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| 00:23.26 | bradphone | hmmhesays: I am so far :P |
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| 00:34.43 | [TK]D-Fender | ~wifisip |
| 00:34.44 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
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| 00:38.46 | peanut- | good thing this is a soft phone then.. |
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| 00:39.14 | peanut- | reading online that voicepulse's IAX2 is always choppy... is that accurate or just a bunch of people with poor configs? |
| 00:39.24 | JT | use SIP |
| 00:39.55 | peanut- | I don't want to use SIP |
| 00:40.59 | [TK]D-Fender | ~sofphone |
| 00:41.02 | [TK]D-Fender | ~softphone |
| 00:41.02 | jbot | something that should be drug out into the street and shot |
| 00:41.24 | peanut- | yes, there's someone that hates everything, I got it. |
| 00:41.34 | [TK]D-Fender | peanut-, http://lolcat.com/emokitten.html |
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| 00:41.39 | peanut- | iax2 is bad, soft phones and 802.11 phones are horrible.. |
| 00:41.57 | [TK]D-Fender | peanut-, Wow, you've learned so much... in one sentence! |
| 00:43.12 | peanut- | if I relently bash an obscure component of VoIP do I get ops? |
| 00:43.26 | [TK]D-Fender | peanut-, lol |
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| 00:52.24 | Iamnach0 | I installed the SVN version of asterisk and addons to play around with chan_mobile, does anyone know anything/had problems with sip calls in the SVN version of asterisk? I know that I can revert back to 1.4.13 and my sip calling is normal... just looking for a bit of info. |
| 00:53.00 | Iamnach0 | should have said too: i have no audio on sip to sip calls with SVN. thats why i am wondering. |
| 00:56.49 | Iamnach0 | echo tests with SVN fail, hello world passes, with reg astersik echo tests pass. |
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| 00:57.55 | Iamnach0 | with SVN i get these errors sometimes: WARNING[8063] chan_sip.c: Hanging up call (somelongstring). - no reply to our critical packet |
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| 01:03.12 | JT | peanut-: or perhaps you should realise that these are actually fact and not just one person "hating everything" |
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| 01:09.17 | JT | peanut-: so, in your infinite wisdom; why do you not want to use SIP? |
| 01:09.35 | J4k3 | peanut's one of those typical dilusional white trash texans who think if they vote republican they'll get rich, or go to heaven. |
| 01:09.49 | J4k3 | this should give you some idea of what you're working with. |
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| 01:16.10 | JT | heh |
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| 01:16.55 | J4k3 | he's also a great fan of the US/Iraq war. |
| 01:20.14 | peanut- | no real reason not to use SIP at the moment, I just want to give iax it's fair chance since it is the native protocol |
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| 01:29.28 | JT | peanut-: that's not really true |
| 01:29.42 | JT | peanut-: asterisk is NOT native VoIP |
| 01:30.09 | JT | and chan_sip is much more stable than chan_iax |
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| 01:31.36 | ectospasm | IAX doesn't have nearly the trouble that SIP has with NAT traversal |
| 01:32.02 | JT | ectospasm: sip has no problem traversing NAT if configured right |
| 01:32.15 | JT | iax has major issues with scalability |
| 01:32.24 | JT | and it's not a widely supported protocol |
| 01:32.25 | ectospasm | Right, but a lot of users can't do that properly |
| 01:32.38 | JT | should they be configuring PBXes? |
| 01:32.39 | ectospasm | Or, they try to traverse two NATs out of the box... |
| 01:32.48 | [TK]D-Fender | IAX "native", lol.... |
| 01:32.49 | ectospasm | JT: more and more are... |
| 01:32.55 | [TK]D-Fender | What will they come up with next! |
| 01:33.09 | JT | [TK]D-Fender: who knows |
| 01:33.22 | [TK]D-Fender | JT : We clearly aren't taking enough drugs.... |
| 01:33.25 | JT | ectospasm: idiots should not be configuring PBXes, that hasn't changed |
| 01:33.27 | JT | yeah |
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| 01:34.04 | ectospasm | What, with PBX Appliances appearing everywhere? Idiots will buy them, nonetheless |
| 01:35.09 | JT | an applicance is already setup |
| 01:35.14 | JT | all the nitty gritty is done |
| 01:35.21 | JT | they just need to answer simple questions |
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| 01:41.05 | kavelot | I have a very simple dialplan on asterisk (like answer, wait 2s and hangup)... but when I call with SIP debug enabled, I see my calling number (so it recognizes my call) and then I hear the busy sound (with message "unavailable network")... any hints on what I should check? |
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| 01:55.05 | JT | TrN: ? |
| 01:55.20 | TrN | ? |
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| 01:57.29 | WilliamK | good evening JT |
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| 02:03.52 | JT | hello WilliamK |
| 02:09.07 | mosty | when i use chan_local, the duration and billsec cdr values are wrong unless i use /n. will adding the /n part break anything? i don't quite understand what the voip-info.org page is saying when it describes this feature |
| 02:09.43 | [TK]D-Fender | mosty, its a good thing. |
| 02:10.58 | mosty | am i correct in thinking that the /n means the duration time is reset to start when the Dial(LOCAL/...) is executed? |
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| 02:15.09 | [TK]D-Fender | mosty, somthieng like that... it sorta gives it exclusivity or something.... can't remember... I heard it only once... |
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| 02:19.27 | mosty | thanks |
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| 02:21.53 | peanut- | heh. it's only choppy when sending to a caller when I originate the call |
| 02:21.55 | Edwin_Quijada | hi |
| 02:21.57 | peanut- | otherwise, it's fine both ways |
| 02:22.09 | Edwin_Quijada | i have an error compiling asterisk from scratch |
| 02:22.29 | peanut- | anyone know a probable cause of that? |
| 02:22.47 | Edwin_Quijada | it cant have -lssl |
| 02:22.53 | Edwin_Quijada | anybody knows? |
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| 02:24.18 | peanut- | um... maybe you should install ssl? |
| 02:24.24 | peanut- | or try #linux as it's not asterisk specific |
| 02:24.57 | fujin | install the openssl development libraries for your distro |
| 02:24.59 | fujin | what distro? |
| 02:25.04 | Edwin_Quijada | peanut- i can compile zaptel and libpri |
| 02:25.08 | Edwin_Quijada | but not asters |
| 02:25.13 | fujin | they're not dependant upon ssl |
| 02:25.25 | Edwin_Quijada | openssl? |
| 02:25.31 | fujin | indeed |
| 02:25.33 | fujin | what distro? |
| 02:25.37 | Edwin_Quijada | debian |
| 02:26.05 | fujin | apt-get install uh; should be apt-get install libssl0.9.8 |
| 02:26.12 | fujin | apt-cache search openssl|grep lib |
| 02:26.24 | fujin | or even try apt-get build-deps asterisk |
| 02:26.29 | fujin | should pull all the dependancies to build it |
| 02:26.43 | Edwin_Quijada | i dont know that aster needs ssl? |
| 02:26.49 | fujin | welp, now ya do |
| 02:27.21 | Edwin_Quijada | i try apt-get build first |
| 02:27.37 | [TK]D-Fender | www.asterisk.org <- read the damn prerequisites... its not Raw Cat science... |
| 02:28.06 | JT | peanut-: problems compiling asterisk are problems that can be supported here |
| 02:28.21 | JT | peanut-: re: choppy calls, did you try my earlier suggestion? |
| 02:30.09 | Edwin_Quijada | [TK]D-Fender: there is no prerequisites |
| 02:30.16 | Edwin_Quijada | in he site |
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| 02:31.06 | [TK]D-Fender | Edwin_Quijada, http://www.asterisk.org/support/install |
| 02:31.14 | [TK]D-Fender | Edwin_Quijada, Yeah... you looked REAL hard.... |
| 02:31.20 | [TK]D-Fender | openssl, and associated -devel |
| 02:31.21 | peanut- | JT: sure they can be, but is this really the place for basic "how does linux work?" questions? |
| 02:31.35 | [TK]D-Fender | NEXT!@!!@ (c) BKW |
| 02:31.57 | peanut- | JT: and do you mean to ask me if I connect to voicepulse with SIP instead of IAX2? |
| 02:35.13 | crudpuppy | peanut-, are you? and if so why |
| 02:40.32 | dan__t | wordup, [TK]D-Fender |
| 02:41.21 | JT | peanut-: i said to make sure all silence supression/detection is disabled |
| 02:41.28 | JT | for choppy voice |
| 02:41.30 | dan__t | I *will* get this Polycom working tonight. |
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| 04:00.05 | Tond | HI is there a way that i cna listen to sip to sip call conversations live? The same way ZapBarge works? |
| 04:00.31 | Tond | I need to do this for quality and training purpuses for my agents taking calls |
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| 04:21.00 | peanut- | JT: oh, yes, I made sure silence suppression was disabled, it didn't change anything. I added jitterbuffer between SIP softphone and asterisk and it cut down alot. |
| 04:21.44 | peanut- | must have been someone else that told me to do SIP instead of IAX2 to voicepulse.. |
| 04:22.37 | JT | i did say to use sip instead of iax2 |
| 04:22.43 | JT | especially if it was iax2 trunking |
| 04:23.20 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 04:24.33 | marc7 | i asked this question earlier today, but i'm not entirely sure this is working... in asterisk's sip.conf, is it possible to have a username as "person@domain.com"? it seems like that's a bad practice... you wouldn't want to have user@domain.com@sip.provider.net |
| 04:27.17 | JT | marc7: doesn't sound like a very good idea |
| 04:29.57 | marc7 | is there any articles on asterisk virtualization? eg... if joe@10.0.1.44 connected, it would be different than joe@10.0.1.55 |
| 04:30.08 | marc7 | or is that slated for a future branch |
| 04:32.21 | JT | ? |
| 04:34.32 | marc7 | if we wanted to host asterisk for more than one business, and there was a steve@domain1.com and a steve@company2.net... it seems like we should either be running two completely different asterisk instances for both companies... because there's no easy way of having two 'steve' users in sip.conf |
| 04:34.53 | JT | probably |
| 04:35.15 | marc7 | we should either be running two instances... or there's a feature set i just haven't heard of |
| 04:35.39 | JT | well |
| 04:35.48 | JT | you can use a database frontend |
| 04:36.02 | JT | and have all users on the one domain |
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| 04:52.16 | gardo | I'm having challenges with my te405p card. |
| 04:53.35 | peanut- | how do I log both ANI and CPN of incoming calls? |
| 04:53.48 | gardo | can anyone check http://www.pastebin.ca/742025 |
| 04:57.20 | gardo | i can't seem to get a line to dialout |
| 05:00.03 | JT | peanut-: record $(CALLERID(ani)} |
| 05:00.53 | peanut- | jt: thanks |
| 05:02.43 | peanut- | where exactly would that go? messages log? |
| 05:03.01 | JT | that's the name of the function |
| 05:03.08 | JT | how you log it is up to you |
| 05:08.18 | peanut- | damnit. |
| 05:08.54 | peanut- | messages => record $(CALLERID(ani)} or something of that sort? |
| 05:09.25 | JT | ~thebook |
| 05:09.26 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
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| 05:10.21 | peanut- | yea I've been looking through that |
| 05:10.40 | peanut- | oh I think I found it |
| 05:10.44 | peanut- | my search-fu is poor. |
| 05:18.32 | i3inary | so um...any limitation as to how many files can sit in your monitor directory? |
| 05:18.49 | i3inary | in /var/spool/asterisk/monitor |
| 05:19.16 | JT | not to my knowledge |
| 05:19.57 | i3inary | cause my .call files seem to be really lagged and i just noticed i have over 11k files in the mon dir |
| 05:20.54 | i3inary | anyone have any good scripts to tar up by date recordings? |
| 05:22.03 | peanut- | JT: is there something you can put in cdr_custom.con to get it to log ani? |
| 05:22.10 | peanut- | it's not covred in the book |
| 05:22.46 | JT | i'm sure cdr_custom.conf is documented somewhere |
| 05:23.19 | peanut- | you'd think so |
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| 05:24.08 | kaldemar | peanut-: take a look at cdr_custom.conf.sample |
| 05:25.37 | peanut- | it's the same as my cdr_custom.conf |
| 05:26.59 | kaldemar | do you think you could try putting "$(CALLERID(ani)}" in there somewhere? |
| 05:32.26 | peanut- | I don't think it works that way. |
| 05:32.42 | peanut- | [Oct 19 05:32:44] WARNING[7983]: cdr_custom.c:97 load_config: Failed to reload configuration file. |
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| 06:51.09 | McDouglas | any suggestions about which wifi phone will work with asterisk? |
| 06:52.04 | McDouglas | (i'm aware of that " all wireless phones suck" quote, but still... i need wireless functionality for at least a few workstations) |
| 06:54.36 | JT | DECT phones + ATAs |
| 06:54.54 | McDouglas | :\ |
| 06:55.08 | McDouglas | how about wifi ones? we have APs in every room |
| 06:55.20 | JT | yeah but it's a poor solution... |
| 06:55.28 | McDouglas | why? |
| 06:55.49 | JT | because the audio quality will be terrible and tempermental |
| 06:56.03 | McDouglas | hmm, even if the wifi coverage is exvelent? |
| 06:56.08 | McDouglas | *c |
| 06:56.31 | McDouglas | we dont have too many foreign aps near either |
| 06:56.33 | JT | unless you can make your users stand perfectly still... |
| 06:57.35 | McDouglas | can't i just send back the phones if the voise quality doesnt meet the usability levels? |
| 06:57.38 | McDouglas | *voice |
| 06:57.59 | JT | i don't know |
| 06:58.06 | McDouglas | i was looking at this btw: http://www.voip-info.org/wiki/view/WIP330 |
| 06:58.07 | JT | would seem hard if they weren't actually faulty |
| 06:58.26 | JT | yeah but the wifi protocol is unsuited to mobile voip |
| 06:58.38 | McDouglas | oh well |
| 06:58.48 | McDouglas | i guess we have to buy some ata then |
| 06:58.55 | JT | packet loss |
| 06:58.58 | JT | jitter |
| 06:59.02 | JT | half duplex |
| 07:00.09 | McDouglas | we have some pansonic dect phones |
| 07:00.24 | McDouglas | will it support call id if we use ata to connect them to asterisk? |
| 07:00.44 | JT | if the phones and the ata supports it |
| 07:03.59 | *** join/#asterisk synthetiq (n=roger@unl201395.nl.customer.alter.net) |
| 07:04.38 | synthetiq | hello, i am trying to use the externnotify feature in voicemail, but my script is never being called...i wonder if anyone has any ideas what would casue this...i thought it was permissions but its not... |
| 07:08.42 | peanut- | McDouglas: I ordered that phone, it'll be here saturday |
| 07:08.57 | peanut- | I didn't buy the "all wifi phones suck" line |
| 07:09.05 | McDouglas | peanut-: actually.. i have been reading reviews right now and they are dissapointing |
| 07:09.59 | peanut- | poor experiences are more likely to generate reviews than good |
| 07:10.08 | JT | peanut-: more like you didn't check before buying |
| 07:10.13 | coppice | but all wifi phones do suck. its their fundamental nature to suck |
| 07:10.16 | JT | and it's not just a product problem |
| 07:10.19 | JT | it's a technology issue |
| 07:10.38 | peanut- | so maybe it can be tweaked to be better |
| 07:10.43 | coppice | run any phone over wifi and it sucks |
| 07:10.58 | peanut- | what's the cause of the suckage? doppler? radio latencyt? |
| 07:11.06 | coppice | you can polish a turd, but its still a turd |
| 07:11.23 | coppice | latency, lack of QoS, and various other issues |
| 07:11.35 | coppice | wifi was never designed for streaming |
| 07:12.10 | coppice | you can set up great VoIP over wifi demos. day to day use is something completely different |
| 07:12.53 | coppice | actually, wifi was never really designed to be in any way fair |
| 07:13.41 | peanut- | all these mooks that say the wip300 sucks with the wrt54g "filed complaints"... I mean WTF |
| 07:14.00 | peanut- | they didn't try to hack it with dd-wrt or openwrt or anything |
| 07:14.04 | peanut- | just "QoS is broken" |
| 07:14.18 | coppice | there is *no* QoS over wifi |
| 07:14.44 | coppice | there was work on a 802.11something to add QoS features, but it never went anywhere |
| 07:17.20 | peanut- | I'm pretty sure openwrt is making progress with 802.11e |
| 07:18.46 | JT | peanut-: products are meant to work out of the box, they are not meant to require hacking with unsupported firmware. |
| 07:19.06 | JT | quite an unreasonable expectation |
| 07:19.27 | coppice | 802.11e is useless unless everyone implements it. how aggressively is it being implemented? |
| 07:20.01 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 07:20.48 | penguinFunk | 802.11n is out |
| 07:20.52 | penguinFunk | might as well go for that |
| 07:20.54 | penguinFunk | 300Mbps |
| 07:21.04 | coppice | it still has no QoS |
| 07:21.14 | JT | coppice: and in any case, what does it have to do with the issue of wifi actually sucking *now*? ;) |
| 07:22.01 | coppice | a great deal. if 802.11e were a base station only thing, things could be sorted out very quickly |
| 07:22.31 | JT | but people are just going to buy wifi phones and expect them to work on <insert-random-wlan> |
| 07:22.40 | JT | most people don't upgrade firmware |
| 07:22.53 | penguinFunk | must be very difficult to implement QoS on wifi standards |
| 07:23.02 | coppice | sure, but its the difference between fixable and not fixable |
| 07:23.29 | coppice | penguinFunk: I think that would be why it stalled for so long |
| 07:23.38 | penguinFunk | yeah most definitely |
| 07:24.05 | penguinFunk | i guess there is demand issue too? |
| 07:24.16 | penguinFunk | only VOIP people usually require QoS |
| 07:24.27 | penguinFunk | most other people want more bandwidth |
| 07:24.31 | coppice | I think there is a lot of demand. many people want to stream video over wifi |
| 07:24.46 | penguinFunk | hence 802.11n = 300Mbps |
| 07:24.48 | coppice | video over wifi is a key driver for 802.11n |
| 07:24.50 | penguinFunk | true |
| 07:25.39 | penguinFunk | anyone use voip/asterisk whatever for video calls yet? |
| 07:25.46 | coppice | "Business and consumer products using 802.11e are expected to become widely available in late 2004 or in 2005." - yeah, right :-) |
| 07:25.53 | penguinFunk | lol |
| 07:25.58 | penguinFunk | bit late aren't they |
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| 07:26.35 | coppice | some things in a web search say 802.11e was ratified, but I don't think that's true. |
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| 07:44.47 | McDouglas | and what if voip is the only thing which would use the wifi connection? i wouldnt need QoS then, right? |
| 07:47.57 | coppice | depends how many phones you have..... and how many freeloaders sending data :-) |
| 07:48.18 | penguinFunk | exactly McDouglas |
| 07:48.40 | penguinFunk | secured wifi = no freeloaders |
| 07:48.58 | coppice | :-) == don't take it literally :-) |
| 07:49.28 | McDouglas | well, frankly we dont use our wifi network at all... it was built in case we have some guests :P |
| 07:49.46 | coppice | ah, a pro-freeloader network |
| 07:49.56 | McDouglas | and only 3-4 person would be required to have wifi phone |
| 07:50.07 | McDouglas | of course, its secured |
| 07:50.22 | penguinFunk | definitely worth testing McDouglas |
| 07:50.37 | McDouglas | i would test.. too bad noone gives us test phones :P |
| 07:50.53 | penguinFunk | still think wifi would not be able to keep up for other reasons though |
