| 00:16.25 | hsv-al | crap |
| 00:16.29 | hsv-al | i missed kimbo slice fight, did he win |
| 00:16.40 | jeev | niope |
| 00:16.45 | hsv-al | ? |
| 00:25.59 | *** join/#asterisk GReggFT (n=GReggFT@fw1-p1.lexi.com) |
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| 00:55.10 | *** join/#asterisk RoyK (n=Roy@ip-105-50-149-91.dialup.ice.no) |
| 00:55.56 | shmaltz | hi everyone |
| 01:05.08 | *** join/#asterisk russellb (n=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:05.09 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 01:28.59 | *** join/#asterisk C4away (n=DJpyro@66.185.107.193) |
| 01:29.33 | C4away | was I mislead to beleive that 1.6 was going to implement t.38 origination and termination, instead of just passthrough? |
| 01:30.22 | jeev | hey c4 |
| 01:30.26 | jeev | i stillllllllllllll haven't used it yet |
| 01:30.29 | jeev | but i promise i'll send it to you when i do. |
| 01:30.44 | russellb | C4away: it _does_ have that included |
| 01:31.01 | russellb | C4away: using app_fax that is currently in asterisk-addons |
| 01:31.09 | russellb | but it's going to be moved to the main asterisk package before 1.6.0 is released |
| 01:31.22 | C4away | oh |
| 01:31.30 | C4away | ok, all I saw in sip.conf was the t.38 passthrough |
| 01:31.38 | C4away | let me make sure I installed addons |
| 01:31.43 | russellb | there isn't any configuration necessary, AFAIK |
| 01:32.43 | C4away | hmm |
| 01:33.05 | C4away | I guess I didn't install addons, I got librpi zap and 1.6.0-beta9 |
| 01:33.25 | C4away | are there any examples for using it? dialplan syntax or something? |
| 01:33.47 | C4away | do I hand the call off to Fax()? |
| 01:34.57 | russellb | yep, you hand the call to the Fax app |
| 01:35.01 | C4away | it is new enough I'm not finding much about it, although getting better results searching for "app_fax" |
| 01:35.03 | russellb | not sure what the apps are called |
| 01:35.13 | russellb | yeah ... it's pretty new |
| 01:35.20 | russellb | documentation is probably slim to none |
| 01:36.06 | C4away | well, I'm getting paid by the hour to implement it, guess and check doesn't seem like a bad option =) |
| 01:36.10 | C4away | lol |
| 01:36.15 | russellb | heh, nice. |
| 01:36.36 | C4away | "300 hours for T.38 testing?" |
| 01:37.07 | C4away | I might get in trouble for that one |
| 01:38.22 | russellb | if you write up a quick how to, please send it my way :) |
| 01:38.26 | russellb | or even post it on bugs.digium.com |
| 01:38.29 | russellb | and we'll get it included with asterisk |
| 01:38.36 | C4away | I'll post it on www.voipcoop.org if I get it working |
| 01:38.54 | C4away | I'll make sure you get the link |
| 01:39.07 | russellb | great, thanks |
| 01:39.09 | C4away | I'll make a bug report too |
| 01:42.42 | C4away | now if I get it working in its current state, will it change between now and 1.6-release? |
| 01:43.14 | russellb | extremely unlikely (but possible) |
| 01:43.24 | C4away | lol, ok |
| 01:43.28 | russellb | the only change planned is that the module is moving from addons to the main asterisk package |
| 01:43.42 | C4away | but it will be referenced the same from the dial plan? |
| 01:44.21 | russellb | yes |
| 01:44.32 | C4away | ok, cool that's all I was worried about |
| 01:45.01 | C4away | heh, I should probably download asterisk-addons-1.6, oops |
| 01:45.08 | russellb | yup :) |
| 01:45.38 | C4away | good thing I caught that or I would have been really conufsed |
| 01:47.03 | C4away | as is my spelling apparently |
| 01:50.03 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 01:50.34 | mackes | Disco |
| 01:50.41 | Strom_C | Inferno |
| 01:50.54 | mackes | Nice |
| 01:51.03 | Strom_C | (burn baby burn) |
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| 01:52.26 | *** mode/#asterisk [+o stevie_ramjet] by ChanServ |
| 01:52.32 | jeev | Strom_C: http://web.lomag.net/~mark/bad-bannerad.jpg |
| 01:52.54 | Strom_C | hahaha, i remember that |
| 01:52.59 | jeev | heh |
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| 01:53.32 | mackes | Oh, that is terrible! |
| 01:53.48 | hsv-al | heh |
| 01:53.52 | hsv-al | look at the IE |
| 01:53.54 | hsv-al | must be 5 or 6 |
