37.20 | joobie | i have just foreced pennytel to g729 |
| 23:37.27 | joobie | and forced this user also to g729 |
| 23:37.31 | joobie | so no more transcoding... |
| 23:38.05 | pabelanger | oryxtec: *CLI> sip set debug on |
| 23:38.15 | pabelanger | oryxtec: retry your outbound call and post the results |
| 23:38.19 | oryxtec | ok |
| 23:38.21 | joobie | BUT hmm.. if this fails pabelanger, why do you think it could be the phone? I thought by recording on the asterisk box, I would be recording the phone's mic quality on my users end and then recording the remote party's mic quality, via the pennytel stream, independant of my user's phone |
| 23:38.24 | oryxtec | plz hold |
| 23:40.04 | oryxtec | pabelanger: i just did set debug on.. but same results |
| 23:40.11 | oryxtec | on asterisk CLI |
| 23:40.13 | pabelanger | joobie: Asterisk does not do anything to the audio stream. And since it always originates from a phone, it is likely the problem. You can try adjusting the +/- audio gain on your phone, other wise you will need to setup asterisk to amplify the audio for this extension |
| 23:40.25 | pabelanger | ~collectdebug |
| 23:40.25 | infobot | somebody said collectdebug was a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 23:40.28 | pabelanger | oryxtec: ^^ |
| 23:40.56 | *** join/#asterisk exothermc (~miles@74.85.89.146) |
| 23:41.27 | exothermc | what would cause: Warning: Failed to connect to the agentx master agent ([NIL]): ? |
| 23:42.20 | joobie | pabelanger, i see.. but when i record on asterisk, the user's voice (my user) is OK volume wise.. it's the remote party they are dialling, their voice is very quiet.. Doesn't this mean that it's the remote party's phone that's the problem, not my end users phone?? |
| 23:42.52 | *** part/#asterisk rsp2k (~rpm@unaffiliated/rsp2k) |
| 23:42.54 | pabelanger | joobie: Do you have control of the remote parties phone? |
| 23:43.28 | joobie | no |
| 23:43.37 | pabelanger | joobie: What path does the outbound call take? IE: Agent phone -> Asterisk -> PRI -> Telco |
| 23:44.12 | joobie | Agent phone (SIP,g729 forced now) -> Asterisk -> Pennytel (SIP,g729 forced now) |
| 23:44.14 | *** part/#asterisk ShadowApex (~william@adsl-99-107-163-198.dsl.lsan03.sbcglobal.net) |
| 23:44.21 | joobie | Pennytel then sends the call international |
| 23:44.48 | pabelanger | Are all calls very quiet? |
| 23:44.56 | joobie | yes |
| 23:45.04 | joobie | i listened to about 5 calls |
| 23:45.15 | joobie | i had to pump my volume right up and was still struggling |
| 23:45.30 | pabelanger | joobie: then I would get Pennytel involved, sounds like a problem with their equipment |
| 23:45.39 | joobie | ahhh |
| 23:45.50 | joobie | well ive just forced this g729 as a start |
| 23:45.50 | pabelanger | joobie: Try the gsm codec, just to confirm |
| 23:45.55 | joobie | we'll see if that fixes it |
| 23:46.03 | joobie | why the gsm codec? |
| 23:46.07 | oryxtec | where logger.conf is located? |
| 23:46.10 | joobie | the thing is with pennytel, they suck with support |
| 23:46.24 | joobie | it's a nightmare even trying to get them to understand you let alone provide good tech support |
| 23:46.24 | pabelanger | joobie: to rule out that g729 codec is the issue |
| 23:46.28 | joobie | ahh |
| 23:46.32 | pabelanger | oryxtec: usually /etc/asterisk |
| 23:46.44 | joobie | pabelanger, can i use alaw instead? like is there a reason you said specifically gsm ?? |
| 23:46.46 | oryxtec | thnk |
| 23:47.08 | pabelanger | joobie: it uses less bandwidth then alaw |
| 23:47.30 | pabelanger | the codec choice does not matter, just use something other then g729 |
| 23:47.49 | joobie | ahh i see |
| 23:48.41 | joobie | pabelanger, presuming the problem exists across different codecs and pennytel dont help (which is likely), is increasing the gain on asterisk the next step? |
| 23:49.09 | pabelanger | joobie: Yes, that would be your only choice. |
| 23:49.53 | pabelanger | And if your ITSP will not help resolve your issue, I would strongly suggest moving to a new provider |
| 23:51.12 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
| 23:51.12 | joobie | yea |
| 23:51.15 | joobie | im toying with that idea too |
| 23:51.22 | oryxtec | i have follow all the steps but myDebugLog at this /var/log/asterisk/ location is not get created :S |
| 23:51.23 | joobie | pennytel's support sucks. |
| 23:51.45 | joobie | pabelanger, what settings on asterisk can i use to increase the gain just on that one channel? |
| 23:55.09 | pabelanger | joobie: *CLI> core show function VOLUME |
| 23:55.30 | joobie | i use 1.4 |
| 23:55.36 | oryxtec | http://pastebin.com/VXJyUWYA |
| 23:55.37 | joobie | .. "No function by that name registered." |
| 23:55.38 | oryxtec | plz check now |
| 23:55.51 | oryxtec | pabelanger: ~~ |
| 23:57.57 | pabelanger | joobie: Hmm, would have to look for 1.4 |
| 23:58.11 | pabelanger | oryxtec: I don't see you SIP debug log |
| 23:58.32 | oryxtec | http://pastebin.com/sgBmkKEy |
| 23:58.37 | oryxtec | at the end |
| 23:58.41 | oryxtec | have a look |
| 23:58.51 | joobie | looks like a good feature tho :P |
| 23:59.21 | pabelanger | oryxtec: Is that the only output? You are missing some messages |
| 23:59.57 | pabelanger | oryxtec: you can drop the settings and dialplans from the pb |