| 07:50.55 | McDouglas | and its a bit expensive to buy them jsut to figure out they arent working well |
| 07:51.12 | penguinFunk | McDouglas: just buy one for now? |
| 07:51.41 | McDouglas | we are a small company.. even one would cost a lot of money to waste :\ |
| 07:51.53 | McDouglas | so they wouldnt approve it |
| 07:52.11 | McDouglas | existing dec+ata is cheaper solution |
| 07:52.12 | penguinFunk | cant you use wired phones? |
| 07:52.26 | penguinFunk | that dont have jitter/latency problems |
| 07:52.41 | McDouglas | nah, these people are running around in a storage facility |
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| 08:04.51 | McDouglas | hmm |
| 08:05.08 | McDouglas | if installed x-lite on my pda that would actualy simulate a wifi voip phone :P |
| 08:09.12 | peanut- | JT: yes, they are supposed to work out of the box, but you don't discount an entire line of devices because they don't work as expected, you make them work.. |
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| 08:11.18 | epaulin | How to dialing to a IAXTal without a asterisk server? |
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| 08:43.18 | mvanbaak | epaulin: look at their website. I think there are some access numbers |
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| 08:46.03 | epaulin | mvanbaak: access numbers? you mean IAXTal number like 234.567.6000? |
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| 08:51.58 | mvanbaak | yup |
| 08:52.28 | mvanbaak | I have a context that I use with the manager originate command. |
| 08:52.39 | mvanbaak | All my local phones are there, and also the routes to outbound |
| 08:52.53 | mvanbaak | now I want 1 local sip device to be routed out using a different iax trunk |
| 08:53.02 | mvanbaak | but I have no idea how to do that |
| 08:53.09 | mvanbaak | anyone any suggestions I can try ? |
| 08:53.20 | mvanbaak | I dont want to alter the script I use to connect to the manager |
| 08:53.32 | mvanbaak | so I'd like some dialplan magic to do this |
| 08:53.36 | mvanbaak | is it at all possible ? |
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| 08:58.43 | Draevyn | This might be a real newbie question, but if I were to upgrade the existing Asterisk 1.0.0 system here at work to the latest version, what potential issues could I have ? |
| 08:59.12 | Draevyn | I've only recently had to get to grips with extensions.conf... the rest was set up years before I joined this company. |
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| 09:08.47 | Uatec | evening |
| 09:10.31 | Draevyn | Mornin' :) |
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| 09:51.05 | ToTo | hai all |
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| 09:52.39 | Draevyn | This might be a real newbie question, but if I were to upgrade the existing Asterisk 1.0.0 system here at work to the latest version, what potential issues could I have ? |
| 09:52.43 | Draevyn | I've only recently had to get to grips with extensions.conf... the rest was set up years before I joined this company. |
| 09:53.25 | tzafrir_home | Draevyn, hmm... you should probably go over both http://svn.digium.com/svn/asterisk/branches/1.2/UPGRADE.txt and http://svn.digium.com/svn/asterisk/branches/1.4/UPGRADE.txt |
| 09:53.27 | stmaher | Man this room is quite without [TK]-Fender.. LOL |
| 09:53.55 | Draevyn | tzafrir : Thanks :) |
| 09:55.58 | bofh666 | Anyone with experience, connecting an Avaya Definity G3 v11 with Asterisk (TE410p)? We can route calls over our link, but it looks like the Avaya isn't sending number information (always: Accepting call from '' to '1235' on channel 0/1, span 4) |
| 09:58.20 | Uatec | hey |
| 09:58.31 | Uatec | i'm trying to use the asterisk manager interface to dial a number |
| 09:58.33 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 09:58.42 | Uatec | i've used telnet to dial manually |
| 09:58.52 | Uatec | but when i write my app to do it. i just get this message: Connect attempt from '192.168.232.75' unable to authenticate |
| 09:59.12 | Uatec | the text sent is exactly the same except maybe the newline characters |
| 09:59.28 | Uatec | i've tried \r\n \012\015 |
| 09:59.42 | Uatec | i'm using c# so i've also tried Environment.Newline |
| 09:59.52 | Uatec | but i always get that same message when i eventually disconnect |
| 10:00.11 | Uatec | i've connected to netcat, and netcat sees the newlines when i use \r\n and \012\015 |
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| 10:10.38 | Dr-Linux | hhm.. |
| 10:11.05 | Dr-Linux | there is a kinda AMI program with asterisk, i forgot the name, something like "adhrorten" or what |
| 10:11.11 | Dr-Linux | can someone remind me? |
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| 10:29.34 | MACscr | Is there a * code to get the current time of the system? |
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| 10:34.28 | *** join/#asterisk fbnts (n=root@89.145.97.160) |
| 10:35.29 | fbnts | hi. Just wondered if anyone had any experience with Cisco SIP phones? I have 3 which were all working fine until I moved Asterisk to an external internet host |
| 10:35.53 | fbnts | now when they try to register, asterisk logs a new registration every 2 seconds or so |
| 10:36.14 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
| 10:37.14 | fbnts | after doing TCP Dump on my local lan, it shows that the phone is sending register, Asterisk is then replying with "trying" then "ok" but the phone then sends an ICMP port unreachable |
| 10:37.19 | fbnts | any ideas? |
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| 10:56.46 | tzafrir_home | bofh666, if "it seems" then you may need a "lower level debugging. The equivalent of "sip debug" is "pri debug span NNN" |
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| 11:02.22 | bofh666 | tzafrir_home: been there, done that. Some of the debugging info: Q.931 / 3.1Khz Audio, 64Kbps circuit mode uLaw. IntID: Implicit PRI, Coding: 0 Number Specified Channel Type 3 Coding CCITT standard, Location: Private network, Progress Descr: Calling equipment is non-isdn TON: National number, NPI: ISDN/Telephony (E.164/E.163) |
| 11:03.27 | bofh666 | tzafrir_home: the 3 other spans are E1 lines to a local telco. The definity is connected using a 'cross cable'. |
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| 11:21.35 | tzafrir_home | bofh666, maybe pastebin hte trace and hope for the best... |
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| 11:34.24 | Dovid | hello ev1. anyone here know ss7 |
| 11:35.02 | Dovid | I am trying to understand SS7, how it works with asterisk etc. (I have been on google and i understand the concept of what it does etc.) but I do not have a clear picture |
| 11:36.05 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:36.41 | coppice | SS7? Why a four year old child counld understand it. Just find yourself a four year old child. |
| 11:38.36 | Dovid | Steve: I am new to it :( i am going through google and I don't fully get it. from what I understand it is a signalling system on the call, call route etc. correct ? |
| 11:42.01 | *** join/#asterisk HarryR`Work (n=harryr@77.240.56.17) |
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| 11:48.51 | coppice | does ISDN make sense to you? |
| 11:50.36 | Dovid | not at all |
| 11:50.41 | Dovid | no idea how it works |
| 11:51.10 | Dovid | all the docs online i have seen are basic in explaining it. |
| 11:52.19 | coppice | well, they are kinda like SIP, except SS7 and ISDN were throught out properly :-) |
| 11:53.07 | Dovid | ok. so like with SIP you have an invite message, over a PRI ss7 would handle such a message ? |
| 11:53.36 | tzafrir_home | Dovid, and then again there are several ss7 stacks at the moment for Asterisk |
| 11:53.47 | Dovid | tzafrir: may I pm ? |
| 11:53.59 | tzafrir_home | Dovid, not sure it would help |
| 11:54.41 | coppice | essentially, yes. there is a data network like IP, except with SS7 its called MTP. this lets nodes exchange messages about the calls they are handling on all the various audio paths. the messages they exchange are conceptually similar to SIP messages |
| 11:57.15 | Dovid | Staeve: Thanks. |
| 11:59.21 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 11:59.55 | Dovid | Steve: does sip work well with ss7. Meaning as of now we are only using SIP. we need to support ss7 for a specific client (who wants us to connect to them in another country) if my carrier has ss7 support will it work or if we want to exchange mssages with them I would need a PRI coming in to my asterisk box? |
| 12:02.07 | HarryR`Work | Dovid, you'd need an E1 or similar directly into your asterisk box |
| 12:02.42 | HarryR`Work | and a zaptel compatible card |
| 12:02.49 | Dovid | and i would have to send the voice traffic over the E1 correct (i cant say over ss7 hey the call will come over ip) |
| 12:03.16 | lirakis | morning |
| 12:03.28 | dan__t | 'morning. |
| 12:03.32 | Dovid | morning |
| 12:03.42 | HarryR`Work | yup, you're effectively bridging from your external IP/SIP provider to them over SS7 |
| 12:04.18 | dan__t | Ok, I do believe my Polycom nightmares are almost complete... |
| 12:04.38 | Dovid | HarryR: can u explain tha last one. i can send over ss7 that i want the call to go over SIP or are you saying that i can send the call to my carrier over sip and let their ss7 handle it? |
| 12:05.40 | HarryR`Work | no, I'm saying that you have to bridge/transcode the calls that come in from the ss7 channel, setup a call on your sip channel and send the voice/whatever over it |
| 12:05.44 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:05.52 | HarryR`Work | and the same in the other direction |
| 12:06.12 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 12:06.24 | HarryR`Work | otherwise why not just use sip if you could do that! |
| 12:07.08 | Dovid | HarryR: OK. So I would have Client <--ss7--> Carrier <--SIP--> Me and that will work ? |
| 12:07.39 | HarryR`Work | yes |
| 12:08.15 | Dovid | ok. thanks |
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| 12:15.26 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:15.58 | bofh666 | tzafrir_home: Debug output of a single call from Avaya to Asterisk: http://pastebin.ca/742307 |
| 12:18.20 | codefreeze | peanut-: saw your Q. last night-- ani is given priority when you store CDR(clid) |
| 12:21.02 | codefreeze | peanut-: also, the CDR(src) |
| 12:25.26 | dan__t | Ok, got the polycom phone to boot it's app and stuff via https |
| 12:25.29 | dan__t | that's pretty rad. |
| 12:29.38 | JT | peanut-: people discount whole lines of products because they are actually stuffed and faulty. stop making up insane excuses. |
| 12:30.36 | *** join/#asterisk bantu (n=Miranda@p54A32945.dip0.t-ipconnect.de) |
| 12:30.49 | [TK]D-Fender | JT : What'd I miss? :) |
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| 12:31.13 | dan__t | Alright, having trouble getting this Polycom phone to register with * this time. |
| 12:31.26 | dan__t | The Polycom phone is behind NAT, but would that matter? |
| 12:32.15 | JT | [TK]D-Fender: peanut saying that people on review websites shouldn't discount linksys wifi phones until they've installed openWRT or dd-WRT etc and fiddled with a thousand options |
| 12:33.10 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 12:33.16 | [TK]D-Fender | JT : I like lower prices for no reason whatsoever personally :p |
| 12:33.41 | JT | [TK]D-Fender: eh? |
| 12:34.10 | [TK]D-Fender | JT : Ah, OTHER definition of "discount" (dismiss, discredit, etc)? |
| 12:34.27 | JT | yes |
| 12:34.52 | [TK]D-Fender | JT : Clearer :) Yeah, I tend to discount all sorts of products because they're SHIT <- |
| 12:35.03 | [TK]D-Fender | ~wifisip |
| 12:35.04 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 12:35.04 | mohsen | I am writing an AGI script (in python) which is supposed to do many business logical things as well as billing. The problem is that when the *caller* hangs up, the channel hangs up too and the agi script can not continue. What's the solution? |
| 12:35.33 | JT | [TK]D-Fender: but you didn't try 10000 unsupported firmwares first??? |
| 12:35.36 | [TK]D-Fender | mohsen: "g" <----- |
| 12:36.21 | [TK]D-Fender | JT : There's a fine line between "desperately seeking validation" and "outright stupidity". |
| 12:36.26 | mohsen | [TK]D-Fender: "g" keeps the channel if the *callee* hangs up. But does not work for the caller hang up as far as I see. |
| 12:36.33 | JT | [TK]D-Fender: very true |
| 12:36.38 | [TK]D-Fender | mohsen: "h" <--- |
| 12:37.01 | bofh666 | JT / [TK]D-Fender: just spend a couple of hunderd euros on several Linksys WRT54g to create a nice mesh, just to find out a CellPhone with SIP client doesn't roam nicely. |
| 12:37.06 | mohsen | [TK]D-Fender: from the help of dial: h - Allow the called party to hang up by sending the '*' DTMF digit. |
| 12:37.08 | [TK]D-Fender | JT : Both categories are patheitcally delusional. |
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| 12:37.43 | mohsen | <PROTECTED> |
| 12:37.44 | JT | bofh666: you did this? |
| 12:37.48 | [TK]D-Fender | mohsen: I didn't say I was talking about a DIAL parameter, did I? :) Go re-read your Asterisk Standard Extensions. |
| 12:38.20 | bofh666 | JT / [TK]D-Fender: Cheap hardware isn't always a solution. I'm waiting for some test hardware from Netgear, to have SE P1i and Nokia N95 roam using SIP connections. |
| 12:38.50 | JT | wifi doesn't roam properly, especially with realtime applications |
| 12:39.00 | JT | netgear is still cheap hardware |
| 12:39.17 | bofh666 | Yep, but not as cheap as the WRT54g stuff. |
| 12:39.20 | mohsen | [TK]D-Fender: No, you did not say that and did not say that you are talking about "h" extension :), but that does not help again. I want to stay in my agi script and do the billing. I do not want to leave it to "h" extension |
| 12:39.37 | [TK]D-Fender | bofh666: Netgear is statistically crappier than most. |
| 12:40.04 | bofh666 | JT: We will be testing with Netgear WFS709TP and APs. Perhaps this will adjust the statistics ;-) |
| 12:40.18 | coppice | netgear is the same price as linksys, but is offered with extra problems free of charge |
| 12:40.28 | bofh666 | coppice: ;-) |
| 12:40.39 | [TK]D-Fender | mohsen: You start in an AGI in normal channel and then upon hangup it aborts your script? |
| 12:41.47 | [TK]D-Fender | bofh666: "failure is NOT an options.... it comes..bundled with the firmware" |
| 12:42.18 | coppice | its the one feature nobody *ever* disables in their firmware |
| 12:42.27 | bofh666 | [TK]D-Fender: will keep you posted. When it works better than our previous setup, I'll let you all know. |
| 12:42.43 | mohsen | [TK]D-Fender: well, not exactly like that. Upon caller hangup, it raises a python exception. If I catch the exception and continue I will get other exceptions when I try to read channel variables (e.g. ANSWEREDTIME) (probably because the channel is hang up.) |
| 12:43.23 | [TK]D-Fender | mohsen: Run your own timer and load all vars prior to dial where possible |
| 12:43.42 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:44.21 | mohsen | [TK]D-Fender: for some vars it might work. But for billing it does not work. You need answeredtime after the dial is finished. |
| 12:44.42 | JT | asterisk has a cdr module |
| 12:45.19 | [TK]D-Fender | mohsen: go dissect a2billing for some "inspiration" :) |
| 12:45.45 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 12:45.59 | mohsen | [TK]D-Fender: okay :) |
| 12:57.27 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 13:00.59 | mohsen | [TK]D-Fender: FYI: The main difference is that the AGI application may terminate if the line is hung up during execution and DeadAGI will not terminate even if the call is hung up during execution, however, the call leg will not automatically enter a "down" state until execution is completed if executed on a live line. As such, commands and applications designed to return the call state will... |
| 13:01.01 | mohsen | ...inaccurately return an "up" status |
| 13:01.49 | [TK]D-Fender | mohsen: Well then again, you KNOW if you pass your DIAL call that its ended... |
| 13:01.52 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 13:01.56 | zdrulio | hello all |
| 13:02.10 | zdrulio | i have a question about fax sending with asterisk |
| 13:02.33 | zdrulio | i have HP fax how is connected to asterisk via pap2t ATA device |
| 13:02.34 | *** join/#asterisk dez71 (i=dez@216.83.0.172) |
| 13:02.37 | dan__t | Under what situations would a SIP proxy be necessary? |
| 13:03.02 | zdrulio | can i send a fax with this scenario |
| 13:03.37 | mohsen | [TK]D-Fender: yes, using deadagi asterisk does not close the pipe when the call hangs up. |
| 13:03.55 | mohsen | so I can continue read the channel vars and do the stuff :D |
| 13:05.53 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 13:05.56 | [TK]D-Fender | dan__t: large scale / redundant installations |
| 13:06.40 | dan__t | LIke, what, same way a traditional proxy would be used? |
| 13:07.10 | [TK]D-Fender | dan__t: what is "traditional" implying? |
| 13:07.21 | dan__t | Say an http proxy, whatever. |
| 13:07.41 | zdrulio | anyone have a experience with fax over asterisk ? |
| 13:08.04 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 13:08.05 | keith4 | zdrulio: just bad experiences |
| 13:08.11 | jstew | greetings. |
| 13:08.13 | dan__t | I think NAT is eating me alive. |
| 13:08.25 | Katty | i ... |
| 13:08.26 | Katty | just.. |
| 13:08.30 | Katty | just. |
| 13:08.39 | Katty | http://birdloversonly.blogspot.com/2007/09/may-i-have-this-dance.html <- just. go look. i can't stop giggling. |
| 13:08.45 | Katty | mishehu: ^- go see bird. |
| 13:08.56 | [TK]D-Fender | dan__t: Gee you might think that a SIP proxy is just like an HTTP proxy but for a different protocol or something! |
| 13:09.03 | jstew | Is the asterisk version that's in the ubuntu 7.10 repos pretty decent? I remember trying to use it on 6.10 and it sucked so I had to compile it myself. |
| 13:09.23 | dan__t | No, actually, I wouldn't think that. Actually. But thanks. |
| 13:09.47 | JT | jstew: generally it's best just to compile |
| 13:10.01 | jstew | yeah, that's what I figured :| |
| 13:10.06 | dan__t | Should I wait for some more 12 year old responses? |
| 13:10.15 | zdrulio | keith4: and waht happens ? are u send a fax over asterisk ? |
| 13:10.16 | JT | dan__t: maybe you should stop being so emo |
| 13:10.20 | dan__t | Give me a break. It was a simple question that deserved a simple answer. |
| 13:10.24 | dan__t | That hurts, JT :< |
| 13:10.29 | jstew | I'm trying to think of things 3 years down the road when I might not be here... lol |
| 13:10.40 | dan__t | My wrists are perfectly fine, thanks. |
| 13:10.40 | [TK]D-Fender | Katty: Mew. |
| 13:10.49 | [TK]D-Fender | Katty: Damn that was cute... its a saver... |
| 13:11.08 | JT | dan__t: you use a sip proxy, a comparison to say, running squid or a load balancer as a front end to apache |
| 13:11.20 | Katty | [TK]D-Fender: ^_^ |
| 13:11.23 | JT | dan__t: you don't need to slash wrists to be emotional ;) |
| 13:11.30 | dan__t | I did not want to wrongly assume, JT. |
| 13:11.35 | Katty | [TK]D-Fender: i wonder what ryan would say to getting a bird :> |
| 13:11.43 | dan__t | But thank you. |
| 13:11.52 | JT | dan__t: almost exclusively used from a server standpoint |
| 13:12.17 | JT | sip proxies can be used for a number of reasons |
| 13:12.48 | dan__t | Ok. |
| 13:12.59 | [TK]D-Fender | ~assume |
| 13:12.59 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 13:13.21 | JT | load balance, security, simplicity |
| 13:13.29 | dan__t | Yeah. |