| 01:54.20 | Strom_C | based on the news stories, that screen capture is from 2001 |
| 01:56.26 | jeev | i just smashed an excedrin pill |
| 01:56.27 | jeev | nito pieces |
| 01:56.33 | jeev | and i took 1/8 of 1 pill heh |
| 01:56.33 | stevie_ramjet | on purpose? |
| 01:56.37 | stevie_ramjet | oh |
| 01:56.39 | jeev | should take my headache away |
| 01:56.51 | jeev | i've had a light headache for 4 days |
| 01:57.20 | jeev | it was a gel tab though |
| 01:57.22 | jeev | is that okay? |
| 01:58.21 | drmessano | That's IE5 |
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| 02:03.07 | hsv-al | drmessano |
| 02:03.08 | hsv-al | http://tinyurl.com/6kqquo |
| 02:05.10 | drmessano | Damn |
| 02:05.27 | hsv-al | thats a heavy mod, with dual turbo |
| 02:05.33 | hsv-al | they have a stock one at the audi dealership here |
| 02:05.36 | hsv-al | 110k+ :( |
| 02:05.47 | jeev | heh |
| 02:05.52 | jeev | someone sold their murceilago |
| 02:05.53 | jeev | to pay me off. |
| 02:05.57 | jeev | i'm still waiting for my check |
| 02:06.18 | C4away | hmm |
| 02:06.26 | C4away | app_fax is not listed as a module |
| 02:07.14 | *** part/#asterisk RoyK (n=Roy@ip-105-50-149-91.dialup.ice.no) |
| 02:17.24 | C4away | hey, anyone know if spandsp is available as a binary for CentOS 5? |
| 02:18.46 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 02:21.56 | hsv-al | http://wiki.easyvoxbox.org/tiki-index.php?page=Spandsp%20and%20Asterisk1.4%20on%20CentOS%205 |
| 02:24.21 | ZX81 | hi all - quick question - I have a call queue with 30 phones in it. One is redirecting with a 302 to another phone. If that person is busy (the one being redirected to) it goes to his voicemail instead of calling everyone else |
| 02:24.25 | ZX81 | ideas? |
| 02:25.50 | Strom_L | ZX81: run the queue with the "i" option |
| 02:26.00 | Strom_L | or perhaps it's "I" |
| 02:26.08 | Strom_L | whichever one ignores call forward requests |
| 02:26.28 | jblack | That would be i |
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| 02:28.17 | ZX81 | ah ok sweet as, cheers man |
| 02:30.20 | ZX81 | should have seen it in core show application :) |
| 02:39.16 | C4away | thanks hsv-al |
| 02:39.52 | C4away | now asterisk is trying to tell me that libspandsp.so.1 is not present even though it is |
| 02:39.57 | C4away | restared asterisk and all that |
| 02:40.33 | hsv-al | i think you have to add /usr/local/lib to the conf file |
| 02:40.40 | hsv-al | and re-run ldconfig |
| 02:40.55 | C4away | when I echo $path I get nothing |
| 02:41.04 | hsv-al | 4. Add the line "/usr/local/lib" to the end of /etc/ld.so.conf and run ldconfig |
| 02:41.32 | hsv-al | pay attention to #5 also |
| 02:41.48 | C4away | lol |
| 02:42.09 | C4away | you mean I should actually read those instructions instead of just as far as I need to in order to find the source of the files I'm looking for? |
| 02:42.42 | hsv-al | read the whole thing, default paths may be different with you |
| 02:43.00 | hsv-al | I had that same issue actually on centos5 few months ago |
| 02:43.18 | C4away | what were you working on? t.38? |
| 02:43.19 | hsv-al | I have like 15 virtual machines / vmware to flake around w/ various distros |
| 02:43.20 | hsv-al | heh |
| 02:44.01 | hsv-al | dont use sip really |
| 02:44.11 | C4away | what did you need spandsp for? |
| 02:44.25 | hsv-al | ...... |
| 02:44.26 | C4away | I've never needed it until I started playing with 1.6 and t.38 |
| 02:44.54 | hsv-al | i havent really touched fax/sip combo functionality with it |
| 02:44.58 | hsv-al | but that tutorial should help |
| 02:45.07 | C4away | thanks |
| 02:45.12 | C4away | oh, you were saying different paths |
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| 03:34.34 | C4away | russellb: I am having a very difficult time finding the syntax for app_fax |
| 03:34.52 | russellb | hrm, i'll poke the people who worked on it to document it |
| 03:35.18 | C4away | do you think it is just Answer(); then Fax(SIP/user123@sip_proxy) |
| 03:35.23 | C4away | or something like that? |
| 03:36.32 | russellb | not quite ... |
| 03:36.47 | file | it will only send a TIFF file, and receive a fax into a TIFF file |