| 13:13.45 | dan__t | I know what a proxy is, just didn't know if it was used in the same context as any other proxy, when applying it to SIP. |
| 13:13.57 | [TK]D-Fender | dan__t: Here, ready up : |
| 13:13.59 | [TK]D-Fender | ~sipnat |
| 13:14.00 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:14.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 13:14.19 | dan__t | Yeah, hey, it's almost my homepage. |
| 13:15.20 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.15 | dan__t | brb |
| 13:19.09 | defswork | Could anuyone recommend a good online resource for understand dial plans ? I've been using freepbx but need to add some customer stuff (simple stuff I hope) and need some pointers |
| 13:23.39 | *** join/#asterisk bl4q (n=Bl@dslb-088-066-247-078.pools.arcor-ip.net) |
| 13:25.41 | keith4 | ~book |
| 13:25.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 13:25.48 | keith4 | defswork: ^^^^^ |
| 13:25.54 | defswork | keith4: I have the book - it's at home :) |
| 13:26.18 | keith4 | lucky for you that they have a free PDF of it, then |
| 13:26.45 | [TK]D-Fender | defswork: ....LOL |
| 13:26.53 | [TK]D-Fender | defswork: Good luck with "simple" |
| 13:27.28 | defswork | [TK]D-Fender: shouldn't be too bad - just want to block some extensions from dialling out |
| 13:27.48 | defswork | outbound* |
| 13:28.01 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 13:28.11 | [TK]D-Fender | defswork: ..... There should be a GUI option for that already and if there isn't.... LOL you are so up a creek... |
| 13:28.43 | defswork | [TK]D-Fender: there isn't |
| 13:28.57 | defswork | [TK]D-Fender: you seem to have a low estimation of my abilities :( |
| 13:29.03 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:29.49 | defswork | [TK]D-Fender: Unless I'm missing something I was just going to add some conditions to outbound-allroute-custom |
| 13:30.23 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 13:30.42 | Bladerunner05 | hello boy, there is a software (other than festival) to let asterisk play voice? |
| 13:32.19 | [TK]D-Fender | defswork: Sure you can probably mod your configs... and the second they get REBUILT by FreePBX your changes get ANNIHILATED <--- |
| 13:32.41 | [TK]D-Fender | Bladerunner05: Cepstral |
| 13:33.04 | defswork | [TK]D-Fender: nah - thats why it has _custom contexts |
| 13:33.09 | [TK]D-Fender | defswork: And yes, running FreePBX alone is enough for me to shove you in a whole new category. |
| 13:33.11 | Bladerunner05 | thanks <[TK]D-Fender> I chheck |
| 13:33.17 | [TK]D-Fender | defswork: but... |
| 13:33.19 | [TK]D-Fender | ~wglwat |
| 13:33.20 | jbot | extra, extra, read all about it, wglwat is well, good luck with all that |
| 13:34.21 | Bladerunner05 | <[TK]D-Fender> does Cepstral interact with asterisk? |
| 13:34.36 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 13:34.38 | Katty | [TK]D-Fender: i've watched the birdie dance 30 times now. |
| 13:34.43 | Sci_05 | morning all |
| 13:35.09 | *** join/#asterisk mirco (n=mirco@p54B249DC.dip.t-dialin.net) |
| 13:36.19 | *** join/#asterisk blq (n=Bl@dslb-088-066-247-078.pools.arcor-ip.net) |
| 13:36.25 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 13:37.23 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-a96769e5bb9a9ef6) |
| 13:38.12 | *** join/#asterisk captiancrash (n=jmoore@70.159.118.70) |
| 13:39.26 | *** part/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 13:39.50 | *** join/#asterisk jfitzgibbon-away (n=chatzill@64.72.237.187) |
| 13:46.39 | defswork | wglwat ? |
| 13:47.18 | JT | defswork: read the line under that. |
| 13:48.18 | *** join/#asterisk dreamind (n=dreamind@p54A78877.dip0.t-ipconnect.de) |
| 13:48.20 | JT | defswork: then try realism. |
| 13:48.22 | dreamind | Hi folks :) |
| 13:48.47 | Katty | [TK]D-Fender: help, i can't stop watching snowball dance |
| 13:49.06 | [TK]D-Fender | JT : Realism is for people who can't handle drugs :D |
| 13:49.06 | *** join/#asterisk gardo (n=gardo@121.97.178.82) |
| 13:49.08 | dreamind | I've some problems with asterisk and app_fax (asterisk 1.4.13 and the latest app_fax from debian sid) - I can receive faxes with rxfax, but depending on the sending fax it doesn't work :( |
| 13:49.22 | dreamind | if I send with capisuite from another linux pc, everything works fine. |
| 13:49.40 | dreamind | but if I send from a (IMHO G3 fax) here, it fails. |
| 13:50.02 | dreamind | I now disabled ECM manually, by hacking the source, but now it seems the resolution is not send correctly :( |
| 13:50.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:50.40 | JT | d ,, q,,dffgmfmllp;lmkplllkaassaaaaaq f,gyujk.nklfoinklklipnkl;;nkl,./ |
| 13:50.46 | JT | grr |
| 13:51.21 | dreamind | huh, defect keyboard? |
| 13:51.49 | bofh666 | he fell asleap I guess? |
| 13:51.56 | bofh666 | ^QA a/e |
| 13:52.08 | JT | problems with XOFF |
| 13:52.20 | dreamind | oh I didn't thought of that - its afternoon here ;) |
| 13:52.45 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 13:54.02 | bofh666 | Over here: 2007-10-19T13:53:31.00Z at 53.199117 N 5.785016 E |
| 13:57.40 | stimpie | someone knows how to use multiple outboundproxies in asterisk? |
| 13:58.02 | JT | DNS SRV |
| 13:59.27 | stimpie | hmm thought about that one problems is that we dont run dns |
| 14:01.57 | Dr-Linux | hi guys |
| 14:02.21 | Dr-Linux | i'm looking for the phone that work for voip and for RJ11 as well |
| 14:02.33 | Dr-Linux | any recommened phone? |
| 14:02.59 | [TK]D-Fender | Dr-Linux: SPA-3102 |
| 14:03.13 | Dr-Linux | [TK]D-Fender: that will work for both? |
| 14:03.32 | [TK]D-Fender | Dr-Linux: Did you think I only read HALF of your question? |
| 14:03.35 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 14:03.44 | JT | stimpie: then run it? |
| 14:04.05 | Dr-Linux | [TK]D-Fender: hehe i know you are kinda serious guy... but very nice :) |
| 14:04.53 | stimpie | JT, its again something extra to maintain |
| 14:05.15 | JT | stimpie: then why are you maintaining multiple sip proxies? |
| 14:05.28 | stimpie | reliability |
| 14:05.43 | Dr-Linux | [TK]D-Fender: is it a sip device or phone? where is it's headset :S |
| 14:05.48 | JT | and you can't set up a simple DNS server.... |
| 14:05.55 | JT | Dr-Linux: it's an ATA |
| 14:06.01 | [TK]D-Fender | Dr-Linux: Use your imagination.... |
| 14:06.12 | Dr-Linux | yeah, it's an ATA |
| 14:06.14 | [TK]D-Fender | Dr-Linux: or at least GOOGLE <--- |
| 14:06.30 | defswork | Dr-Linux: it's a little box with 2 analog ports and ethernet |
| 14:06.34 | Dr-Linux | well, i'm already using a number of ATA's Fender had suggested , working good |
| 14:06.39 | Dr-Linux | SPA 2000 |
| 14:06.57 | stimpie | JT, I can but I try too keep things minimal |
| 14:07.19 | Dr-Linux | but i want some phones that i can connect with ATA's RJ11 ports |
| 14:07.23 | JT | stimpie: ok, you're not making much logical sense, rethink your approach... |
| 14:07.28 | Dr-Linux | also if the same phone is voip phone that good |
| 14:07.44 | stimpie | JT, thats what iam doing right now ;-) |
| 14:07.48 | [TK]D-Fender | Dr-Linux: Go download a data sheet on this unit. |
| 14:08.28 | sepen | [TK]D-Fender, after compile the lastest wanpipe drivers, my sangoma 101/2 ATF card is working perfectly, 2many thanks!! |
| 14:08.41 | JT | Dr-Linux: you mean like a normal analogue phone? |
| 14:08.55 | [TK]D-Fender | sepen: Glad to hear... that was damn odd that HWPROBE would see it but not init... |
| 14:09.06 | [TK]D-Fender | sepen: What Wanpipe were you using that failed? |
| 14:09.12 | Dr-Linux | JT: that's correct, currently we are using plantronic phones |
| 14:09.57 | Dr-Linux | JT: basically i've 5 MultiTech gateways which i configured with Asterisk .. super voice quality, so i need something like plantronic, |
| 14:10.00 | sepen | 2.3 and now 3.2.1 |
| 14:10.24 | Dr-Linux | since we bought thoese plantronic 3 years ago .. so i thought maybe you guys could give me better suggestion |
| 14:10.36 | JT | Dr-Linux: how about... an IP Phone? |
| 14:12.26 | Dr-Linux | JT: we have about 30 cisco 7930 that's working in our different office for internal PBX |
| 14:12.47 | Dr-Linux | but for our call centers we have integrated MultiTech voip gatways |
| 14:13.13 | JT | why not get ip phones |
| 14:13.13 | Dr-Linux | we wanna utillize our MT gateways since they have good voice quality |
| 14:13.18 | Dr-Linux | we had them before asterisk |
| 14:13.22 | JT | ip phones have much better quality |
| 14:13.51 | Dr-Linux | s/7930/7960 |
| 14:13.53 | [TK]D-Fender | 7930? |
| 14:13.56 | [TK]D-Fender | ah |
| 14:13.58 | [TK]D-Fender | better |
| 14:14.33 | defswork | [TK]D-Fender: despite my uslessness I did it ;) |
| 14:14.35 | Dr-Linux | typo |
| 14:15.05 | defswork | [TK]D-Fender: I'm really useless - just very new to asterisk etc.. and not got time with real job to learn it as quick as I would like |
| 14:15.13 | defswork | not really* :| |
| 14:15.14 | Dr-Linux | JT: yes i'd agree but our servers located in US datacenters, but our callcenters/sip users are in Pakistani |
| 14:15.17 | *** join/#asterisk blq (n=Bl@dslb-088-067-025-131.pools.arcor-ip.net) |
| 14:15.20 | Dr-Linux | we have bad internet here |
| 14:15.25 | Dr-Linux | so interenet is a problem |
| 14:15.36 | JT | Dr-Linux: but you're using voip anyway... |
| 14:15.38 | defswork | Dr-Linux: I only the staff here access to the good internet |
| 14:16.18 | Dr-Linux | defswork: where? |
| 14:16.26 | defswork | Dr-Linux: in the office |
| 14:16.33 | defswork | Dr-Linux: the bad internet would corrupt them |
| 14:16.34 | Dr-Linux | where is your office? |
| 14:16.49 | defswork | Dr-Linux: Birmingham, England |
| 14:17.13 | Dr-Linux | man you guys have good and stable internet |
| 14:17.27 | defswork | we have both bad and good internets |
| 14:17.37 | Dr-Linux | our best internet over here, often goes down .. i.e. 5 to 10 times daily |
| 14:17.43 | Dr-Linux | and never stable |
| 14:17.52 | defswork | We don't have any horses |
| 14:18.05 | defswork | so not need for the stable internets |
| 14:18.17 | Dr-Linux | you don't have callcenter |
| 14:18.28 | defswork | Dr-Linux: No - we moved them all to india |
| 14:18.45 | Dr-Linux | actually we can't afford downtime |
| 14:19.00 | defswork | Our downtime is free - we have to pay to be working |
| 14:19.31 | Dr-Linux | i think cisco 7960 is the best voip phone and we have many, but we liked MultiTech gateways bcoz of bandwidth |
| 14:19.43 | Dr-Linux | [TK]D-Fender: how's your experience with MT gateways? |
| 14:19.53 | Dr-Linux | i guess you were playing with them? |
| 14:26.00 | JT | cisco do not make that good voip phones |
| 14:27.43 | *** join/#asterisk ManxPower (n=manxpowe@126.sub-75-200-16.myvzw.com) |
| 14:27.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:27.49 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:30.21 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 14:30.45 | [TK]D-Fender | Dr-Linux: Touched one ONCE. I DESPISE them. my head office has one and they are using an odd frame size and the unit is DUMB |
| 14:31.46 | dhpeterson_ | i've just spent most of today putting cisco 7940's and 60's onto our internal asterisk |
| 14:31.56 | dhpeterson_ | found it pretty straightforward |
| 14:32.12 | dhpeterson_ | except for that stupid default where it looks to the DHCP server as the default TFTP server |
| 14:32.13 | dhpeterson_ | yeah right |
| 14:32.36 | dhpeterson_ | oh and for some reason i can't get it to read RINGLIST.DAT :) |
| 14:32.47 | Nugget | Welcome to the world of Cisco phones. :) |
| 14:32.52 | dhpeterson_ | but for the most part - dialing, hold, conf, transfer, all worked with asterisk out of the box |
| 14:32.59 | dhpeterson_ | :) |
| 14:33.09 | dhpeterson_ | i have also used polycom ip500 and found them pretty good also |
| 14:33.17 | dhpeterson_ | they all seem to have their little config vagaries tho :) |
| 14:33.36 | dhpeterson_ | anyway #chan there's my $0.02 ;P |
| 14:33.40 | dhpeterson_ | heh |
| 14:33.44 | dhpeterson_ | flame away :) |
| 14:34.35 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 14:35.11 | ThatKidKel | Does anyone have any suggestions for an IP Phone behind a firewall? Something to keep the phone's port int he firewall open during idle times? |
| 14:35.20 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:35.36 | syzygyBSD | ThatKidKel: just noop it every 30 seconds |
| 14:35.36 | keith4 | ThatKidKel: NAT? |
| 14:35.45 | keith4 | or an actual firewall? |
| 14:35.50 | ThatKidKel | keith4.. both |
| 14:36.19 | ThatKidKel | we use nat all over the place, but one of our employees has went to another location, and whiel he can initiate a call and it works, after about 2 minutes, if a call goes to him the invite is dropped at the firewall |
| 14:36.26 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 14:36.33 | ThatKidKel | syzygyBSD.. can you furhter elaborate? |
| 14:36.35 | mocker | Dr-Linux: How hard was it to get BLF info on the 7960? |
| 14:36.47 | keith4 | so make a firewall hole |
| 14:36.58 | mocker | I have a bunch of 7941s and I'm wondering if they can do presence. |
| 14:36.59 | ThatKidKel | keith4.. we don't have control over the remote site's firewall |
| 14:37.01 | mocker | er, blf |
| 14:37.29 | mocker | The voip-info page looks... daunting. |
| 14:37.37 | [TK]D-Fender | ThatKidKel: "qualify=yes" for its sip.conf entry |
| 14:38.00 | [TK]D-Fender | mocker: Cisco does not support presence in SIP |
| 14:38.08 | [TK]D-Fender | mocker: only SCCP |
| 14:38.14 | keith4 | ew |
| 14:38.24 | ThatKidKel | [TK]D-Fender.. I've found with that, it will go to UNREACHABLE status after about 1 minute or so |
| 14:38.26 | mocker | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+Presence+for+Cisco+79x1+Phones |
| 14:38.37 | [TK]D-Fender | ThatKidKel: here : |
| 14:38.38 | [TK]D-Fender | ~sipnat |
| 14:38.39 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 14:38.39 | Dr-Linux | mocker: BLF info? |
| 14:38.39 | mocker | It looks like it's possible, just verry hacky. |
| 14:39.00 | *** join/#asterisk huey23 (n=huey23@64.192.209.34) |
| 14:39.10 | mocker | Things I won't do: "The Asterisk code needs to be patched to send the NOTIFY in the correct format." |
| 14:39.14 | [TK]D-Fender | mocker: Oh God.... NO COMMENT |
| 14:39.37 | [TK]D-Fender | Cisco = mistake |
| 14:39.48 | [TK]D-Fender | Maybe just *1* comment :p |
| 14:40.01 | mocker | [TK]D-Fender: I recommended Polycom. ;) |
| 14:40.37 | dhpeterson_ | ThatKidKel: this is a real problem - really with the SIP protocol |
| 14:40.58 | huey23 | [TK]D-Fender: can you look at something for me please? |
| 14:41.15 | JT | dhpeterson_: not so, it's a problem with misconfiguration |
| 14:41.32 | dhpeterson_ | ok |
| 14:41.41 | dhpeterson_ | is it a single phone or set of phones behind the fw? |
| 14:41.44 | *** join/#asterisk guyzmo (n=guyzmo@nenya.mithrandir.net) |
| 14:42.09 | huey23 | [TK]D-Fender: http://pastebin.com/m528507a0 |
| 14:42.51 | huey23 | [TK]D-Fender: that is one phone, i want to change the sntp config for all phones at once...and something is overriding my settings |
| 14:43.49 | [TK]D-Fender | huey23: Well thats clearly a PHONE LEVEL override and that takes precedence over the master. Told you your approach to this was wrong from the beginning... |
| 14:44.01 | huey23 | :P |
| 14:44.17 | huey23 | [TK]D-Fender: i have a hard time understanding sometimes...it's not my listening skills :) |
| 14:44.20 | guyzmo | hi, I got a voip phone (the e65) and a sip account with my internet provider, and my phone can't register to the service given my lan configuration (and I can't figure why), would it be possible and useful to use asterisk as a sip proxy ? if yes, can anyone guide me on good docs to rtfm ? :) |
| 14:44.54 | [TK]D-Fender | guyzmo: |
| 14:44.55 | syzygyBSD | ~book |
| 14:44.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:44.59 | [TK]D-Fender | ~b2bua |
| 14:44.59 | jbot | from memory, b2bua is a back 2 back user agent |
| 14:45.05 | [TK]D-Fender | Asterisk --^^^ |
| 14:45.13 | huey23 | [TK]D-Fender: therefore, if change the settings in the phone, it should work |
| 14:45.44 | [TK]D-Fender | huey23: Depending on a predictable set of variables.... |
| 14:45.45 | guyzmo | ok |
| 14:46.06 | guyzmo | so I take it as a yes :) |
| 14:46.07 | guyzmo | nice then |
| 14:46.22 | huey23 | [TK]D-Fender: from the pastebin...would you be able to point me in the right direction in to what's overriding my hard settings in the phone? |
| 14:46.39 | [TK]D-Fender | huey23: that IS your hard setting on the phone. |
| 14:47.31 | ThatKidKel | [TK]D-Fender.. So In that Asterisk NAT Solutions, I would be #9. Asterisk on the public internet, and the client behind a firewall/NAT. I set NAT=yes, qualify=100 .. after a few sceonds, it went to UNREACHABLE.. now calls can't go through to it |
| 14:47.37 | [TK]D-Fender | huey23: Stop bastardizing your configs and trying to provision them only to override on the phone itself. |
| 14:47.47 | [TK]D-Fender | ThatKidKel: the FIRST link.... |
| 14:48.05 | [TK]D-Fender | ThatKidKel: Qualify=100? ICK. "yes" <------------ |
| 14:48.21 | [TK]D-Fender | ThatKidKel: yes = 2000 (if you know whats good for you) |
| 14:48.58 | _x86_ | pop quiz... from the smart jack to the CSU... RJ48C or RJ48S? |
| 14:49.16 | ManxPower | qualify=100 would screw up just about any system |
| 14:49.44 | huey23 | [TK]D-Fender: the phone is not DHCP, the network settings say 192.168.2.20, i input 192.168.2.21 and the phone remains 192.168.2.20 |
| 14:49.48 | ManxPower | ThatKidKel: also, to make sure it is not a case issue use nat=yes and not NAT=yes |
| 14:49.54 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:49.55 | stimpie | I have an outboundproxy defined but message still to go the domain used in the dial command |
| 14:49.56 | [TK]D-Fender | _x86_: "X" |
| 14:50.30 | Katty | _x86_: rj48s is for data |
| 14:50.48 | ManxPower | smartjack to csu/dsu would use a straight thru cable. A standard ethernet cable would work |
| 14:50.56 | ManxPower | stimpie: paste the Dial command |
| 14:51.21 | stimpie | exten => _X!,7,Dial(SIP/${EXTEN}@${SIPDOMAIN}) |
| 14:51.41 | ManxPower | stimpie: don't do that. Dial(SIP/${EXTEN}@sipconfentry) |
| 14:51.53 | [TK]D-Fender | huey23: Input it where? |
| 14:52.10 | Katty | _x86_: and i know that 48c is usually for 1.54MBPS, and 48s is for local stuffs... |
| 14:52.11 | huey23 | on the phone |
| 14:52.19 | ManxPower | [TK]D-Fender: I think huey23 is trying to con you into teaching him how to configure his phone. |
| 14:52.40 | _x86_ | Katty: 48c and 48s are the same thing, one is straight-through, the other is crossed |
| 14:52.46 | _x86_ | Katty: RJ45 is used for LAN |
| 14:52.54 | huey23 | [TK]D-Fender: i know how to configure a phone...i just don't know what's overriding my settings when i input the settings in my phonew |
| 14:53.16 | _x86_ | [TK]D-Fender: thanks |
| 14:53.28 | stimpie | ManxPower, ${SIPDOMAIN} could be anything so I cant create an entry for it |
| 14:54.06 | Katty | _x86_: so i'm going to say RJ48s goes from the smart jack OSU card and the RJ48c goes from the wall-mount thing to the CSU/DSU |
| 14:54.40 | [TK]D-Fender | huey23: I asked you WHERE, don't jsut say "the phone". what MENU? |
| 14:54.51 | Katty | _x86_: of course, i really have no idea. |
| 14:55.05 | huey23 | [TK]D-Fender: the network configuration menu on the phone |
| 14:55.12 | [TK]D-Fender | huey23: And yes.. you know how to configure a phone... thats why its working so well.... |
| 14:55.46 | huey23 | [TK]D-Fender: i can take that...but i didn't set these phones up...that's why i am asking for insight |
| 14:56.01 | [TK]D-Fender | huey23: </contradictions> |
| 14:56.15 | [TK]D-Fender | huey23: try setting your network params in the BootROM. |
| 14:56.32 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 14:56.56 | huey23 | [TK]D-Fender: if i change the config through the mac.cfg files it works fine...if i change it on the phones through the net config menu, it doesn't...i'll check the booROM file but could it be a permissions issue? |
| 14:57.21 | [TK]D-Fender | huey23: I didn't say FILE anywhere... I said in the BOOTROM. |
| 14:57.44 | [TK]D-Fender | huey23: not in the Application config menu, on the BOOTROM itself |
| 14:57.46 | _x86_ | [TK]D-Fender: i need to make a new cable to go from the smart jack to my CSU... any ideas on where i can find information on how to do the pin-out? |
| 14:58.09 | ManxPower | _x86_: I guess you were not listening. |
| 14:58.42 | ManxPower | stimpie: then you really can't do much to control the call if you are dialing by domain |
| 14:59.01 | [TK]D-Fender | _x86_: http://www.google.ca/search?hl=en&q=rj48X+pin-out&btnG=Search&meta= |
| 14:59.03 | ManxPower | for one thing you would need srvlookups enabled since you are dialing by domain rather than hostname. |
| 14:59.11 | _x86_ | ManxPower: what did i miss? |
| 14:59.32 | ManxPower | _x86_: the fact that most any ethernet cable will for as a straight thru T-1 cable. |
| 14:59.45 | [TK]D-Fender | _x86_: RJ48 = RJ45 crimp head with specific PIN-OUT's |
| 15:00.05 | [TK]D-Fender | _x86_: For which you can use a standard CAT5 STRAIGHT CABLE |
| 15:00.10 | _x86_ | [TK]D-Fender: yes, i know... hence why i was looking for the PIN OUTS |
| 15:00.20 | _x86_ | hmm... negative |
| 15:00.24 | [TK]D-Fender | _x86_: And I've just gone and linked you |
| 15:00.36 | ManxPower | _x86_: you don't have an ethernet cable to look at the pinouts? |
| 15:00.37 | [TK]D-Fender | ~[TK]D-Fender |
| 15:00.38 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 15:00.47 | _x86_ | i tried a standard straight-through LAN cable and it did not get a link (although the cable tested fine, and straight-through) |
| 15:00.59 | ManxPower | Oooohhhh! a Google proxy. I wonder if he has any open ports. |
| 15:01.16 | [TK]D-Fender | ManxPower: EXIT ONLY <- |
| 15:01.19 | _x86_ | haha |
| 15:01.24 | ManxPower | _x86_: then you have a problem other than a cable issue. |
| 15:02.53 | tzafrir_home | ManxPower, everything is being logged in the proxy as well |
| 15:03.06 | tzafrir_home | Every single request |
| 15:03.13 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 15:03.36 | keith4 | _x86_: http://en.wikipedia.org/wiki/RJ48 ? |
| 15:07.34 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 15:07.37 | roxlu | hi! |
| 15:09.03 | roxlu | I just got me a voip account with a provider.... I've done atest with "Talkin 2 Ya" and it works. I've installed Asterisk on my server, but what would be my next step if I want to start using asterisk? |
| 15:09.32 | _x86_ | ah, aparantly, RJ48X has a "shorting bar" |
| 15:09.45 | _x86_ | no wonder standard RJ45 doesn't work |
| 15:09.53 | ManxPower | _x86_: you are an idiot. |
| 15:10.18 | _x86_ | ManxPower: http://en.wikipedia.org/wiki/Registered_jack |
| 15:10.28 | ManxPower | I have at LEAST 10 T-1s with standard ethernet cables between the smartjack and csu/dsu |
| 15:10.39 | ManxPower | _x86_: Yes, and the shorting bar is for creating loopbacks. |
| 15:10.41 | *** join/#asterisk jstew (n=jstewart@fw-ext.fusionary.com) |
| 15:10.46 | ManxPower | you don't need it. |
| 15:11.27 | _x86_ | hmm |
| 15:11.41 | jstew | Hey, I wanted to install trixbox but centos does not support my mobo chipset, so I'm using ubuntu. What software can I install to get something equivalent to trixbox? |
| 15:11.44 | ManxPower | But if you want to waste time and money special ordering an RJ48X, more power to you. |
| 15:11.57 | jstew | I'm putting freePBX on. What else am I missing? |
| 15:11.58 | _x86_ | I took the cable that came with a rhino channel bank, took it from the smart jack to the CSU on my asterisk box, and it works |
| 15:12.06 | ManxPower | jstew: we don't know, as we don't use it or support it here. |
| 15:12.28 | _x86_ | took a packaged CAT6 cable (straight-through) from the smart jack to the CSU, and it wouldn't even sync |
| 15:12.32 | jstew | Alright.. I guess I'll just roll my own then |
| 15:12.59 | ManxPower | _x86_: does the cable work as an ethernet cable? |
| 15:13.13 | _x86_ | haven't tried that... |
| 15:13.15 | *** join/#asterisk Arc_Ressiv (i=RHeart@67.108.111.146.ptr.us.xo.net) |
| 15:13.18 | ajohnson | heh |
| 15:13.19 | ManxPower | maybe you should. |
| 15:13.35 | ajohnson | I can't tell you how many of those cables I have on my T1 circuits |
| 15:13.36 | ManxPower | and does the "ethernet cable" have all 4 pairs connected? |
| 15:14.03 | _x86_ | the only problem with the current cable, is that the boot is too big and I think it pushes against the aluminum on the case and causes the link to intermittenly bounce |
| 15:14.21 | _x86_ | ManxPower: yeah it has to... to meet cat6 standards |
| 15:14.46 | _x86_ | also, zttool is telling me the clock source on the T1 is "internal" |
| 15:14.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 15:14.51 | ManxPower | _x86_: cat 6 is a cable spec, not a jack/plug spec |
| 15:15.02 | ManxPower | _x86_: zttool lies about that. |
| 15:15.22 | ManxPower | I ask again, does that cable work as a standard ethernet cable? |
| 15:15.46 | huey23 | [TK]D-Fender: i can't figure it out what you mean by saying "look in the bootROM"...i will just have to change all the mac.cfg files |
| 15:15.48 | _x86_ | i tell you again, i haven't tried... i'm 50+ miles away from it ;) |
| 15:15.49 | *** join/#asterisk bhrobinson (n=Flagg732@63.133.153.98) |
| 15:16.14 | ManxPower | _x86_: seems kind of silly yo try to diagnose a hardware issue from 50+ miles away, doesn't it. |
| 15:16.16 | _x86_ | how can i verify the T1 is recieving timing from the LEC, and not trying to provide timing? |
| 15:16.30 | _x86_ | ManxPower: not sure it's a hardware issue at this point |
| 15:16.46 | ManxPower | huey23: "bootrom". Turn on phone, at the setup, start, whatever menu press setup BEFORE it says "loading application" |
| 15:17.32 | huey23 | ManxPower: i'll give it a shot but I believe i have allready tried it...thanks |
| 15:17.57 | ManxPower | huey23: those options cannot be set in the web interface and cannot be set in the config files. |
| 15:18.30 | ManxPower | it is pointless to have things like vlan settings, config file download settings and IP config settings be saved in a config file. |
| 15:18.51 | [TK]D-Fender | huey23: Mean reboot your darn phon and goi into SETUP <- |
| 15:19.07 | huey23 | ManxPower: i see what you're saying but why would they do that if the "startup setup" is different than changing the settings in the menu? |
| 15:19.27 | ManxPower | huey23: what is "startup setup"? |
| 15:19.37 | huey23 | that's what you just told me to do |
| 15:19.48 | ManxPower | no, the bootrom setup. |
| 15:20.00 | [TK]D-Fender | huey23: BEFORE SIP loads |
| 15:20.15 | ManxPower | [TK]D-Fender: huey23 seems to be channeling _x86_ |
| 15:20.56 | huey23 | [TK]D-Fender: i got it...i've done it...and something is overriding it |
| 15:21.16 | _x86_ | ManxPower: the problem is the T1 is randomly bouncing |
| 15:22.08 | _x86_ | zaptel.conf shows the span is setup to receive timing, I've replaced the CSU, I've replaced the cable from the smart jack to the CSU, and the problem persists |
| 15:22.11 | huey23 | [TK]D-Fender: just completed the setup before sip loads...and still the same issue |
| 15:22.22 | ManxPower | huey23: I think you are living in an alternate universe where the laws of physics are different. |
| 15:22.31 | _x86_ | LEC is persistant that the problem is not on their end, and every time I call them, they run patterns to the smart jack clean |
| 15:22.43 | ManxPower | _x86_: what brand of CSU are you using? |
| 15:22.46 | _x86_ | Sangoma |
| 15:22.50 | _x86_ | both times |
| 15:22.53 | _x86_ | A102D-x |
| 15:22.57 | ManxPower | I didn't know Sandoma sold stand alone CSUs |
| 15:23.10 | ManxPower | _x86_: maybe you could start saying "sangoma" instead of CSU. |
| 15:23.26 | huey23 | ManxPower: maybe...but something is still overriding the phone :P |
| 15:23.33 | roxlu | does someone knows an application like "Talkin 2 Ya" for the Mac? |
| 15:23.41 | *** part/#asterisk bhrobinson (n=Flagg732@63.133.153.98) |
| 15:23.45 | _x86_ | dude, in the telco world, the interface that connects to a smart jack is called the CSU... regardless of internal/external |
| 15:23.46 | [TK]D-Fender | huey23: Why aren't you using DHCP? |
| 15:23.49 | ManxPower | huey23: personally I think you are not saving the config |
| 15:24.10 | ManxPower | _x86_: correct. but most people here are not from the telcom world. |
| 15:24.14 | *** join/#asterisk eldon (i=eldon@nat/digium/x-2ad05f7ca0cccd00) |
| 15:24.27 | _x86_ | ...ok |
| 15:24.47 | huey23 | [TK]D-Fender: i just work with it...i did't set it up...besides, what are the benefits? |
| 15:24.53 | _x86_ | i've replaced the sangoma card that magically connects to the smart jack without an external CSU |
| 15:24.56 | _x86_ | how's that? :) |
| 15:24.59 | huey23 | ManxPower: i saved the config |
| 15:25.52 | [TK]D-Fender | huey23: You seem rather clueless about your phone. Perhaps you should try and get some training materials, or hire a consultant. |
| 15:26.34 | [TK]D-Fender | huey23: Go call up Polycom and find a tech in your area |
| 15:26.47 | huey23 | [TK]D-Fender: i believe i am more clueless about the software that runs the phone system here because that's what is overriding the phone |
| 15:27.42 | [TK]D-Fender | huey23: Well unless somebody is actually going to sit down and dedicate a few hours video conferencing what you're doing, this will be even more painful.... |
| 15:29.23 | [TK]D-Fender | Katty: The solution to your office woes! http://bestpicever.com/pic-1559-Hello-Kitty-AK47 |
| 15:29.53 | *** join/#asterisk entelechy (n=chatzill@mail.beanproducts.com) |
| 15:30.05 | huey23 | [TK]D-Fender: ok...thanks for the help but, the phone setup is fine, something in the asterisk system is overriding my phone settings when i input them or the phone doesn't have proper permissions to change it's own configs |
| 15:30.42 | [TK]D-Fender | huey23: Sorry, but that is quite simply impossible. * does not hand out IP addresses or configure your phone. |
| 15:31.12 | [TK]D-Fender | huey23: And permissions is up to you based on how you are provisioning them |
| 15:31.35 | badcfe | hello. is there a way to continue dialplan navigation when the callee of the dial() application hangs up? i want to keep the callers leg and continue ivr on that channel .. |
| 15:32.21 | [TK]D-Fender | badcfe: "g" |
| 15:32.40 | [TK]D-Fender | NEXT!@!@! (c) BKW |
| 15:32.41 | badcfe | [TK]D-Fender: ah an option to dial i guess. thanks! |
| 15:32.48 | huey23 | the * box holds the config files...* may not hand them out...but it holds them and that is where i have to change them |
| 15:33.09 | huey23 | [TK]D-Fender: because obviously the phone cannot change them |
| 15:33.46 | [TK]D-Fender | huey23: more appropriately "the server that just happens to have Asterisk installed on it too". |
| 15:33.56 | ManxPower | huey23: asterisk box != asterisk |
| 15:34.16 | [TK]D-Fender | huey23: And you cannot set network parameters in provisioning, only direct on the bootrom and through DHCP |
| 15:34.20 | huey23 | [TK]D-Fender: very well |
| 15:34.32 | ManxPower | there is NOTHING you can do in the Asterisk config to configure your phones. |
| 15:35.00 | huey23 | [TK]D-Fender: i can change the .cfg on the server...and that seems like what i am going to have to do |
| 15:35.13 | [TK]D-Fender | huey23: Not for IP address issues.... |
| 15:35.16 | keith4 | does huey23 have one box that's server dhcp, tftp, and running asterisk? |
| 15:35.21 | [TK]D-Fender | huey23: (the PHONE'S that is) |
| 15:35.34 | keith4 | s/server/serving |
| 15:36.07 | huey23 | [TK]D-Fender: it's sntp issue...not ip |
| 15:36.09 | [TK]D-Fender | keith4: You know I don't recall even hearing about what METHOD was being used to provision the phones.... |
| 15:36.43 | keith4 | that's unusual... don't you usually demand all possible information before helping someone? ;-) |
| 15:37.03 | badcfe | [TK]D-Fender: when i do g then asterisk sends a CANCEL to the outbound gw. hmm what i meant was that i want to continue the dialplan _after_ the destination hups, not emidiately .. |
| 15:37.53 | badcfe | [TK]D-Fender: sorry, i forgot to tell that the outbound channel is SIP. when i use g option in dial, it does CANCEL just after receiving trying |
| 15:38.27 | [TK]D-Fender | badcfe: if the remote side cancels, dialplan continues on anyways REGARDLESS of "g" |
| 15:38.57 | ManxPower | badcfe: paste the ACTUAL Dial line you are having problems with. |
| 15:39.47 | [TK]D-Fender | huey23: Well is your DHCP server passing an NTP server address? |
| 15:39.55 | ManxPower | both from extensions.conf and when you see that line in the CLI. only paste ONE like for each of the two places. |
| 15:41.07 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:43.09 | badcfe | ManxPower: its just the Dial(SIP/${TARGET}@peer_a|g) where having that "g" option maked the _local_ asterisk cancel the call once the remote sip proxy tells it 100 Tryin |
| 15:43.23 | badcfe | <PROTECTED> |
| 15:44.08 | [TK]D-Fender | badcfe: pastebin your dialplan and CLI output of the failed attempt please... |
| 15:44.39 | ManxPower | badcfe: you are telling asterisk to timeout the call in "g" seconds |
| 15:44.50 | ManxPower | as "g" is not a number.... |
| 15:45.00 | ManxPower | try Dial(SIP/${TARGET}@peer_a,,g) |
| 15:45.02 | [TK]D-Fender | badcfe: And yes... it WOULD be nice if you put your parameters in the RIGHT order :p |
| 15:45.05 | ManxPower | or || of course. |
| 15:45.27 | ManxPower | [TK]D-Fender: I thought asterisk was supposed to spit out an error if the timeout is not a number. |
| 15:45.43 | [TK]D-Fender | ManxPower: Expectations-- |
| 15:46.01 | ManxPower | I remember whining and yelling and complaining about that issue a year or two ago and the developers added that feature to make me stop bothering them |
| 15:46.45 | *** join/#asterisk blq (n=Bl@dslb-088-067-043-146.pools.arcor-ip.net) |
| 15:48.50 | ManxPower | <PROTECTED> |
| 15:48.50 | ManxPower | Oct 19 10:48:35 WARNING[31822]: app_dial.c:1214 dial_exec_full: Invalid timeout specified: 'g' |
| 15:49.01 | ManxPower | Yup, it DOES generate a WARNING. |
| 15:49.17 | [TK]D-Fender | ManxPower: \o/ |
| 15:49.32 | ManxPower | so either badcfe is an idiot or it was removed in 1.4 |
| 15:50.27 | Uatec | hey |
| 15:50.35 | Uatec | in my CLI i'm getting: Oct 19 16:53:02 NOTICE[12466]: chan_misdn.c:4011 cb_events: Got Unknown Event |
| 15:50.37 | Uatec | periodically |
| 15:50.55 | Uatec | every 21 seconds past the minute and ever 2 seconds past the minute |
| 15:51.07 | Uatec | how can i find out what even it received/ |
| 15:51.08 | Uatec | ? |
| 15:51.18 | badcfe | im an idiot if you like, the cancel was due to an error in the dialplan. asterisk cancel the outbound call cause i had a shit here locally |
| 15:51.31 | badcfe | im at version 1.4.13 |
| 15:51.46 | ManxPower | badcfe: Asterisk should have generated an error when you used "g" as a timeout. |
| 15:52.06 | ManxPower | or more correctly a warning |
| 15:52.35 | badcfe | it did. i was blind |
| 15:52.49 | badcfe | cause i had sip debug enabled. ok no excuse |
| 15:52.50 | [TK]D-Fender | Or perhaps verbosity was not set to a level to display it. Or... he is perhaps just a little out of focus... |
| 15:52.59 | [TK]D-Fender | bile-- |
| 15:53.02 | badcfe | im out of focus |
| 15:53.16 | ManxPower | badcfe: the biggest problem with SIP debug is it makes non-sip errors hard to find. That is why I only use sip debug as a last resort |
| 15:53.31 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-a7b48a208b7125bc) |
| 15:53.39 | ManxPower | most errors are not something that sip debug will help with. |
| 15:53.46 | badcfe | wich leads us smoothly to my other question: is there some way of continueing dialplan after _all parties_ have hung up |
| 15:54.31 | badcfe | hehe. i actually now calls a macro on a hangup extention and i see that only the first gets done and then whole the channel is gone before the rest is execed. |
| 15:54.37 | ManxPower | badcfe: no. "g" will continue the dialplan if the called party hangs up. exten => h will be run when the callER hangs up |
| 15:54.38 | [TK]D-Fender | badcfe: "h" <- |
| 15:54.42 | _x86_ | badcfe: use the h extension |
| 15:54.43 | huey23 | [TK]D-Fender: what do you mean "passing" an address? |
| 15:55.01 | ManxPower | and don't put the exten => h in an include => It won't work as you expect. |
| 15:55.44 | badcfe | yesyes. the caller hangsup, and the h extention is in effect. but i call a macro there, wich dies in the middle of excecution (wich does not happen for this macro when the caller is still there) |
| 15:55.45 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:55.53 | k31th | seems impossible to find a isdn bri card, only manufacture i can find is digium. |
| 15:56.31 | badcfe | ManxPower: you mean i shouldnt put exten => in some file that i include from the dialplan? |
| 15:56.44 | ManxPower | badcfe: no, I said DO NOT do that. |
| 15:56.57 | ManxPower | put the exten => h in the same context as the Dial |
| 15:57.35 | *** join/#asterisk exvito (n=exvito@195.245.132.93) |
| 15:57.47 | *** join/#asterisk sevard (n=sev@192.235.0.85) |
| 15:59.20 | badcfe | no this problem i have with the macro is in a context where i sit in a waitexten and the caller hangsup |
| 15:59.30 | *** join/#asterisk skirmisha (n=viki@87-126-55-7.btc-net.bg) |
| 15:59.44 | skirmisha | anyone famialr with fax over ip |
| 15:59.52 | badcfe | http://pastebin.ca/742513 |
| 16:00.42 | [TK]D-Fender | badcfe: ... "h" <--- |
| 16:00.49 | badcfe | you see that the macro get interupted |
| 16:01.03 | badcfe | why does it not complete execution this macro? |
| 16:01.14 | skirmisha | ???/ |
| 16:01.38 | De_Mon | ~fax |
| 16:01.38 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 16:02.04 | De_Mon | really? heh |
| 16:02.56 | De_Mon | [TK]D-Fender where do you see "h" ? |
| 16:03.02 | De_Mon | oh, nevermind I see it :) |
| 16:04.21 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 16:04.27 | skirmisha | any idea how to detect fax on sip calls? |
| 16:05.45 | De_Mon | ~hylafax |
| 16:05.46 | jbot | A telecommunication system for UNIX systems. URL: http://www.hylafax.org |
| 16:06.00 | De_Mon | hrm oh well |
| 16:06.12 | skirmisha | is it working without ISDN cards? |
| 16:06.19 | skirmisha | i have pure ip to ip network |
| 16:07.38 | skirmisha | hehe |
| 16:07.47 | skirmisha | comon guys |
| 16:07.53 | skirmisha | any ideas and experience |
| 16:08.43 | tzafrir_home | k31th, there are plenty of cards |
| 16:08.56 | *** join/#asterisk tripps (n=ss@c-76-31-153-101.hsd1.tx.comcast.net) |
| 16:09.00 | tzafrir_home | disclaimer: I work for a company that produces one |
| 16:09.08 | k31th | tzafrir_home: which is? |
| 16:09.22 | tzafrir_home | k31th, for instance, the openvox clone is a clone of the Junghanns one |
| 16:09.32 | tzafrir_home | (I work for xorcom) |
| 16:10.36 | *** join/#asterisk blq (n=Bl@dslb-088-066-254-168.pools.arcor-ip.net) |
| 16:11.41 | tzafrir_home | Basically you'll find cheaper single-port cards that are generally for the consumer market and more expensive multi-port cards which are priced for "proffessionals". |
| 16:11.47 | badcfe | does someone understand whats going on in the snippet here? http://pastebin.ca/742513 |