| 03:37.08 | file | it will not do gatewaying (have one side be T38, dial another place and do translation) |
| 03:37.22 | russellb | you would have to originate a call between a SIP channel and the transmit fax app (don't know what they are called) |
| 03:37.31 | file | ReceiveFax and SendFax |
| 03:37.48 | russellb | *CLI> originate SIP/user123@sip_proxy application SendFax myfaxfilethingy.tiff |
| 03:38.18 | C4away | hmm |
| 03:38.26 | russellb | or ... you receive a call on your fax did or whatever ... and you just send it to ReceiveFax in the dialplan |
| 03:39.33 | C4away | I'm trying to do SS7-(sip/ulaw)->CallRouter-(sip/ulaw)->FaxGateway-(sip/t.38)->customer's ATA |
| 03:40.21 | file | where does Asterisk fit in? |
| 03:40.46 | C4away | the callrouter and faxgateway in that example are both asterisk systems, the router being 1.4 and the fax gateway being 1.6 |
| 03:40.47 | russellb | asterisk could be any of those things, except the ATA< heh |
| 03:40.55 | C4away | or the ss7 |
| 03:41.02 | C4away | as openss7 is stagnent |
| 03:41.26 | file | Asterisk can not do gatewaying which I believe is what you are describing above |
| 03:41.37 | C4away | while feasable, I don't have the $2000/month to lease some A-links from Qwest to play around with ss7 |
| 03:41.42 | russellb | sounds like just passthrough, actually |
| 03:42.00 | C4away | well it is coming in ULAW and going out T.38 |
| 03:42.08 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 03:42.12 | russellb | why? get an ATA that does T.38 |
| 03:42.21 | C4away | it does |
| 03:42.32 | russellb | oh ... you mean on the gateway it's not t.38 .. |
| 03:42.37 | russellb | faxing is so silly. |
| 03:43.01 | file | Asterisk can not do that, you will have to find an alternate solution if you wish to use T38 |
| 03:43.04 | russellb | and that is my final answer |
| 03:43.16 | russellb | file: thanks for stepping in <3 |
| 03:43.21 | C4away | the call comes in from the SS7 network and is converted to SIP/ulaw, then passes through the call router that knows wich feature server the customer is on, in this case it is a fax call so it passes it to the fax gateway, the fax gateway then dials the customer's ata and sends it as a t.38 call |
| 03:43.33 | C4away | yea |
| 03:43.40 | C4away | hmm, I had heard that 1.6 could |
| 03:43.55 | file | 1.6 can do passthrough, sending, and receiving |
| 03:44.18 | file | it can not translate between audio based faxing and t38 |
| 03:44.20 | C4away | ah |
| 03:44.31 | russellb | coming soonish/eventually |
| 03:44.32 | C4away | so if it comes in t.38 and goes out t.38 asterisk doesn't touch it |
| 03:44.35 | russellb | but don't hold your breath |
| 03:44.51 | file | russellb: shhhh, quiet you! |
| 03:45.05 | C4away | what's it take to make that happen? |
| 03:45.25 | russellb | time |
| 03:45.37 | C4away | by that I mean, what would I need to do to help that progress along? |
| 03:45.39 | russellb | and we're spread thinner than ... um ... something very thin, as it is |
| 03:45.48 | russellb | nothing you can do ... |
| 03:45.58 | russellb | unless you were to write it yourself, heh |
| 03:46.00 | C4away | yea |
| 03:46.17 | file | russellb: I heard air can be thin |
| 03:47.04 | C4away | ok, so what would I need to know to write the patch? I've never worked with codec translation |
| 03:47.14 | brookshire | russellb: are there not t.38 addons? |
| 03:47.24 | russellb | not for gatewaying. |
| 03:47.45 | russellb | C4away: it's a very much non-trivial task |
| 03:48.07 | C4away | I usually jump into projects taking on a non-trivial task as my "hello world" =) |
| 03:48.56 | brookshire | is t.38 even a codec? |
| 03:49.04 | russellb | no, it's not |
| 03:49.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:50.05 | C4away | this one suggests implementing it as a codec: http://bugs.digium.com/view.php?id=11761 |
| 03:52.39 | C4away | interesting idea, seems more of a hack than it is worth |
| 03:52.59 | russellb | i'm surprised that hasn't been closed since there isn't a patch ... |
| 03:53.03 | C4away | can you originate a t.38 call from a file? |