| 16:12.27 | ManxPower | badcfe: not until you paste the actual dialplan |
| 16:12.46 | ManxPower | like from extensions.conf |
| 16:13.35 | *** join/#asterisk ToTo (n=ToTo@207.176.6.103) |
| 16:14.19 | *** join/#asterisk saint_ (n=saint@c-69-242-118-124.hsd1.nj.comcast.net) |
| 16:15.27 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:15.55 | ManxPower | skirmisha: don't expect FAX to work over SIP. If you can still find the code app_nvfaxdetect.c would do it for you. |
| 16:20.21 | *** join/#asterisk UnixDog (n=UnixDog@adsl-69-234-206-85.dsl.irvnca.pacbell.net) |
| 16:20.31 | UnixDog | zeeek you here |
| 16:20.40 | badcfe | seems like that when a macro gets called from the h extention due to caller hangup, then sometimes just the first part of the macro gets executed. the rest is not because the channel goes away. however, it seem as tho the first application called in the macro does always get executed |
| 16:22.07 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 16:22.21 | ManxPower | badcfe: I doubt you can call a macro from exten => h |
| 16:23.16 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:23.40 | Zeeek | Voip Users COnferenc e is about on |
| 16:23.50 | Zeeek | http://voipusersconference.org about asterisk |
| 16:23.51 | badcfe | ManxPower: well i do a Goto from h right to a s,1,Macro(celcdr) and it goes like shown in http://pastebin.ca/742513 |
| 16:24.36 | Zeeek | irc on freenode.net #voip-users-conference |
| 16:24.38 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:24.46 | Zeeek | give us a shout |
| 16:25.12 | badcfe | maybe i should embed the macro invokation into Dial. hmm i should read thre that dial application doc. |
| 16:25.22 | ManxPower | of course without seeing your actual dialplan this is pure speculation |
| 16:26.04 | ManxPower | badcfe: good. Now I know what it looks like on the console. show me what it looks like in extensions.conf. I will NOT ask a 4th time. |
| 16:26.51 | Dandre | hello, |
| 16:27.00 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 16:27.20 | ManxPower | "show dialplan" is not the same as "show me the info from extensions.conf. |
| 16:30.01 | Dandre | I am trying to update a config file with the manager interface as stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+UpdateConfig |
| 16:30.01 | Dandre | The problem is that I want to update foo variable only if it is set to bar. So I have set Match-xxxxxx: bar in my update command but it doesn't seem to be taken into account |
| 16:30.51 | Dandre | can the Match line be used elsewhere than for a delete command? |
| 16:31.43 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:33.21 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 16:33.44 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:34.02 | ManxPower | I give up. badcfe, you are on your own |
| 16:37.20 | *** join/#asterisk kv0s (n=kv0s@p4FD2777B.dip.t-dialin.net) |
| 16:37.22 | kv0s | Hi! |
| 16:38.20 | kv0s | Is it possible to use a special trunk only for calls to mobiles? So i can use a special SIP-Provider for my mobile calls ... and my normal isdn-line for use with flatcalls....?!? |
| 16:38.44 | exvito | hi... I googled for this but couldn't find a clear enough answer... I'm doing some initial tests with iaxmodem so as to integrate asterisk/PSTN with hylafax. However, the IAX registration period, between iaxmodem and asterisk over 127.0.0.1, seems to be forced to 60 seconds. Everything seems to work fine but I keep getting registration messages polluting my logs... Any ideas on how to change this behaviour ? |
| 16:39.13 | Dandre | I am trying to update a config file with the manager interface as stated here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+UpdateConfig |
| 16:39.13 | Dandre | The problem is that I want to update foo variable only if it is set to bar. So I have set Match-xxxxxx: bar in my update command but it doesn't seem to be taken into account |
| 16:39.13 | Dandre | can the Match line be used elsewhere than for a delete command? |
| 16:44.29 | _x86_ | hmm |
| 16:44.29 | _x86_ | http://pastebin.ca/742543 |
| 16:44.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:45.19 | _x86_ | note how wanpipemon -i w2g1 -c Ta is showing all kinds of line code violations, bit errors, and OOF errors... |
| 16:45.27 | _x86_ | what would this indicate to you guys? |
| 16:51.22 | *** join/#asterisk techie (n=techie@adsl-76-214-12-76.dsl.lsan03.sbcglobal.net) |
| 16:51.29 | *** part/#asterisk techie (n=techie@adsl-76-214-12-76.dsl.lsan03.sbcglobal.net) |
| 16:53.18 | exvito | ...answer to myself (for the sake of channel logs!): change minregexpire/maxregexpire in iax.conf |
| 17:02.17 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:03.11 | *** join/#asterisk celord]cR (n=cesar@201.195.35.62) |
| 17:16.30 | *** join/#asterisk Shaun222 (n=shaun@ip68-4-127-67.oc.oc.cox.net) |
| 17:16.41 | Shaun222 | how can i view if a sip extension is on a call? |
| 17:17.15 | [TK]D-Fender | Shaun222: "show channels" |
| 17:18.11 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:18.11 | tzafrir_home | sip show channels |
| 17:18.25 | r0d3nt | viva toorcon. |
| 17:21.15 | Strom_M | r0d3nt: you're at toorcon? |
| 17:21.15 | r0d3nt | yessir. |
| 17:21.15 | Strom_M | awesome. i'm coming in this afternoon |
| 17:21.15 | r0d3nt | in a seminar right now |
| 17:21.15 | r0d3nt | sweeeeeet |
| 17:21.17 | r0d3nt | i look forward to meeting you again =) |
| 17:21.19 | Strom_M | plus i'm talking on sunday |
| 17:21.23 | *** part/#asterisk guyzmo (n=guyzmo@nenya.mithrandir.net) |
| 17:21.32 | r0d3nt | sweeeeeet. |
| 17:21.38 | r0d3nt | i'll be there =) |
| 17:21.42 | Strom_M | yay |
| 17:22.43 | keith4 | i suggested adding 'exten => _s-.,1,Goto(s-NOANSWER,1)' to the standard extension macro, to the asterisk admin at work... and he says it's not valid syntax |
| 17:22.59 | keith4 | he's running 1.4.2... i have 1.2.13 |
| 17:23.10 | Strom_M | keith4: that's valid |
| 17:23.42 | keith4 | perhaps i mis-interpreted his email |
| 17:24.02 | keith4 | maybe he meant that adding that line didn't resolve the problem |
| 17:24.49 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 17:25.01 | [TK]D-Fender | keith4: That could be true since we haven't seen what you're doing in its entirety |
| 17:25.34 | keith4 | i'm not doing anything, i'm trying to get the damn asterisk admin to fix his setup |
| 17:25.36 | De_Mon | tell him to get his ass on irc |
| 17:25.42 | keith4 | i just did ;-) |
| 17:26.38 | keith4 | okay, here's the deal. we ahve some crap-tastic intecom pbx (at least, I have an intecom digital phone...), and he's grafted asterisk onto that |
| 17:27.07 | [TK]D-Fender | All hail the FrankenPBX |
| 17:27.21 | [TK]D-Fender | Now with marshmallows! |
| 17:27.26 | De_Mon | when tk said we haven't seen what you're doing in its entirety I do believe he was talking about your dialplan |
| 17:27.27 | keith4 | i have my "real" pbx extension forwarded to my asterisk extension... if my SIP phone is unavailable, i would expect people to be dumped into asterisk voicemail |
| 17:27.45 | lirakis | I am trying to SET(CDR(userfield=blah)) but it is not showing up in mysql cdr's ... |
| 17:28.02 | blitzrage | lirakis: wrong format |
| 17:28.03 | [TK]D-Fender | keith4: asterisk extension? ASSUMED voicemail? |
| 17:28.05 | [TK]D-Fender | ~assume |
| 17:28.06 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 17:28.08 | lirakis | mysql_custom.conf has userfield mapped .. and i see it setting |
| 17:28.13 | blitzrage | Set(CDR(userfield)=blah) |
| 17:28.13 | keith4 | heh |
| 17:28.23 | De_Mon | he said expect not assume ;) |
| 17:28.39 | lirakis | blitzrage: yeah .. sorry i do " Executing Set("Zap/23-1", "CDR(userfield)=balance") in new stack" |
| 17:28.45 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1167861656.dsl.bell.ca) |
| 17:28.55 | [TK]D-Fender | De_Mon: Sad attempt to dodge whats coming :p |
| 17:28.59 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 17:29.04 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:29.12 | keith4 | i have 2 phone numbers in the office... one on the proprietary pbx, one on asterisk |
| 17:29.24 | De_Mon | [TK]D-Fender I just know how much though went into AVOIDING that word :) |
| 17:29.25 | keith4 | i hate the proprietary system, so i forwarded that number to my asterisk number |
| 17:29.26 | lirakis | blitzrage: but it is not showing up in MySQL cdr table... im not sure if its showing in the flat file.. its hard to dissect them |
| 17:29.33 | [TK]D-Fender | keith4: Stop describing and show us dialplan, CLI output, etc.... |
| 17:29.37 | blitzrage | not sure -- I avoid MySQL like the plague |
| 17:29.49 | [TK]D-Fender | blitzrage: I avoid cliches like the plague... |
| 17:29.53 | keith4 | [TK]D-Fender: it's not my system! i have limited access... hold on |
| 17:30.00 | blitzrage | <-- cliche to the max |
| 17:30.27 | [TK]D-Fender | keith4: WHINEcryBITCHnagCUSSscreamSIGHwhimper |
| 17:30.47 | [TK]D-Fender | keith4: #drphil |
| 17:30.51 | keith4 | lol |
| 17:30.53 | De_Mon | lol |
| 17:31.10 | De_Mon | keith4 you have two choices 1) stop caring or 2) just do it |
| 17:31.17 | lirakis | blitzrage: hrmm.. (shrug) |
| 17:31.24 | [TK]D-Fender | keith4: pwned |
| 17:31.25 | blitzrage | ya, not too sure |
| 17:31.27 | keith4 | damn! i tried to call TK's bluff... and he wasn't bluffing |
| 17:31.34 | lirakis | any one else have a tip for getting userfieild into the mysql cdr table?/ |
| 17:31.45 | [TK]D-Fender | :D |
| 17:31.49 | keith4 | i'm trying to get the * admin in here... in the meantime, this is all I know: http://pastebin.ca/742591 |
| 17:31.52 | bkruse | this is off topic, but does anyone know how to determine what app is using your soud device? |
| 17:32.07 | bkruse | s/soud/sound/g |
| 17:32.09 | keith4 | bkruse: lsof |
| 17:32.15 | De_Mon | lirakis you know whats going in the userfield right? so grep the flatfile for that and find out |
| 17:32.21 | bkruse | keith4: thats what I thought, I must have been using the wrong /dev device |
| 17:32.33 | De_Mon | bkruse alsa or oss? |
| 17:32.40 | bkruse | De_Mon: alsa |
| 17:32.52 | bkruse | keith4: lsof never seems to work with the /dev device :/ |
| 17:33.21 | keith4 | bkruse: lsof | grep /dev/snd ? |
| 17:33.40 | De_Mon | bkruse lsof should tell you whats using /dev/snd |
| 17:34.01 | bkruse | keith4: k, trying :] |
| 17:34.06 | roxlu | hi |
| 17:34.09 | De_Mon | or /dev/dsp (Iirc) |
| 17:34.16 | bkruse | keith4: perfect! |
| 17:34.25 | bkruse | De_Mon, keith4: thank you :] |
| 17:34.31 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 17:35.36 | De_Mon | google also suggests fuser -v /dev/dsp |
| 17:35.37 | De_Mon | http://www.webservertalk.com/archive291-2006-7-1469073.html |
| 17:35.46 | bkruse | thats the one I was tinking of.. |
| 17:35.47 | bkruse | I have |
| 17:35.49 | bkruse | thinking* |
| 17:35.58 | [TK]D-Fender | keith4: -- Channel 0/1, span 1 got hangup <-- THEY. HUNG. UP. |
| 17:36.34 | roxlu | I just bough a subscrbtion for a voip account... Can someone explain me abit how I can connect Asterisk to this account? or if this is even possible? |
| 17:36.37 | [TK]D-Fender | errr |
| 17:36.41 | [TK]D-Fender | bad aim :| |
| 17:36.58 | [TK]D-Fender | asdasdasdadklkad |
| 17:37.20 | De_Mon | roxlu um, ask your voip provider |
| 17:37.37 | [TK]D-Fender | roxlu: ... |
| 17:37.39 | [TK]D-Fender | ~book |
| 17:37.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:37.40 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^ |
| 17:38.50 | ManxPower | This is not #teach-me-to-use-asterisk |
| 17:38.53 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 17:40.06 | [TK]D-Fender | keith4: I think I see why.... pastebin your entire dialplan please. |
| 17:40.14 | roxlu | De_Mon: I'm just wondering how this works in gerenal? I'm using a softphone now and I login using my porviders account, but how can I put asterisk between is? |
| 17:40.17 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:40.39 | [TK]D-Fender | roxlu: ..... |
| 17:40.41 | [TK]D-Fender | ~book |
| 17:40.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 17:40.43 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 17:40.49 | keith4 | [TK]D-Fender: i'll try to get it... could be a while |
| 17:41.06 | roxlu | oke thanks |
| 17:41.08 | De_Mon | roxlu download that book it explains the basics |
| 17:41.26 | roxlu | Ok I'll |
| 17:41.30 | [TK]D-Fender | roxlu: * regs to your provider, your phone regs to *. * takes calls from each however you choose and processes the calls however you choose |
| 17:41.46 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 17:41.47 | mocker | Need to find some free time to finish my NSLU2 setup for the next user group meeting. |
| 17:41.48 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
| 17:41.49 | keith4 | [TK]D-Fender: fwiw, it happens SIP to SIP, so zap channel isn't involved |
| 17:42.24 | [TK]D-Fender | keith4: Again I'm quite certain I know why... I just need the evidence :) |
| 17:43.01 | keith4 | ok, stand by |
| 17:43.03 | roxlu | [TK]D-Fender: but is that also possible with a softphone? |
| 17:43.24 | [TK]D-Fender | roxlu: Phone = spftphone for you. |
| 17:43.29 | roxlu | oke |
| 18:01.48 | *** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162) |
| 18:01.58 | *** join/#asterisk naitram (n=chatzill@216.77.58.40) |
| 18:02.14 | naitram | anyone used adhearsion? |
| 18:02.42 | JerJer | naitram: i have one time |
| 18:03.06 | JerJer | i don't completely get it - then again ruby is foreign to me |
| 18:03.07 | naitram | JerJer: what did you think? Ready for prime time? |
| 18:03.26 | JerJer | last i knew Jay was doing a major overhaul, so i kinda don't think so |
| 18:03.29 | JerJer | but its a guess |
| 18:04.04 | JerJer | i am currently experimenting with my own framework - Using Catalyst, which is Perl |
| 18:05.04 | JerJer | which I am having some of the same conceptual problems - so perhaps my problem isn't with ruby so much than it is with the MVC concept |
| 18:05.19 | JerJer | then again i have always hated OO programing |
| 18:07.24 | naitram | JerJer: I am trying to control the dial plan (call, hangup etc..) from an external app (php scripting) but cant seem to get the AMI to work right. Thought a higher level approach might help |
| 18:07.29 | keith4 | [TK]D-Fender: http://pastebin.ca/742630 |
| 18:07.51 | [TK]D-Fender | keith4: priorityjumping=yes <-----------GUILTY! |
| 18:08.19 | [TK]D-Fender | keith4: that = EVIL |
| 18:08.19 | keith4 | don't axe-murder the messenger! |
| 18:08.25 | keith4 | this isn't my setup |
| 18:08.29 | keith4 | MY setup works fine |
| 18:08.46 | keith4 | wtf is priorityjumping? |
| 18:08.47 | [TK]D-Fender | keith4: iautofallthrough=yes <- I also don't recommend |
| 18:09.02 | [TK]D-Fender | keith4: It jumps to n+101 on busy, etc... |
| 18:09.10 | [TK]D-Fender | keith4: 1.0 grade shit |
| 18:09.26 | keith4 | will removing that break anything else in the dial plan? |
| 18:09.33 | keith4 | same for autofallthrough? |
| 18:09.40 | [TK]D-Fender | keith4: nothing that doesn't deserver to be written properly :) |
| 18:10.10 | Katty | [TK]D-Fender: i found a headbanging, moshing, parrot. |
| 18:10.16 | keith4 | for example.... look at [outbound-local]... isn't that going to break? |
| 18:10.17 | [TK]D-Fender | keith4: Oh, and your exten => _s-.,1 line WAS a good idea. |
| 18:10.18 | Katty | [TK]D-Fender: i r teh giggle. |
| 18:10.21 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:10.28 | [TK]D-Fender | keith4: For your macro |
| 18:10.52 | [TK]D-Fender | keith4: exten => _91NXXNXXXXXX,102,Congestion() <-----ICK! Yes this guy is a schmuck |
| 18:11.04 | keith4 | he's... new to this |
| 18:11.33 | *** part/#asterisk naitram (n=chatzill@216.77.58.40) |
| 18:11.35 | [TK]D-Fender | :p |
| 18:11.54 | [TK]D-Fender | keith4: Ok, go kick his ass for all the trouble and time this has taken. |
| 18:12.02 | keith4 | that's why when all the channels from the * box to the crap-PBX are taken, i just get a busy signal |
| 18:12.06 | [TK]D-Fender | Katty: a NEW one? |
| 18:12.38 | Katty | [TK]D-Fender: `oh yes. |
| 18:12.41 | [TK]D-Fender | keith4: I wasn't saying Congestion was bad... look at the PRIORITY |
| 18:12.44 | Katty | [TK]D-Fender: and a beatboxing parrot. |
| 18:12.48 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 18:12.54 | [TK]D-Fender | Katty: WAY too much free time... |
| 18:13.14 | Katty | [TK]D-Fender: http://www.youtube.com/watch?v=UnFV-fvgOu0 just view. it's very short. |
| 18:13.23 | Katty | [TK]D-Fender: 26 seconds. |
| 18:13.30 | keith4 | [TK]D-Fender: right. that's probably why he turned priorityjumping on |
| 18:13.37 | keith4 | i'll set him straight, no worries |
| 18:19.58 | mocker | ~adhearsion |
| 18:21.31 | Shaun222 | whats the best codec to use with polycom 5xx 6xx phones? |
| 18:21.38 | Qwell | Shaun222: what are the xx? |
| 18:21.46 | Shaun222 | the default asterisk confs show ulaw |
| 18:21.56 | Shaun222 | 501's 550's 601's.. etc. |
| 18:22.03 | Qwell | Shaun222: if xx == 50, then g722 |
| 18:22.07 | Qwell | otherwise g711 |
| 18:22.20 | Qwell | for the "best" codec - assuming "best" means "best sounding" |
| 18:22.22 | Katty | [TK]D-Fender: lol, found a bird that says 'got a beak! i gots a beakbeakBEAK' |
| 18:23.31 | mocker | Qwell: Did the stuff about g722 in http://blogs.eweek.com/signaling_it/content001/voip/hd_voice_hampered_by_asterisks_codec_negotiations.html get resolved? |
| 18:24.15 | De_Mon | Katty I am amazed at the number of beatboxing birds on utube |
| 18:24.17 | Shaun222 | Qwell: best meaning.. being able to communicate with the other person with the least amount of problems.. clear sound, minimal breakups, noise, jitter, etc.. |
| 18:24.44 | Katty | De_Mon: indeed. |
| 18:25.16 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:26.07 | Shaun222 | right now looks like the polycom's are using ulaw. |
| 18:27.20 | Katty | De_Mon: i bet birds require a LOT of attention |
| 18:27.52 | De_Mon | Katty did you hear about the watchdog -parrot that caught a home invader this week? |
| 18:28.33 | gardo | Anyone has experience using Adit 600 w/ TE405P card? |
| 18:29.12 | Katty | De_Mon: no! |
| 18:29.14 | Katty | De_Mon: link? |
| 18:29.27 | De_Mon | I'll have to find it, gimme a few minuites it was on cnn |
| 18:29.38 | Netgeeks | Birds do require alot of attention, nearly as much as dogs |
| 18:30.35 | _x86_ | Qwell: hey, you wouldnt happen to have any ideas on why i'm getting all these errors and sporadic T1 drops: http://pastebin.ca/742543 |
| 18:31.14 | Qwell | _x86_: you're using Sangoma |
| 18:32.14 | _x86_ | Qwell: that's a generic response ;) |
| 18:32.24 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 18:33.41 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 18:33.54 | Jason99 | Is there a way to execute a context when a phone registers? I'm trying to log in MySQL everytime a phone registers to the server |
| 18:34.01 | mocker | _x86_: Did you use a script to generate that pastebin? ;) |
| 18:34.07 | Qwell | Jason99: regexten/regcontext |
| 18:34.14 | Qwell | erm, wait, no |
| 18:34.26 | Jason99 | hehe |
| 18:35.09 | Jason99 | so regexten doesn't do that? |
| 18:35.49 | *** join/#asterisk luke-jr (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 18:36.19 | mocker | _x86_: You're sure it's not the line? |
| 18:36.21 | Katty | [TK]D-Fender: my lovely afternoon of youtube birdie watching was ruined by a call from the telco :< |
| 18:36.26 | Katty | [TK]D-Fender: please burn them. kthx. |
| 18:36.38 | mocker | Can you have the telco loop and test clean to the demarc? |
| 18:36.41 | hmmhesays | what'd they have to say? |
| 18:37.09 | Katty | hmmhesays: well, oddly ehough, they sounded like the birds... "squakkkkkkk squaaakkk silly bird silly bird" |