| 03:53.37 | C4away | for example could I receive a ulaw fax to a tiff, then turn around and send that tiff out to the customer over t.38? |
| 03:54.09 | russellb | yes, you can do that |
| 03:54.31 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.1) |
| 03:57.50 | *** join/#asterisk Torikun (n=rusher@75-171-135-108.hlrn.qwest.net) |
| 03:57.55 | Torikun | Yo |
| 03:58.04 | brookshire | that actually, kinda seems efficient |
| 03:58.04 | C4away | ok, what about doing the same thing but using a buffer... receive -> render ->read rendered data stream -> send over t.38 ? |
| 03:59.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:01.19 | C4away | I think I'm going to receive to a file then create a call file pointing to an agi script that sends the fax |
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| 05:16.45 | hsv-al | is zzzzzzzzzzzzzzzzzzzzzz |
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| 06:16.06 | Shotygun | Is it possible that ext 'h' is not executed at a hangup inside a macro even if it's defined at the calling context? 1.4 branch |
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| 06:47.45 | kamanashisroy | hi, is there anything like account template in asterisk ? specially in realtime ? or is there any example of implementation ? |
| 06:53.26 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 06:54.28 | _gm | kamanashisroy: realtime is all broken |
| 06:56.55 | kamanashisroy | _gm: do you have any suggestion ? |
| 06:57.11 | _gm | what exactly you are trying to do |
| 06:57.39 | kamanashisroy | _gm: I am trying to design an integration module so that drupal users have asterisk features .. |
| 06:57.56 | kamanashisroy | _gm: I am trying to integrate drupal users to asterisk account .. |
| 06:58.34 | kamanashisroy | _gm: and I am hoping to use a builder module that will use some predefined template to create accounts .. |
| 07:00.02 | _gm | kamanashisroy: asterisk has nothing to do with your templates, all you will be doing is creating your own custom template, and you will be using your templates in php or whatever |
| 07:00.08 | _gm | to create conf files |
| 07:00.35 | _gm | btw, i have created perl scripts to add remove sip users, and calling them from php |
| 07:01.10 | *** join/#asterisk vgster (n=vgster@93.96.221.240) |
| 07:01.18 | _gm | have you ever heard of perl cpan module Asterisk::config |
| 07:01.26 | _gm | its a asterisk configuration parser written in perl |
| 07:01.54 | _gm | an* |
| 07:01.55 | kamanashisroy | _gm: yes I have seen cpan module .. |
| 07:02.20 | kamanashisroy | _gm: I mean that cpan module that parses asterisk config .. |
| 07:02.24 | _gm | kamanashisroy: believe me i spent 3 weeks, and got nothing, all i was trying to do is make Realtime ldap driver work |
| 07:02.38 | _gm | 3 months* |
| 07:02.38 | kamanashisroy | _gm: :) |
| 07:02.43 | _gm | they don't want to fix it |
| 07:02.55 | _gm | and nobody is interested. |
| 07:03.00 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:03.19 | _gm | plus, i dont think a realtime db backend necessory, |
| 07:04.00 | _gm | why we cant have static configs in mysql/ldap, i dont bother issuing a reload ;) |
| 07:04.16 | kamanashisroy | _gm: you are thinking right .. that is good for performance .. |
| 07:04.51 | _gm | 3com has to be restarted, even after a little change, and asterisk you are all set after issuing some reloads :) |
| 07:05.01 | _gm | then whats the point of realtime |
| 07:05.33 | _gm | btw, i am working on a gui for asterisk these days |
| 07:06.30 | _gm | and using perl for writing asterisk config files, |
| 07:06.46 | _gm | php for frontend |
| 07:06.46 | kamanashisroy | _gm: OK .. |
| 07:06.57 | _gm | and python as an agent on asterisk machine |
| 07:07.07 | _gm | xmlrpc |
| 07:07.07 | kamanashisroy | _gm: :) |
| 07:07.23 | kamanashisroy | _gm: *** xmlrpc *** .. important thinking .. |
| 07:07.27 | _gm | :) |
| 07:07.41 | _gm | we are going to release an opensource version of our product soon |
| 07:07.46 | _gm | in august hope so |
| 07:07.52 | kamanashisroy | _gm: in my design I want to allow the asterisk server to reside in different machine and shared to different sites .. |
| 07:08.04 | _gm | thats what we are doing |
| 07:08.32 | _gm | btw, asterisk is just a module of our product, you can develope your own modules and fit them in our managment portal |
| 07:08.33 | kamanashisroy | _gm: nice .. do you have the xmlrpc api ready to use ? |
| 07:09.27 | _gm | hm everything is ready almost |
| 07:09.38 | _gm | we are fixing bugs and today starting QA |
| 07:09.46 | kamanashisroy | _gm: I am thinking to call xmlrpc functions to insert account, context, queue member etc .. not for authrizing an account .. |
| 07:10.00 | kamanashisroy | I mean asterisk does not call xmlrpc .. but the shared sites do that .. |
| 07:10.04 | _gm | yes thats fast |
| 07:10.23 | _gm | we first trying ssh/scp for config files |
| 07:10.28 | _gm | but its way too much slow |
| 07:10.50 | kamanashisroy | _gm: so you use xmlrpc + perl to write config right ? |
| 07:10.54 | _gm | yes |
| 07:11.21 | kamanashisroy | if it is shared then you must think of security issues .. |
| 07:11.28 | _gm | Asterisk::config perl module helped me a lot |
| 07:11.34 | kamanashisroy | like one site is not violating freedom of others .. |
| 07:11.48 | _gm | python agent is running on secure socket layer |
| 07:11.58 | _gm | auths the user from kerberos |
| 07:12.04 | kamanashisroy | _gm: thanks .. that is great suggestion .. I am thinking to compile my templates .. some contexts to config files .. |
| 07:12.27 | kamanashisroy | OK |
| 07:12.35 | _gm | yes good |
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| 07:29.48 | kamanashisroy | _gm: why do you use ldap .. why not use mysql realtime ? |
| 07:32.17 | tzafrir_laptop | why use mysql realtime? |
| 07:32.47 | tzafrir_laptop | real-time is a run-time overhead |
| 07:33.03 | tzafrir_laptop | rewriting configuration is a configuration change overhead |
| 07:34.16 | _gm | kamanashisroy: management portal we are developing as all based on ldap, so we thought to give a try to realtime ldap |
| 07:34.23 | tzafrir_laptop | kamanashisroy, speaking of templates, are you aware of the [template](!) syntax? |
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| 07:35.35 | kamanashisroy | tzafrir_laptop: syntax ? is there any existing template already which contains the syntax ? let me know .. |
| 07:35.57 | tzafrir_laptop | kamanashisroy, for asterisk config files |
| 07:36.18 | tzafrir_laptop | It is applied by the configuration files reader |
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| 07:37.00 | kamanashisroy | I see .. something like reading config files from database .. ! |
| 07:37.06 | tzafrir_laptop | http://svn.digium.com/svn/asterisk/branches/1.4/doc/configuration.txt |
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| 07:37.28 | kamanashisroy | tzafrir_laptop: I am talking about account template .. |
| 07:37.50 | kamanashisroy | tzafrir_laptop: following .. |
| 07:38.04 | tzafrir_laptop | [account-templace](!) |
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| 07:38.45 | kamanashisroy | tzafrir_laptop: you are great .. I think I am looking for this stuff .. |
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| 07:39.23 | kamanashisroy | tzafrir_laptop: sounds like OO feature ! |
| 07:42.12 | kamanashisroy | tzafrir_laptop: awesome ! |
| 07:44.18 | kamanashisroy | tzafrir_laptop: is there anyway to specify a template in realtime ? :) |
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| 07:45.50 | kamanashisroy | tzafrir_laptop: suppose I want some of my drupal users will be under a specific config template .. and I have the template names in the database foo,bar etc .. |
| 07:46.53 | tzafrir_laptop | kamanashisroy, hmm... I don't suppose |
| 07:46.58 | kamanashisroy | tzafrir_laptop: may be we can solve this problem writing a config module that loads config on demand and then overrides the values at last .. |
| 07:48.02 | kamanashisroy | tzafrir_laptop: or it can be embeded in the core config module that it loads the templates on demand when doing realtime query .. |
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| 07:48.34 | kamanashisroy | I mean it can either be a part of core config module or it can be a part of mysql_realtime for example .. |
| 07:48.52 | kamanashisroy | the former is better while it is not in my hand to implement ! |