| 18:37.37 | _x86_ | mocker: no script... Sangoma's tech support asked for a bunch of crap ;) |
| 18:37.52 | luke-jr | Cisco 7660 refuses to use DHCP's TFTP and won't let me change it-- any ideas? |
| 18:37.53 | _x86_ | mocker: LEC is running patterns to the smart jack clean |
| 18:38.15 | mocker | _x86_: Can they throw a loop signal to the card? |
| 18:38.21 | mocker | I know that some cards support that. |
| 18:38.27 | mocker | Or you could probably manually loop to them. |
| 18:39.18 | [TK]D-Fender | luke-jr: Cisco & AEL? load chan_masochist.so :p |
| 18:39.59 | hmmhesays | hmm how do I tell the clock on the polycoms to sync with a time server |
| 18:40.04 | luke-jr | [TK]D-Fender: without the TFTP working, it doesn't even hit Asterisk :þ |
| 18:41.07 | mocker | luke-jr: Did you set the option 66? |
| 18:41.19 | Katty | hmmhesays: through their IP >> General |
| 18:41.35 | Katty | hmmhesays: there's also a bit on the phone, but it's easier to paste it in (= |
| 18:41.43 | Katty | hmmhesays: rather than going through the whole menu thingy |
| 18:41.51 | [TK]D-Fender | hmmhesays: sntp <- sip.cfg |
| 18:41.56 | Jason99 | Qwell, i just tried regexten/regcontext but doesnt appear to work |
| 18:42.01 | mocker | luke-jr: Something like 'option boot-server code 66 = string;' in your dhcpd.conf |
| 18:42.06 | Katty | or that :P |
| 18:42.13 | Qwell | Jason99: yeah, I misread what you wanted |
| 18:43.07 | [TK]D-Fender | Jason99: option : use regexten/regcontext, and scan that context periodically and trigger on change. |
| 18:43.34 | [TK]D-Fender | Jason99: but no "live" way I can think of that doesn't involve source code |
| 18:44.05 | luke-jr | mocker: DHCP is working fine; this is one of many phones that just isn't taking it |
| 18:44.19 | Jason99 | I dont really understand how regexten/regcontext works... I tried as per (http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+regcontext) but it doesnt seem to work like they say |
| 18:44.24 | luke-jr | all the other phones get it from DHCP correctly and allow me to edit manually |
| 18:44.28 | luke-jr | this phone won't do either |
| 18:44.48 | mocker | luke-jr: Same firmware versions? |
| 18:45.04 | luke-jr | mocker: not sure; how can I check? |
| 18:45.31 | lirakis | haha! i figured out setting the userfield in MySQL ... in cdr_mysql.conf you have to uncomment the userfield setting (duh) |
| 18:45.58 | codefreeze | lirakis: congrats! |
| 18:46.16 | luke-jr | POS3-08-2-00 |
| 18:46.34 | mocker | luke-jr: POS, there's the problem. |
| 18:46.42 | mocker | :P |
| 18:46.49 | luke-jr | :/ |
| 18:46.53 | [TK]D-Fender | prophetic++ |
| 18:47.00 | luke-jr | they're both that |
| 18:47.11 | *** part/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 18:47.11 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 18:47.17 | *** part/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 18:47.25 | mocker | luke-jr: And you've swapped them to make sure it's not the port they're connected to? |
| 18:47.30 | mocker | i.e. switch working one with bad one |
| 18:47.42 | luke-jr | yep |
| 18:47.47 | mocker | hmm. |
| 18:48.00 | mocker | that sucks then. tcpdump to see if it's even trying then? |
| 18:48.23 | luke-jr | tcpdump where? |
| 18:48.28 | luke-jr | can't get in the middle-- it's PoE |
| 18:48.51 | peanut- | [Oct 19 18:44:44] NOTICE[7983]: chan_sip.c:14848 handle_request_register: Registration from '200 <sip:200@10.0.4.6>' failed for '10.0.3.5' - No matching peer found trying to setup hard phone, has user and pass which isn't 200, that's in the "phone number" field which sets itself to "2004" if left blank, why is it not connecting? |
| 18:49.28 | mocker | luke-jr: On the dhcp/tftp server. |
| 18:49.28 | *** join/#asterisk knarfly (n=vladimir@adsl-11-248-246.mia.bellsouth.net) |
| 18:49.46 | luke-jr | tftp doesn't see anything |
| 18:50.13 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 18:50.20 | ManxPower | the phone is trying to register as user "200". |
| 18:51.17 | [T]ank | i dropped all calls in and out to my asterisk server. When looking at the CLI> I was getting this error message: http://pastebin.ca/742680 what causes that and how is it resolved? |
| 18:52.14 | ManxPower | [T]ank: you are using IAX2 trunking, I assume. |
| 18:52.20 | [T]ank | yes |
| 18:52.33 | ManxPower | I believe there are some trunking options you can play with. |
| 18:53.32 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:53.42 | [T]ank | ManxPower: where would that be? |
| 18:54.13 | ManxPower | [T]ank: I looked in /path/to/src/asterisk/configs/iax.conf The same place I look at for getting option information |
| 18:54.26 | ManxPower | 'also, for some reason, I thought that bug was fixed long ago. what version of asterisk are you using? |
| 18:54.47 | ManxPower | [T]ank: BTW, just paste to the channel if it's just 1 or 2 lines. |
| 18:55.12 | [T]ank | i never do the paste right, DAMMIT!!! |
| 18:55.13 | [T]ank | :-D |
| 18:56.14 | hmmhesays | i'm having trouble finding where to set the sntp server address on the polycom ip 501 |
| 18:56.27 | ManxPower | hmmhesays: I always do it via DHCP. |
| 18:56.37 | hmmhesays | not an option here |
| 18:56.40 | [T]ank | its in the SIP.cfg file for the polycoms |
| 18:56.53 | [T]ank | and i think it is just labled ntp |
| 18:56.58 | hmmhesays | yeah I see all the time settings, but nowhere to put the host |
| 18:57.22 | [T]ank | should be something like ntp='' |
| 18:57.28 | [T]ank | and you fill in the '' |
| 18:57.36 | hmmhesays | tcpIpApp.sntp.address="" |
| 18:57.38 | hmmhesays | yeah found it |
| 18:58.25 | *** join/#asterisk exarv (n=robert@h8441179167.dsl.speedlinq.nl) |
| 18:58.29 | hmmhesays | thanks |
| 18:58.33 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 18:58.39 | *** join/#asterisk l2trace99 (n=asd@fl-67-76-209-28.sta.embarqhsd.net) |
| 19:00.06 | exarv | Anybody here, who knows a bit about generating CDR's in asterisk 1.4? |
| 19:03.56 | codefreeze | exarv: whats the prob? |
| 19:04.12 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
| 19:04.17 | exarv | I've recently upgraded from 1.2 to 1.4 |
| 19:04.46 | mocker | http://www.trixbox.org/forums/vendor-moderated-forums/hudlite-trixbox-ce/hudlite-keeps-downloading-0day |
| 19:05.02 | exarv | and we have a system, where you dial in, then in the ivr, you type another phonenumer, and through lcr the call is routed to outside |
| 19:05.12 | exarv | I used forkcdr and resetcdr in 1.2 |
| 19:05.23 | exarv | but in 1.4 it has a completly different behaviour |
| 19:05.52 | exarv | I need two cdr records. 1 with the dialin params, and the start/end time of the complete call |
| 19:06.02 | exarv | and ine cdr with only the dialout time |
| 19:06.10 | Sci_05 | can anyone give me an idea what would cause this to come up when doing an iax2 reload "chan_iax2.c:9071 set_config: Ignoring bindport on reload" It never did it before just poped up one day. |
| 19:06.26 | codefreeze | exarv: how do you do the dial out? |
| 19:06.26 | Sci_05 | I kow the bindport is in the iax.conf |
| 19:06.29 | exarv | but mostly i get the dialin cdr with a duration of 0 seconds. |
| 19:06.32 | knarfly | Can someone help me with this http://www.pastebin.ca/742690 |
| 19:06.46 | exarv | through a Dial command |
| 19:06.51 | knarfly | it doesn't redirect bitch's calls like I want |
| 19:06.53 | exarv | several channels (zap/sip/iax) |
| 19:07.31 | hmmhesays | so the new followme app seems pretty cool |
| 19:07.35 | [TK]D-Fender | ;exten => s/3152917411,n,Goto(blocking,s,1) ; Block Certain Callers <--- you can't just through in a callid filtered exten with "n", you have to have a seperate "1" for it to start |
| 19:07.37 | hmmhesays | too bad you can't db it |
| 19:07.54 | exarv | Dial(${DIALOUTNR},60,2) |
| 19:08.02 | [TK]D-Fender | hmmhesays: nothing you can't do with a tiny bit of dialplan.... it was completely unnecessary |
| 19:08.11 | hmmhesays | [TK]D-Fender: true |
| 19:08.26 | knarfly | [TK]D-Fender: can you explain a little more detail...I'm not sure I follow |
| 19:08.34 | djMax | I'm trying to upgrade a pretty old installation of * to 1.4.13. How would I go about finding out what types of things I need to watch out for? |
| 19:08.47 | djMax | (I see a bunch of "questionable" modules, first off) |
| 19:08.55 | *** join/#asterisk Lisa696 (i=julian@200.58.204.164) |
| 19:08.56 | codefreeze | exarv: you do any authentification? |
| 19:09.06 | Lisa696 | any person talk spanish |
| 19:09.07 | codefreeze | (just curious) |
| 19:09.17 | [TK]D-Fender | knarfly: your exten with the CID filter on it is an "n" priority. That means that you needed exten => s/3152917411,1 <- somewhere |
| 19:09.18 | exarv | codefreeze :) |
| 19:09.44 | exarv | codefreeze: no. we have 15 premium dialin numbers with several rates (from 0.03 euro /minute til 0.80 euro/minute. |
| 19:09.51 | mocker | djMax: UPGRADE.txt |
| 19:09.52 | [TK]D-Fender | knarfly: You can't just think that you can add the contidional stuff "between" the other extens priorities pysically because it LOOKS like its in "order" |
| 19:10.03 | exarv | and depending on the costs, you can dial to certain countries or not |
| 19:10.04 | [TK]D-Fender | knarfly: order is, by & large, an illusion. |
| 19:10.28 | knarfly | [TK]D-Fender: let me try to correct is based on this info. Thanks...I got this from an example on some site somewhere. |
| 19:10.40 | djMax | I guess the first question for upgrade.txt is that I have an SVN revision number, not a version number. How can I correlate the two? |
| 19:10.42 | roxlu | Can I only add one SIP account in xlite softphone? |
| 19:13.44 | [TK]D-Fender | roxlu: Yes |
| 19:13.57 | codefreeze | exarv: Well, the long and short of it, is that cdr's in 1.4 are being hacked on. Bug fixes only. Yes, they will be different than 1.2. No, they won't be perfect. When I 'finish', hopefully most of the big holes will be plugged. Can you pastebin the parts of your extensions.conf that answer the incoming call, and re-route it out with a dial, and do the forkcdr thing? |
| 19:14.26 | roxlu | [TK]D-Fender: is there an opensource softones which allows multiple sip? |
| 19:14.27 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:14.48 | [TK]D-Fender | roxlu: Ekiga |
| 19:15.01 | exarv | codefreeze: sure. how do you want the pieces of code? by mail? |
| 19:15.07 | roxlu | Thanks |
| 19:15.17 | Shaun222 | anybody know why my polycom is downloading the phone.cfg and saying it's current when i just changed it.. |
| 19:15.29 | deeperror | I have 2x channel banks and I'm looking to get the second one setup. How do i define the channels in zapata.conf for the second bank? Or a link to some info on this? |
| 19:15.32 | Shaun222 | my phone line configs are staying what they where in the previous config |
| 19:15.38 | ManxPower | djMax: and the output of "show version" is? |
| 19:15.53 | codefreeze | well, I guess the ideal right now will be pastebin. But hang onto it, we might want to turn this into a bug report |
| 19:15.58 | codefreeze | ~pb |
| 19:15.58 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:16.00 | knarfly | [TK]D-Fender: so would this be more like what works http://www.pastebin.ca/742700 |
| 19:16.01 | roxlu | [TK]D-Fender: .... and for the mac> |
| 19:16.34 | roxlu | wengophone maybe? |
| 19:16.49 | [TK]D-Fender | knarfly: Yes |
| 19:16.55 | exarv | codefreeze: aha, pastbin.. (reading wikipedia right now about it..) |
| 19:17.00 | [TK]D-Fender | roxlu: Dunno, go try |
| 19:17.11 | roxlu | yes okay |
| 19:17.29 | codefreeze | exarv: see what jbot says above.... I suggest pastebin.ca, I've had good luck with it |
| 19:20.30 | *** join/#asterisk vyamba (n=chatzill@194.42.96.226) |
| 19:20.53 | exarv | codefreeze: ok, busy copying / pasting the important parts of the code |
| 19:21.11 | Shaun222 | anybody know why my sip line settings are not updating from the config the phone just downloaded? |
| 19:21.53 | knarfly | [TK]D-Fender: thanks...as my grandpa used to say...that worked slicker than snot on a door knob. |
| 19:22.07 | [TK]D-Fender | ..... |
| 19:22.13 | djMax | ManxPower, SVN-trunk-r46489 |
| 19:22.28 | hmmhesays | [TK]D-Fender: have you ever seen a poly 501 not boot if it can't find the boot server? |
| 19:22.32 | [TK]D-Fender | Shaun222: since we have no way of knowing what you TRIED stting it up like... NO |
| 19:22.44 | ManxPower | djMax: trunk really doesn't have a version |
| 19:22.44 | [TK]D-Fender | hmmhesays: no. they will boot with the last config they loaded |
| 19:22.50 | hmmhesays | this one doesn't |
| 19:22.56 | Shaun222 | [TK]D-Fender: i'm using the standard phone.cfg and sip.cfg, they are downloaded via ftp on boot.. |
| 19:22.58 | hmmhesays | it hangs on "cannot find a boot server" |
| 19:23.03 | Shaun222 | i make a change to the line in the phone.cfg |
| 19:23.19 | Shaun222 | rebooted phone and it still is showing the old info.. |
| 19:23.24 | Shaun222 | the app.log downloads the config just fine. |
| 19:23.28 | [TK]D-Fender | Shaun222: and untill I see all of your configs I won't be able to say much... |
| 19:23.32 | ManxPower | Shaun222: and you are watching the logs of your FTP server |
| 19:23.40 | djMax | yeah, I figured, just not sure how to know whether my configs are going to explode or not |
| 19:23.40 | Shaun222 | ManxPower:yes. |
| 19:23.40 | ManxPower | to see the file being downloaded |
| 19:23.54 | djMax | other than the hard way. :) |
| 19:23.55 | [TK]D-Fender | Shaun222: and any settings made on the web interface or phone itself take precedence |
| 19:24.19 | Shaun222 | these settings where always set by the config. |
| 19:24.42 | [TK]D-Fender | Shaun222: Still not seeing anything.... |
| 19:25.11 | Shaun222 | what do you want to see and i'll show you |
| 19:25.59 | *** join/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 19:26.01 | [TK]D-Fender | <[TK]D-Fender>Shaun222: and untill I see all of your configs I won't be able to say much... |
| 19:26.39 | exarv | codefreeze: http://pastebin.ca/742720 |
| 19:27.36 | terrymr | I'm trying to get a PRI going here ... all my incoming calls appear to get hung up with a cause of 6 - channel unacceptable - anybody seen this before ? |
| 19:27.58 | mcab | hmmhesays: that'll happen if the phone is reformatted, then can't find its bootserver |
| 19:28.31 | hmmhesays | but after that it should boot fine right? |
| 19:28.37 | Shaun222 | [TK]D-Fender: here's the phone.cfg http://pastebin.ca/742723 |
| 19:28.40 | hmmhesays | bah, not the system clock is 6 hours off |
| 19:28.44 | [TK]D-Fender | hmmhesays: assuming your configs aren't screwed up |
| 19:28.53 | Shaun222 | sip.cfg is the default one that comes with the polycom's nothing changed in that... |
| 19:28.59 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 19:29.01 | hmmhesays | [TK]D-Fender: it boots fine when it grabs the config off the server |
| 19:29.52 | [TK]D-Fender | Shaun222: sip.conf please |
| 19:29.53 | mcab | hmmhesays: when you reformat the phone, it loses it's app and configs - if it can't contact the bootserver after that there's nothing much it can do except keep rebooting and retrying... |
| 19:30.11 | hmmhesays | mcab, but it grabs the app, loads it and the phone works |
| 19:30.20 | hmmhesays | then upon reboot it freaks out again |
| 19:30.39 | mcab | hmmhesays: hmmm |
| 19:30.43 | djMax | how do I figure out if these modules in /usr/lib/asterisk/modules are ok? There are about 30 of them |
| 19:30.46 | exarv | codefreeze: also, when I dial to the IVR, and the php script is still running (so after the forkcdr, and resetcdr) and no dialout has happened yet. And I hangup the phone, I get no cdr's at all. |
| 19:31.08 | mcab | hmmhesays: can you pastebin a <mac>-boot.log and <mac>-app.log? |
| 19:31.36 | Shaun222 | [TK]D-Fender: http://pastebin.ca/742726 |
| 19:31.52 | peanut- | if I have extensions 100 and 200, shouldn't I be able to call them directly from eachother? when I dial 100 from 200 it gives me a 404 |
| 19:32.24 | [TK]D-Fender | Shaun222: sip.conf please <--- |
| 19:32.53 | jfitzgibbon | peanut-: only if 200's context contains a 100 extension |
| 19:32.55 | *** part/#asterisk exvito (n=exvito@195.245.132.93) |
| 19:33.13 | Shaun222 | [TK]D-Fender: whats the sip.conf have to do with the phone settings? |
| 19:33.40 | [TK]D-Fender | Shaun222: You have some inconsistencies in your phone setup I want to verify against it |
| 19:33.55 | djMax | It really looks like "app_eval.so" is gone, but not sure how to verify |
| 19:34.04 | hmmhesays | http://www.pastebin.ca/742728 |
| 19:34.27 | ManxPower | djMax: when you do a "make install" it TELLS YOU THE MODULES THAT IT SEES INSTALLED THAT WERE NOT PART OF YOUR CURRENT INSTALL. |
| 19:34.38 | djMax | yes, I see that. There are 30 of them. |
| 19:34.55 | Shaun222 | [TK]D-Fender: http://www.pastebin.ca/742729 |
| 19:34.57 | djMax | but I have no idea what to DO about that, since I didn't put them there in the first place (I assume on old * did) |
| 19:35.02 | ManxPower | djMax: then remove all 30 unless you have some custom ones like G729 or other non-standard modules |
| 19:35.03 | hmmhesays | ok, any reason this 501 would be 6 hours off on the timeserver? |
| 19:35.19 | ManxPower | hmmhesays: wrong timezone, of course. |
| 19:35.45 | [TK]D-Fender | Shaun222: # allow=g722,ulaw <- only 1 per line, and G.722 only works in passthrough right now IIRC... I would advise ulaw |
| 19:35.47 | hmmhesays | hrm, on the poly? |
| 19:36.01 | lirakis | i set "createlink=no" in /etc/asterisk/agents.conf ... but my userfield keeps getting appended with the recording file name... i dont want it to. :( i dont know where else it would be set to do this |
| 19:36.09 | ManxPower | I have 3 non-standard modules on most of my systems, app_rxfax.so, app_txfax.so, and app_nvfaxdetect.so |
| 19:36.10 | Shaun222 | [TK]D-Fender: ok |
| 19:36.16 | hmmhesays | yeah my gmt offset was not set |
| 19:36.27 | [TK]D-Fender | Shaun222: Next, PB up an "ls -l" of your provisioning folder. |
| 19:36.53 | peanut- | jfitzgibbon: I can't even dial 100 from 100, it worked before, I borked something |
| 19:37.01 | Shaun222 | so use ulaw only with the polycom 550's? |
| 19:37.07 | mcab | hmmhesays: got a <mac>-app.log too? |
| 19:37.23 | lirakis | does asterisk require a restart to take the "createlink" setting in agents.conf into effect.. or is just a reload? ... it doesnt seem to have worked with a reload .. unless it is also set some where else |
| 19:37.56 | hmmhesays | I do |
| 19:38.08 | Shaun222 | [TK]D-Fender: my provisioning folder? the folder with my phone.cfg and stuff i'm assuming. |
| 19:38.19 | jfitzgibbon | peanut-: pastebin your sip.conf and extensions.conf if you want any specific advice |
| 19:38.27 | [TK]D-Fender | Shaun222: yes |
| 19:38.30 | Shaun222 | [root@pbx1 extn222]# ls |
| 19:38.30 | Shaun222 | 000000000000.cfg 0004f213d61d-app.log 0004f213d61d-boot.log phone.cfg sip.cfg sip.ld sip.ver |
| 19:38.34 | Shaun222 | there you go. |
| 19:38.52 | [TK]D-Fender | Shaun222: should ahve a <mac>.cfg to point to your config files. |
| 19:39.07 | Shaun222 | [TK]D-Fender: Fri Oct 19 12:17:50 2007 1 68.4.127.67 376 /home/extn222/000000000000.cfg b _ o r extn222 ftp 0 * c |
| 19:39.12 | Shaun222 | it downloads the default one just fine |
| 19:39.16 | [TK]D-Fender | Shaun222: copy 00000000.cfg to 0004f213d61d.cfg |
| 19:39.39 | [TK]D-Fender | Shaun222: and PB its contents |
| 19:39.50 | Shaun222 | i'll try it.. |
| 19:40.00 | hmmhesays | http://www.pastebin.ca/742736 |
| 19:40.17 | codefreeze | exarv: whats DIALPARAMS, 2, and 3? |
| 19:40.30 | Shaun222 | [TK]D-Fender: http://www.pastebin.ca/742738 |
| 19:41.22 | [TK]D-Fender | Shaun222: Looking... |
| 19:42.33 | [TK]D-Fender | Shaun222: What do you see on the first line-key currently? |
| 19:43.00 | Shaun222 | well it says Line 1 |
| 19:43.14 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 19:43.18 | mcab | hmmhesays: I don't think that was the link you thought it was :-) |
| 19:43.20 | Shaun222 | when i look in the phone's software under sip->line1 i see the old extention which was 302 |
| 19:43.22 | exarv | aha, sorry.. that would ZAP/G1/0031... with the dialout number. of SIP/Supplier/dialoutnumber. |
| 19:43.23 | Shaun222 | and not 222 |
| 19:43.26 | Shaun222 | which is what it should be |
| 19:43.35 | *** join/#asterisk nybble (n=jhurley@about/apple/performa/nybble) |
| 19:44.13 | [TK]D-Fender | Shaun222: humour me and copy the <mac>.cfg as I requested and reboot... |
| 19:44.26 | exarv | codefreeze: it's a value taken from the database. |
| 19:44.32 | Shaun222 | [TK]D-Fender: doing that now actually |
| 19:44.57 | Shaun222 | same.. shows the address as 302 |
| 19:45.11 | [TK]D-Fender | Hrm |
| 19:45.11 | Shaun222 | oh wait |
| 19:45.12 | exarv | and those are three 'dial parameter' for one supplier. 1 main dialout, and 2 for fallback (didn't include the fallback stuff in the macro... |
| 19:45.19 | Shaun222 | it downloaded 0000000000 again... |
| 19:45.24 | Shaun222 | and didnt try the mac |
| 19:45.31 | Shaun222 | let me move the 00000... |
| 19:45.32 | [TK]D-Fender | Shaun222: flush the logs, reboot, and then pastebin the logs |
| 19:45.36 | [TK]D-Fender | Shaun222: a HA |
| 19:45.50 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 19:46.39 | Shaun222 | hmm |
| 19:46.43 | exarv | codefreeze: have to go now. i'll try to contact you later... ok? |
| 19:47.03 | *** join/#asterisk DrNelly (n=nfl@88-97-15-221.dsl.zen.co.uk) |
| 19:47.16 | codefreeze | ok-- right for now, I'll file this as a scenario in the stuff I'm doing... |
| 19:47.44 | nybble | (i know this isnt the asterisknow room)... But if someone knows, Is TFTP server enabled by default in asterisknow beta6? |
| 19:47.48 | codefreeze | I'm still working on a set of xfer scenarios that deeperror sent me a week ago |
| 19:47.51 | exarv | codefreeze: thnx. if you need anymore info, just ask.. robert at exa-omicron.nl |
| 19:47.58 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 19:48.06 | codefreeze | exarv: will do |
| 19:48.08 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:48.21 | exarv | codefreeze: thnx.. see you later! bye |
| 19:48.46 | DrNelly | hi, can anybody help me setup a Grandstream 503 FXO port, I cant get the CalledID to be sent to asterisk. |
| 19:50.33 | Shaun222 | [TK]D-Fender: same.. |
| 19:50.37 | Shaun222 | one sec getting the logs |
| 19:50.42 | [TK]D-Fender | Shaun222: flush the logs, reboot, and then pastebin the logs <- |
| 19:50.46 | peanut- | can you not set your CPN with SIP? |
| 19:50.50 | peanut- | is that only an IAX2 thing? |
| 19:51.32 | DrNelly | peanut, it keeps on sending the ID from the sip.conf file and not the real caller id |
| 19:51.48 | peanut- | what id |
| 19:51.58 | DrNelly | userid |
| 19:52.19 | hmmhesays | tcpIpApp.sntp.gmtOffset="-6" is there anything else I have to change in my config to get this thing to sync right |
| 19:52.25 | hmmhesays | now it is applying gmt only |
| 19:52.59 | Shaun222 | [TK]D-Fender: http://www.pastebin.ca/742749 |
| 19:53.14 | peanut- | userid as cid? |
| 19:53.38 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 19:53.38 | djMax | ok, so I took the leap. I don't seem to have zap channels anymore? |
| 19:53.39 | DrNelly | no, I get [myuserid] |
| 19:54.02 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 19:54.05 | peanut- | ... |
| 19:55.33 | [TK]D-Fender | Shaun222: 1019194807|cfg |3|02|Edit|Loaded local file: /ffs0/local/0004f213d61d-phone_cfg.zzz |
| 19:55.50 | djMax | "no channel type registered for Zap"? |
| 19:55.52 | [TK]D-Fender | Shaun222: I think your phone has a local config but no auth to UPLOAD it to your provisioning folder. |
| 19:56.06 | [TK]D-Fender | Shaun222: This may contain some overrides which would explain your problem. |
| 19:56.12 | Shaun222 | how can i fix that |
| 19:56.25 | Shaun222 | do i just need to reset to fact defaults? |
| 19:56.27 | Shaun222 | cuz that sucks. |
| 19:56.28 | Shaun222 | :) |
| 19:56.31 | [TK]D-Fender | Shaun222: one way to verify this is to add a directory entry on the phone and then soft-reboot it. It should upload the directory file. |
| 19:56.47 | peanut- | why does IAX2 suck so much with voicepulse? SIP is flawless when I connect with that |
| 19:56.59 | [TK]D-Fender | Shaun222: Factory defaults would probably not be a bad idea (regardless) |
| 19:57.18 | [TK]D-Fender | Shaun222: Actually... I'm iffy on perms.... you DO have logs... |
| 19:57.29 | [TK]D-Fender | Shaun222: Factory Reset them... |
| 19:57.32 | Shaun222 | k |
| 19:57.52 | [TK]D-Fender | peanut-: Because |
| 19:57.52 | Shaun222 | oh sweet this phone has a way better reset menu |
| 19:57.56 | Shaun222 | i can reset the configs only |
| 19:58.22 | Shaun222 | local configs anyway... hope it keeps my ftp settings. |
| 19:58.27 | [TK]D-Fender | Shaun222: "local config" |
| 19:58.35 | [TK]D-Fender | Shaun222: Yes, indeed Polycom = really nice |
| 19:58.56 | peanut- | [TK]D-Fender: do all iax2 providers suck or just voicepulse? |
| 19:59.09 | Shaun222 | i dont remember that on the 601's |
| 19:59.17 | [TK]D-Fender | peanut-: IAX2 has potential "issues", but I'll START with VoicePulse |
| 19:59.21 | Shaun222 | i've been buying 550's lately |
| 19:59.42 | [TK]D-Fender | Shaun222: 550 = kinda waste unless you have no control over lighting.... |
| 19:59.48 | peanut- | dissapointing |
| 19:59.53 | *** part/#asterisk terrymr (n=terrymr@192.220.217.189) |
| 20:00.14 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:00.18 | Alowishus | anyone dealt with the D-Link DIV-140 FXO gateway? |
| 20:00.31 | DrNelly | peanut, I have been using Gradwell.net for uk IAX2 provider and no no issues. |
| 20:00.37 | knarfly | how does one clear out the log files in asterisk...if I simply erase them will that do or is there a method for this? |
| 20:00.46 | Shaun222 | [TK]D-Fender: voice quality seams better to me, when my buddy had sombody on speaker one day i walked in thinking the guy was actually in the room... Lighting was the big go getter with them though. |
| 20:00.52 | Shaun222 | at night it's hard to see who's calling |
| 20:00.57 | Shaun222 | or when dark i should say |
| 20:01.11 | Shaun222 | for my night guys they like them much better. |
| 20:01.22 | [TK]D-Fender | Shaun222: Well if its a MUST... its a nasty price premium though... sad really... |
| 20:01.28 | knarfly | ' /var/log/asterisk/cdr-csv/master.CSV can I just erase this and let * start over? |
| 20:01.44 | Shaun222 | ok.. well resetting the configs looks to have fixed the problem... at least it pulled the new info... who's to say if i make a change if it will take it or if i'll have to reset it again |
| 20:02.10 | codefreeze | knarfly: should be able to... if you are scared do a cat /dev/null > /var/log/ast/cdr/mas... instead |
| 20:02.10 | Shaun222 | [TK]D-Fender: cant remember what i paid for the other ones i got but i just bought two of them new for 400 total |
| 20:02.51 | Shaun222 | ebay style though. |
| 20:03.14 | [TK]D-Fender | Shaun222: EW |
| 20:03.22 | [TK]D-Fender | OH... 200$ each |
| 20:03.28 | Shaun222 | ya |
| 20:03.28 | [TK]D-Fender | Shaun222: still :/ ebay |
| 20:03.31 | djMax | so my main purpose in upgrading was imap vm storage. Will this pull existing messages into the imap server? |
| 20:03.40 | Shaun222 | for 2, still in plastic |
| 20:04.08 | Shaun222 | but ya, not cheap phones |
| 20:04.13 | Shaun222 | they do look nicer to me though |
| 20:04.25 | Shaun222 | the older polycom's look like kid toys.. |
| 20:04.36 | Shaun222 | one thing i hated when i switched away from the cisco's. |
| 20:04.40 | Shaun222 | cisco's looked so nice |
| 20:05.01 | [TK]D-Fender | Shaun222: If you consider fake-silver/gray "adult", sure :) |
| 20:05.06 | Shaun222 | haha |
| 20:05.11 | [TK]D-Fender | Shaun222: That aside its a 601 body still :) |
| 20:05.15 | Shaun222 | hey... i said nicer... :) |
| 20:05.22 | Shaun222 | ya it is. |
| 20:05.35 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:05.40 | Shaun222 | the silver must just give off some contrast or somthing.. dunno. |
| 20:05.41 | [TK]D-Fender | So, hows the reboot? |
| 20:05.52 | peanut- | SIP quality on voicepulse is great though |
| 20:05.55 | Shaun222 | it worked, pulled new info. |
| 20:05.58 | peanut- | and this WIP300 works well |
| 20:06.04 | [TK]D-Fender | Shaun222: I think the newer LCD & backlight help bring it out. |
| 20:06.19 | Shaun222 | i just dont know if it's going to accept a change in the future or if i'm going ot have to keep resetting them |
| 20:06.26 | Shaun222 | i disabled the web interface in my config this time... |
| 20:06.40 | Shaun222 | so nobody should be messin around with that |
| 20:06.41 | [TK]D-Fender | Shaun222: Good.. they should have removed it ages ago |
| 20:07.03 | Shaun222 | ya, or it should be disabled by default or somthing |
| 20:07.04 | djMax | sounds like imap vm isn't really ready for primetime, is this accurate? |
| 20:07.07 | Shaun222 | lock it down.. |
| 20:07.55 | Shaun222 | [TK]D-Fender: thanks for the help. I'm going to to move on to my next task :) |
| 20:10.18 | [TK]D-Fender | Shaun222: Is it all good now? |
| 20:10.31 | [TK]D-Fender | Shaun222: Ah, I see. Good to hear. |
| 20:10.50 | Shaun222 | [TK]D-Fender: yes, it's good. |
| 20:11.26 | [TK]D-Fender | Shaun222: you should remove the 000000000.cfg file, and rename phone.cfg to phone222.cfg and update <mac>.cfg to point to that |
| 20:11.40 | [TK]D-Fender | Shaun222: Generic ass configs = mistake |
| 20:11.43 | [TK]D-Fender | :) |
| 20:12.15 | Shaun222 | well each phone logs into ftp as a seperate user, but i get what your saying. |
| 20:12.35 | [TK]D-Fender | Shaun222: typically I like to load the server IP into sip.cfg as well and only override it on the phoneXXXX.cfg as needed for remote phones, etc |
| 20:12.43 | [TK]D-Fender | Shaun222: I suppose... |
| 20:13.01 | [TK]D-Fender | Shaun222: Kind of a mixed bag, but you seem to have a handle on most of this. |
| 20:13.17 | [TK]D-Fender | Shaun222: rather refreshing to see actually... |
| 20:13.33 | Shaun222 | thats not a bad idea, would make ip changes easy. |
| 20:13.41 | [TK]D-Fender | Shaun222: too many complete numbskulls out there wasting quality phones :) |
| 20:13.54 | Shaun222 | would probably be a better idea for me to hard code the ip in the configs rather than a name.. in case dns fails |
| 20:14.37 | [TK]D-Fender | Shaun222: if your IP is fixed you're far better off... |
| 20:15.07 | peanut- | [TK]D-Fender: can you not set CPN in SIP like you can in IAX2? |
| 20:15.32 | knarfly | anyone else running * on FreeBSD ? |
| 20:15.33 | [TK]D-Fender | peanut-: CPN = CPID? |
| 20:15.56 | Shaun222 | ya it's fixed. |
| 20:16.38 | peanut- | [TK]D-Fender: CPID? |
| 20:16.46 | [TK]D-Fender | Shaun222: I'd say pull the host then unless you're planning on changing it any time soon. |
| 20:17.19 | Shaun222 | ya, i'm going to do that... i dont plan on moving it, no need to i own the ip space so it moves with me :) |
| 20:17.24 | [TK]D-Fender | peanut-: Called Party ID. AKA : you call someone, system looks up THEIR name and displays on your screen so you know what that exten represents. Is this what you are referring to? |
| 20:17.43 | peanut- | exten => _1NXXNXXXXXX,1,Set(CALLERID(num)=5555555555) |
| 20:18.07 | Qwell | like I said... CPN is incredibly ambiguous |
| 20:18.23 | peanut- | when I make calls with IAX2 it forwards CPN correctly, with SIP it comes up as unknown |
| 20:18.36 | peanut- | so it's probably sending nothing or something invalid |
| 20:18.46 | Qwell | [TK]D-Fender: apparently CPN == CID num |
| 20:19.18 | peanut- | Calling Party Number |
| 20:21.23 | [TK]D-Fender | peanut-: WHERE is it coming up unknown? |
| 20:22.59 | *** join/#asterisk objective (n=objectiv@ool-43536c3b.dyn.optonline.net) |
| 20:23.21 | peanut- | when I call my cell, that previously reported whatever number I put in extensions.conf, comes up unknown when I call over the SIP trunk |
| 20:24.01 | [TK]D-Fender | peanut-: what "sip trunk"? |
| 20:24.15 | peanut- | teh sip connection to voicepulse |
| 20:24.18 | peanut- | instead of iax |
| 20:24.39 | [TK]D-Fender | peanut-: And have you seen anything confirming that they even permit you to set your CID? |
| 20:25.00 | objective | they do |
| 20:25.08 | peanut- | they do with IAX2 |
| 20:25.22 | objective | they do with both |
| 20:25.28 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 20:25.32 | peanut- | how do you do it with SIP? |
| 20:25.51 | [TK]D-Fender | peanut-: something else must be amiss. pastebin your sip peer enty masking onlyt he password |
| 20:26.03 | *** part/#asterisk knarfly (n=vladimir@adsl-11-248-246.mia.bellsouth.net) |
| 20:27.29 | *** join/#asterisk ltd (n=z@nox.amused.net) |
| 20:27.30 | peanut- | why must something be amiss? |
| 20:27.35 | *** part/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 20:27.42 | [TK]D-Fender | peanut-: well its supposed to be working and it isn't. |
| 20:27.55 | [TK]D-Fender | peanut-: So lets see if something is wrong with your peer setup |
| 20:28.27 | jcanfield | anyone else get an email from asterisk-users about excessive bounces? I wonder if something is wrong with the list? |
| 20:29.00 | champster | Primary D-Channel on span 2 up??? |
| 20:29.00 | champster | I have an Panasonic DBS 576 that I am using like a mux for my fax machines. |
| 20:29.00 | champster | It is using a PRI cable. |
| 20:29.00 | champster | If the Panasonic goes down, Asterisk says Primary D-Channel on span 2 up / down, etc. |
| 20:29.00 | champster | How do I get the span to resync? shouldn't unpugging the cable give it a fresh start? |
| 20:29.01 | champster | Currently, to fix this, I have to stop asterisk and Zaptel, and restart them. |
| 20:29.03 | champster | Please advise if there is a way to recover 1 span. |
| 20:29.29 | peanut- | my sip peer config is just fine |
| 20:29.37 | peanut- | everything works except CID |
| 20:29.48 | ManxPower | champster: it should restart on it's own when the line to the panasonic comes back |
| 20:29.58 | [TK]D-Fender | peanut-: fine go and assume that nothing in there could possibly interfere.... |
| 20:29.59 | [TK]D-Fender | ~assume |
| 20:30.00 | jbot | About assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
| 20:30.12 | peanut- | for SIP clients in sip.conf, is callerid=<1232343456> supposed to set it? |
| 20:30.13 | [TK]D-Fender | Anyways, I've got to be off... back substantially later... |
| 20:30.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:30.25 | champster | IT doesn't and hasn't |
| 20:30.26 | ManxPower | peanut-: yes |
| 20:30.30 | Shaun222 | [TK]D-Fender: later! |
| 20:30.43 | [TK]D-Fender | peanut-: and you did not set a NAME... perhaps thats part of it.. |
| 20:30.44 | ManxPower | champster: then I suspect the panasonic is confuzing asterisk/zaptel so badly that it cannot recover |
| 20:30.52 | [TK]D-Fender | Shaun222: You're welcome, and good luck. |
| 20:31.17 | champster | IT would be nice if there was a command to down a line or span. |
| 20:31.26 | peanut- | ManxPower: just set to a 10-digit? |
| 20:31.30 | objective | peanut- : why don't you just call voicepulse? they'll be happy to show you how to configure it |
| 20:31.31 | ManxPower | champster: people have been asking that for YEARS |
| 20:31.46 | ManxPower | peanut-: well "1" is not a valid leading callerid number |
| 20:32.04 | peanut- | 5125555555 |
| 20:32.06 | l2trace99 | any one usings queues ? |
| 20:32.12 | ManxPower | peanut-: correct. |
| 20:32.21 | champster | Luckily the rest of the system stays up when one span goes down. |
| 20:32.23 | peanut- | so assuming it's a valid number, it should work and forward through to outside phones you call |
| 20:32.26 | roxlu | When I just want to test my fresh asterisk install with a softphone, do I need to read the part about zapata.conf in the asterisk book? |
| 20:32.51 | ManxPower | peanut-: assuming your carrier permits that, yes |
| 20:32.54 | ManxPower | roxlu: no |
| 20:33.01 | roxlu | thanks |
| 20:33.33 | Shaun222 | voicepulse needs to add support for outgoing CIN |
| 20:34.08 | Shaun222 | actually they need to add support for in and out.. |
| 20:35.03 | roxlu | When I do: dialplan reload, I see this: http://paste-it.net/4026 |
| 20:35.58 | peanut- | actually they need IAX2 to not suck balls. |
| 20:36.13 | Shaun222 | whats wrong with there IAX2? |
| 20:36.17 | Shaun222 | works fine for me.. |
| 20:36.26 | objective | IAX2 is not their problem... IAX2 just doesn't scale at all for carriers |
| 20:36.53 | Shaun222 | they could get some better bw |
| 20:37.13 | Shaun222 | i'm tired of seeing problems with sprint.. |
| 20:38.26 | peanut- | it does? when I use AIX2 to connect to them instead of SIP the voice cuts out alot |
| 20:38.33 | peanut- | *IAX2 |
| 20:39.08 | objective | if there isn't a compelling reason for you to use IAX, you should just use SIP |
| 20:39.37 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 20:39.43 | rantsh | hia all |
| 20:40.04 | objective | and send them an email about the callerid issue, they'll figure it out... they're one of the only itsp's that actually answer the phone and emails |
| 20:40.15 | rantsh | I gotta quick question, I know this might not be the place to ask this but, I trust you guys, and no one else |
| 20:40.37 | rantsh | anyone knows how I can make a ser server accept a call for *2? |
| 20:40.38 | peanut- | Shaun222: did you change anything in iax.conf from their template they provided? |
| 20:43.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 20:45.29 | rantsh | if not anyone knows where I can find people who knows about SER? |
| 20:46.34 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 20:49.26 | Shaun222 | peanut-: i didnt use there iax.conf |
| 20:49.35 | Shaun222 | i stripped out the crap i needed. |
| 20:49.52 | Shaun222 | so it's simular.. |
| 20:52.07 | *** join/#asterisk neax (n=newdle@203-114-176-86.dsl.sta.inspire.net.nz) |
| 20:54.23 | roxlu | I've read through chapter 4 of the book, ... but nowhere is described how to create the test call?? how can i test my config? |
| 20:55.50 | lirakis | later all |
| 20:55.52 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:58.18 | peanut- | Shaun222: yea I should have done that, I found the problem, it needed tos=ef |
| 20:58.55 | peanut- | apparently time warner recognizes RTP as realtime but not IAX2 |
| 20:59.18 | peanut- | now it's crystal clear on IAX2 |
| 20:59.22 | ManxPower | tos=ef should not be valid |
| 20:59.32 | peanut- | whynot |
| 20:59.45 | ManxPower | because of you want DSCP EF then the TOS would be 0xb8 |
| 21:00.42 | peanut- | it's not set to 0xef it's set to ef |
| 21:01.16 | ManxPower | I was not aware Asterisk's TOS supported anything except for hex and I was not aware that it supported DSCP, only IP TOS. |
| 21:01.21 | ManxPower | maybe it is something added in 1.4 |
| 21:02.08 | ManxPower | As far as I can tell IP TOS 0xb8 is the same as DSCP EF |
| 21:02.39 | ManxPower | remember, asterisk will NOT throw an error if you have invalid options in it's config files. |
| 21:02.56 | ManxPower | so you could set screwmicrosoft=yes in sip.conf and it would not generate an error. |
| 21:03.12 | Qwell | it would warn you |
| 21:04.25 | Shaun222 | whats regexten used for in the sip.conf? |
| 21:04.59 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 21:08.08 | hmmhesays | hey microsoft has made a lot of people a lot of money |
| 21:08.15 | hmmhesays | (those of use that can fix things) |
| 21:08.57 | mvanbaak | Shaun222: for sip peers registering with asterisk |
| 21:09.13 | mvanbaak | regcontext + regexten |
| 21:09.25 | roxlu | can someone help me please with this? http://paste-it.net/4027 |
| 21:09.28 | mvanbaak | for example, set regcontext to [registered-sip] |
| 21:09.31 | mvanbaak | erm |
| 21:09.41 | mvanbaak | set it to: regcontext=registered-sip |
| 21:09.51 | roxlu | I've created a basic config, like the book describes.. when I do show peers I see my softphone info |
| 21:09.57 | mvanbaak | for every sip peer set: regexten=<peer number> |
| 21:10.22 | mvanbaak | as soon as a sip peer registeres it will automagically insert an exten in the context 'registered-sip' |
| 21:10.27 | mvanbaak | sample: |
| 21:10.32 | mvanbaak | you have a peer called 1000 |
| 21:10.41 | mvanbaak | with regexten set to 1000 |
| 21:10.54 | mvanbaak | in global you set regcontext=registered-sip |
| 21:10.54 | roxlu | though when I call "1" I get a message "Nobody on the line.. (or something like that)" |
| 21:11.10 | mvanbaak | as soon as sip peer 1000 registeres with asterisk it will insert: |
| 21:11.16 | mvanbaak | [registered-sip] |
| 21:11.18 | roxlu | mvanbaak: are you talking to me? |
| 21:11.26 | mvanbaak | exten => 1000,1,NoOp() |
| 21:11.34 | mvanbaak | I'm talking to Shaun222 |
| 21:11.45 | roxlu | oh :D sorry (i had a 1000 as well) |
| 21:11.53 | Shaun222 | i'm not really understanding either.. |
| 21:12.22 | mvanbaak | Shaun222: ok |
| 21:12.24 | roxlu | if someone could have a look at my config.. please have a loot at http://paste-it.net/4027 |
| 21:12.27 | mvanbaak | you start asterisk |
| 21:12.35 | Shaun222 | sounds like it's for tracking which sip devices are alive? |
| 21:12.52 | mvanbaak | Shaun222: it's for tracking which sip devices are registered |
| 21:13.07 | Shaun222 | ok, right thats what i was saying. |
| 21:13.31 | mvanbaak | it handy specially in combination with dundi |
| 21:13.50 | hmmhesays | uck dundi |
| 21:14.00 | mvanbaak | hmmhesays: I like dundi |
| 21:14.48 | Shaun222 | hmm, voicepulse's configs now show connect 3 and then 2 |
| 21:14.52 | Shaun222 | i'm using 1 and 2 |
| 21:15.52 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-64aa4004af1e9a4a) |
| 21:16.05 | Shaun222 | whats with this tos in the iax.conf.. |
| 21:16.07 | Shaun222 | i dont have that. |
| 21:19.59 | roxlu | When I type "sip reload" on the CLI, I see 3 lines, but it doens't return back to the CLI,..... I have to press [enter], is this correct? |
| 21:20.10 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 21:21.32 | mvanbaak | roxlu: yeah, that can happen |
| 21:21.37 | mvanbaak | I have it too sometimes |
| 21:22.03 | roxlu | okay |
| 21:22.16 | roxlu | but... I see, I need to call 1000 to test it... |
| 21:23.16 | roxlu | the book definitely should add that in chapter 4... |
| 21:23.16 | peanut- | ManxPower: apparently it's new then, in iax.conf.sample it has ;tos=ef |
| 21:23.30 | peanut- | and listed all the other ToS, and they wren't in hex |
| 21:28.36 | *** join/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 21:32.21 | *** part/#asterisk xlyz (n=xz@host-84-223-120-104.cust-adsl.tiscali.it) |
| 21:35.59 | roxlu | When I have this config http://paste-it.net/4028 |
| 21:36.28 | Shaun222 | anybody know how the messages button on the polycom's is suppose to work... looks like it calls itself.. |
| 21:36.42 | roxlu | and I call to 1000 (using x-lite) I get a nice music... but how can I test the 1,2,3,4,5,6 extens I have defined?? |
| 21:37.03 | *** join/#asterisk newbie`` (i=nouser@117.102.56.98) |
| 21:37.49 | ReDNeQ | sup |
| 21:38.37 | mvanbaak | I dont se where you have music defined roxlu |
| 21:38.54 | roxlu | mvanbaak: me neither :( |
| 21:39.09 | roxlu | only the Playback weasels |
| 21:39.10 | mvanbaak | roxlu: must be more in your extensions.conf |
| 21:39.16 | *** join/#asterisk kavelot (i=x@201-68-27-8.dsl.telesp.net.br) |
| 21:39.41 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 21:39.47 | roxlu | mvanbaak: that paste **is** from my extensions.conf ?? |
| 21:40.28 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
| 21:40.41 | kavelot | I have no phones connected (yet) to Asterisk... I'm just testing how it handles calls now... so far I tested, it's working, but I hear no sound... if I do Answer, Wait(5), Hangup I notice it waits 5s before the busy signal, but if I try to play something, logs show it's playing, but it doesn't play... any hints? |
| 21:40.46 | kavelot | do I need zaptel for that? |
| 21:40.56 | mvanbaak | roxlu: then I'm lost |
| 21:41.04 | mvanbaak | you need to paste cli output |
| 21:41.16 | roxlu | ok |
| 21:41.32 | roxlu | i'll call again to 1000 (using th elogged in user 1000) and paste it |
| 21:41.45 | mvanbaak | kavelot: no, you dont need zaptel to do a playback |
| 21:42.15 | l2cache | How would I set a variable have the ${EXTEN} string in it? I want the dst variable to have '${EXTEN}@example' in it |
| 21:42.17 | kavelot | weird, because that's a basic installation of Trixbox, I didn't change much except extensions.conf and sip.conf |
| 21:43.32 | roxlu | mvanbaak: I pasted it here: http://paste-it.net/4029 together with my config |
| 21:44.11 | Shaun222 | when a sip phone dials another extension, is that extension stored somewhere i can call back.. like FROMEXTEN or somthing |
| 21:44.39 | l2cache | anyone know how i could do a set command to put the STRING of ${EXTEN} in a variable? |
| 21:44.41 | mvanbaak | roxlu: are you sure you hear music |
| 21:44.52 | roxlu | not anymore..... with that pasted config |
| 21:44.53 | mvanbaak | or is it just the sound of a circular phone system |
| 21:44.57 | l2cache | not the exten variable, just that specific text |
| 21:45.04 | roxlu | but I removed some thing |
| 21:45.43 | roxlu | mvanbaak: but how does this generally work? .. I logged in with the user 1000, than I call 1000.. but what can I do than? |
| 21:47.13 | roxlu | mvanbaak: ? |
| 21:47.40 | mvanbaak | what context has the sip peer 1000 have ? |
| 21:48.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:49.37 | roxlu | mvanbaak: this is my config http://paste-it.net/4030 |
| 21:49.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:50.20 | mvanbaak | ok |
| 21:50.27 | roxlu | is that one correct? |
| 21:50.35 | mvanbaak | so if you call 1000 from a softphone configured as 1000 you will call yourself |
| 21:50.44 | roxlu | yes |
| 21:50.48 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.33) |
| 21:50.55 | mvanbaak | sounds pretty useless to me |
| 21:51.03 | mvanbaak | I can talk to myself even without a phone |
| 21:51.12 | roxlu | but thats just for testing purposes? |
| 21:51.13 | peanut- | haha |
| 21:51.30 | roxlu | How can I test it else? |
| 21:51.48 | peanut- | ok this WIP300 wifi phone isn't as horrible as everyone says |
| 21:51.54 | peanut- | sound quality is excellent |
| 21:52.04 | peanut- | ~wifi |
| 21:52.04 | jbot | well, wifi is see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 21:52.42 | roxlu | mvanbaak: do I need to create another sip file? |
| 21:52.43 | mvanbaak | roxlu: use the echo app |
| 21:52.47 | roxlu | sip entry.. |
| 21:52.54 | roxlu | oh where is that? / how does that work |
| 21:53.15 | mvanbaak | in asterisk cli: show application Echo |
| 21:53.58 | roxlu | okay done that (i see some nice purple lines) |
| 21:54.03 | roxlu | .. with green |
| 21:57.44 | l2cache | anyone know how i could do a set command to put the STRING of ${EXTEN} in a variable? |
| 22:00.29 | kimo_sabe | l2cache: a variable like ${EXTEN}? |
| 22:00.40 | l2cache | the string of ${EXTEN} not its contents |
| 22:00.47 | l2cache | into a variable |
| 22:01.18 | l2cache | i basically need to put Set(dev=${EXTEN}@sippeer1) but i need the exten to show up as text |
| 22:01.26 | l2cache | so when I call it it functions |
| 22:02.18 | l2cache | im putting that variable into the asterisk DB. So i want it to have the ${EXTEN} text when I call it |
| 22:04.23 | l2cache | Anyone? |
| 22:07.27 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 22:07.48 | l2cache | Nevermind,I needed to put the text "${EXTEN}" into the asterisk DB. But i will just execute a system command from the dialplan with a asterisk -rx 'database put blah blah blah' |
| 22:08.13 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:09.14 | kimo_sabe | l2cache: I'm curious, why are you trying to store a literal "${EXTEN}" in the asterisk db? |
| 22:09.39 | l2cache | Because I need to call it later for a call forwarding app |
| 22:09.50 | l2cache | the app will do Set(dbdial=${DB(forward/${EXTEN})}) |
| 22:10.12 | l2cache | then the next priority will be Dial(SIP/${dbdial}) |
| 22:10.41 | kimo_sabe | l2cache: why can't the forwarding app just expand the ${EXTEN}, and you just store it's contents? |
| 22:11.09 | l2cache | the three options in the DB for forwarding are 8002212452@provider......................${EXTEN}@sippeer and ${EXTEN}@sippeer |
| 22:11.27 | l2cache | so i need to have the literal text "${EXTEN}" in the DB for the dial to work later on |
| 22:11.33 | l2cache | because all options are not the same formatting |
| 22:12.26 | l2cache | the EXTEN for when you are setting up the forwarding, and the forwarding function itself are two separate values. The value needs to be used when it is forwarding....that make sense? |
| 22:13.17 | *** join/#asterisk mltlnx (n=mltlnx@64.3.170.41.ptr.us.xo.net) |
| 22:13.18 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 22:15.59 | Lisa696 | hello |
| 22:16.22 | Lisa696 | i have a problem with the default password of frepbx... any person help me.. plz.. |
| 22:17.27 | Alowishus | is the problem that you don't know it? :) /join #freepbx |
| 22:17.49 | *** join/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 22:18.48 | l2cache | anyone know how to put a string as a variable....i want to have the contents of test be "${EXTEN}" literally.... |
| 22:19.06 | kimo_sabe | l2cache: $$ or \$ don't work to escape the $? |
| 22:22.04 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:22.13 | peanut- | <PROTECTED> |
| 22:22.48 | l2cache | checking |
| 22:22.57 | l2cache | $${EXTEN} returned just what I typed |
| 22:24.34 | l2cache | \${EXTEN} did not work |
| 22:25.42 | kimo_sabe | ${${EXTEN}:1} ... eww |
| 22:25.50 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:28.16 | Lisa696 | hello.. |
| 22:28.24 | HarryR | Howday |
| 22:28.30 | Lisa696 | [Oct 19 17:25:11] NOTICE[6668]: manager.c:1020 authenticate: 127.0.0.1 tried to authenticate with nonexistent user 'user' |
| 22:28.30 | Lisa696 | <PROTECTED> |
| 22:28.49 | Lisa696 | any person help. me??? |
| 22:29.41 | *** join/#asterisk metfan2007 (n=metfan20@201.103.41.55) |
| 22:30.53 | metfan2007 | Hi all!! I'm receiving a lot of "chan_sip.c:4105 set_destination: Can't find address for host" messages for a few Aastra phones, any idea about what does it means? |
| 22:31.42 | HarryR | metfan2007, it just means the ip address of your phone doesn't have any resolvable hostname associated |
| 22:31.48 | __freedom__lover | \quit |
| 22:32.32 | metfan2007 | HarryR, mmmm, and how can I resolve it? :S |
| 22:32.46 | HarryR | setup reverse dns for all your ip addresses |
| 22:33.01 | HarryR | or find some magic asterisk variable which disables dns checks |
| 22:33.12 | HarryR | but... i'm just guessing based on past experience and the error message |
| 22:33.14 | metfan2007 | oh.... |
| 22:33.14 | metfan2007 | ok |
| 22:33.34 | HarryR | either way it's not a big issue |
| 22:33.53 | HarryR | Lisa696, is the account setup properly? |
| 22:33.57 | metfan2007 | but the funny thing is that the message appears only in 2 phones, the other phones are Ok |
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| 22:34.11 | HarryR | if you do a dns lookup on their ip addresses, what do you get? |
| 22:34.25 | HarryR | compared to the other phones you get warnings for |
| 22:37.02 | l2cache | still did not work |
| 22:37.39 | l2cache | does anyone know how to put the string of another variable name into another variable....For example - The text ${EXTEN} needs to be in the dest variable. |
| 22:38.39 | Lisa696 | ? |
| 22:38.42 | Lisa696 | or do you understand which can be the solution before my 2 problems? |
| 22:38.42 | Lisa696 | that of the key in the web .. that does not serve me. |
| 22:38.42 | Lisa696 | or for that this message goes out in my shell? |
| 22:40.04 | Lisa696 | do not I have great idea? |
| 22:40.05 | Lisa696 | how can I check the configuration? |
| 22:40.09 | Lisa696 | HarryR? |
| 22:40.16 | l2cache | Please help |
| 22:41.53 | Lisa696 | :( |
| 22:44.05 | Shaun222 | i have both SIP and IAX extensions, i want to check to see which extension exists and then call it |
| 22:44.15 | Shaun222 | what would be the best way to do that? |
| 22:47.36 | Lisa696 | I NO HAVE IDEA. |
| 22:47.58 | Lisa696 | shaun222 you have freepbx? |
| 22:58.41 | *** join/#asterisk ggrossman (n=ggrossma@adsl-76-195-249-17.dsl.pltn13.sbcglobal.net) |
| 22:59.33 | ggrossman | hi, trying to set up an asterisk appliance to connect to junction networks which requires rsa auth. but the appliance seems to be missing /usr/lib/asterisk/res_crypto.so? |
| 23:01.32 | HarryR | ggrossman, re-compile asterisk and make sure you have crypto support enabled? |
| 23:02.02 | ggrossman | hi harry, this is an asterisk appliance from digium. it comes with a precompiled asterisk for blackfin processor. I might be able to cross-compile it myself... |
| 23:03.20 | Qwell | ggrossman: hmm, interesting |
| 23:03.58 | HarryR | I'm presuming they didn't include it because it's underpowered? |
| 23:04.10 | Qwell | no |
| 23:04.16 | ggrossman | entirely possible. emailed support but was hoping someone from digium was hanging out here |
| 23:04.41 | Qwell | I don't actually even remember why we don't include it |
| 23:05.27 | ggrossman | so I have this appliance dialing calls out successfully to jnctn.net, but they use RSA for authentication, so incoming calls aren't working due to the missing crypto support |
| 23:05.50 | Qwell | ggrossman: yeah, I see how that would be a problem :) |
| 23:06.19 | Lisa696 | http://pastebin.com/d3fc97a7a i have this problem... any person help me plz |
| 23:07.49 | Qwell | ggrossman: I'll try to remember on Monday, and I'll have to look into that. Do keep going with the support route though, so it can be properly reported and such |
| 23:07.52 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 23:08.02 | ggrossman | qwell: cool, thanks! |
| 23:09.43 | ectospasm | today I had a guy call in and try to get his IAXy to talk to his AT&T VoIP directly. He seemed very confused at the mention of Linux... |
| 23:09.56 | Qwell | ectospasm == ? |
| 23:10.47 | ectospasm | I'm a tech support monkey at Digium |
| 23:10.55 | Qwell | yeah, I figured that much :p |
| 23:11.07 | ectospasm | heheh |
| 23:11.13 | ectospasm | I'm Trey |
| 23:11.17 | Qwell | ahh |
| 23:11.25 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 23:14.13 | mosty | i have two asterisk servers, each has local sip extensions, and the two servers trunk via iax. i have setup my the sip phones to show line presences for other extensions registered to the same server, is it possible to show line presences for a sip extension on the other server? |
| 23:16.41 | *** join/#asterisk saftsack (n=saftsack@pD9E04FEA.dip.t-dialin.net) |
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| 23:17.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:22.01 | _Sam-- | jbot: seen [tk]-fender |
| 23:22.04 | jbot | i haven't seen '[tk]-fender', _Sam-- |
| 23:22.18 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 23:22.28 | _Sam-- | jbot: seen [TK]D-Fender |
| 23:22.29 | jbot | [tk]d-fender <n=joe@MTRLPQ02-1177745839.sdsl.bell.ca> was last seen on IRC in channel #asterisk, 2h 51m 37s ago, saying: 'Shaun222: You're welcome, and good luck.'. |
| 23:23.02 | Shaun222 | _Sam--: he left, he said he wouldnt be back for a while. |
| 23:23.47 | *** join/#asterisk BiG^DoG (n=stevebai@c-71-204-211-58.hsd1.de.comcast.net) |
| 23:23.57 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 23:24.24 | _Sam-- | thanks. |
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| 23:25.34 | *** join/#asterisk el_critter (n=chatzill@190.74.96.121) |
| 23:26.28 | el_critter | hi |
| 23:28.31 | el_critter | I have a VoIP provider, does my asterisk connect to theirs or it works the other way? |
| 23:29.36 | BiG^DoG | anyone here familiar with the SPA3102 configuration? |
| 23:30.35 | mosty | BiG^DoG, the basics, yes |
| 23:31.16 | BiG^DoG | All I've done is insert it in line ... not hooked it into asterisk yet |
| 23:31.26 | BiG^DoG | so PSTN line rings and it goes through to analog phone |
| 23:31.49 | BiG^DoG | normally, when someone would call and get the answering machine |
| 23:31.59 | BiG^DoG | the answering machine would disconnect if they hung up in the middle of it |
| 23:32.11 | _Sam-- | BiG^DoG : where in delaware are ya? |
| 23:32.27 | BiG^DoG | now that the SPA3102 is in line, the answering machine doesn't hang up |
| 23:32.46 | BiG^DoG | if someone hangs up during the message, the message continues and I get the loud busy signal on the machine |
| 23:32.57 | BiG^DoG | I'm sure it's some kind of line disconnect time setting I need to tweak |
| 23:33.25 | mosty | BiG^DoG, does the answering machine have a pass-through connection? |
| 23:33.58 | BiG^DoG | yes... it's telephone company to 3102 to answering machine to telephone |
| 23:34.59 | mosty | would it work putting the answering machine in front of the spa3102? |
| 23:35.29 | BiG^DoG | probably because then the PSTN would go straight to the answering machine but that's not the goal |
| 23:35.56 | TJNII | Oh... Is there an option you can throw in into sip.conf to diable call waiting? I think I saw it somewhere. |
| 23:36.02 | BiG^DoG | i'm trying to find the setting in the 3102 that I need to tweak that says if you detect a drop in line voltage of this length of time, disconnect the call |
| 23:36.30 | mosty | BiG^DoG, but the answering machine will only "pick up" if the phone doesn't, right? |
| 23:36.41 | BiG^DoG | right |
| 23:37.02 | mosty | and when the 3102/phone hangs up after leaving a message, then the answering machine should stop recording |
| 23:37.20 | BiG^DoG | correct |
| 23:37.51 | BiG^DoG | but our wonderful telemarketers hang up as soon as they hear the answering machine and then we get an annoying beep beep beep message |
| 23:38.22 | Shaun222 | i'm dialing extension to extension and for some reason i dont hear a ringing if the other extension is not availible. |
| 23:42.38 | mosty | BiG^DoG, then get a better answering machine |
| 23:42.49 | BiG^DoG | ok thanks |
| 23:42.59 | wiljacket | BiG^DoG: looks like there should be a PSTN Disconnect Detection config that lets you tweak a bunch of options on the spa3102 |
| 23:43.05 | JT | don't use an answering machine |
| 23:43.14 | JT | use voicemail if you're hooking it to asterisk |
| 23:43.32 | BiG^DoG | I haven't hooked it into asterisk yet |
| 23:43.42 | BiG^DoG | I'm easing into it |
| 23:44.01 | wiljacket | JT is right, you could also play the telemarketer-killing tone on your setup then too |
| 23:44.06 | BiG^DoG | right now I just have to get the phones to work as normal with the SPA3102 inserted inline |
| 23:44.28 | JT | doesn't work that way |
| 23:44.49 | JT | you don't share lines incoming to a pbx |
| 23:46.30 | Lisa696 | http://pastebin.ca/742979 |
| 23:46.41 | Lisa696 | i have a problem... any person help me. plz |
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