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| 08:01.06 | marc7 | why aren't number=> entries in followme.conf painfully obvious? maybe they are... |
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| 08:02.38 | loompek | are variables shared in multiple contexts? |
| 08:02.45 | loompek | s/in/over/ |
| 08:02.59 | Shotygun | They are shared over multiple contexts per channel. |
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| 08:06.28 | puzzled | hi |
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| 08:16.04 | Strom_M | wheeeeeeeee |
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| 08:23.07 | OcterA | hi all |
| 08:24.24 | Strom_M | hi |
| 08:24.26 | LuisTorres | hi |
| 08:24.29 | jblack | hi |
| 08:24.32 | Strom_M | hi |
| 08:25.06 | OcterA | do you know if asterisk work on arm? because I can't detect inband DTMF, and with the same config file on a x86 it's work perfectly |
| 08:25.25 | Strom_M | OcterA: are the two calls using the same codec? |
| 08:26.16 | OcterA | Strom_M : yes, the problem don't come of the codec/phone, because with de exactly same config, it's works on a x86 |
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| 08:26.39 | Strom_M | OcterA: are you /certain/? because inband dtmf only works if the call is using g711 |
| 08:27.01 | Strom_M | and I just want to ruls out the possibility that the call is negotiating with a different codec on the ARM box |
| 08:27.07 | Strom_M | s/ruls/rule/ |
| 08:27.12 | OcterA | Strom_M : I think the problem come of the kernel, do you know if asterisk need any driver in the kernel to works the DTMF? |
| 08:27.20 | OcterA | The codec is alaw |
| 08:27.28 | Strom_M | ok |
| 08:27.36 | Strom_M | AFAIK it's not kernel-dependent at all |
| 08:28.02 | Strom_M | the only kernel drivers are for zaptel |
| 08:28.18 | tzafrir_laptop | OcterA, missing some CPU cycles? |
| 08:28.30 | OcterA | Strom_M : And we need zaptel to detect DTMF? |
| 08:28.34 | Strom_M | NO |
| 08:28.36 | Strom_M | grr |
| 08:28.37 | OcterA | ok |
| 08:28.38 | Strom_M | listen to me |
| 08:28.50 | Strom_M | please just do me a favor and check what asterisk says the codec is for the call ONCE THE CALL IS UP |
| 08:29.09 | OcterA | I have already check this, the codec is alaw |
| 08:29.52 | OcterA | tzafrir_laptop : I don't understand your question => I'm french |
| 08:30.22 | tzafrir_laptop | DTMF-detection is a processing done in the CPU |
| 08:30.22 | OcterA | tzafrir_laptop what did you mean by missing CPU cycles? |
| 08:30.39 | tzafrir_laptop | and maybe the CPU is too busy? |
| 08:30.59 | OcterA | you think thats not work because my CPU is not efficent (200Mhz) |
| 08:32.08 | tzafrir_laptop | OcterA, Zaptel does not detect DTMFs. All of the DTMF detection of Asterisk is done in the core of Asterisk |
| 08:32.29 | tzafrir_laptop | The kernel code of Zaptel does not do DTMF detection |
| 08:33.33 | tzafrir_laptop | as a test: record the stream (with monitor) and look at it with e.g. audacity . Does it make sense? |
| 08:35.00 | OcterA | if I make the DTMF with a phone, the frequency are good, record and analysed with FFT, but asterisk don't view them, and the generating of DMTF don't work on the arm |
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| 08:35.45 | Strom_M | OcterA: why not just forget all the inband DTMF nonsense and use something sensible like RFC2833? |
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| 08:36.29 | sysadmin-lb22 | OFFTOPIC hi all can I setup a web page in such a way that if a client clicks some link "my sip number" their default voip phone will be launched and tries to call that sip number ? |
| 08:36.56 | OcterA | I can't make this, because the server is linked with a analog platform, and I need DTMF because the RFC2833 isn't on analog line |
| 08:37.47 | florz | sysadmin-lb22: that's a mostly client-side thing - but obviously there are SIP URIs ... |
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| 08:51.12 | kams_brb | OcterA: add an asterisk mediator that does heavy task on a High MH CPU ? |
| 08:51.53 | marc7 | what do you guys suggest I put in a friend's entry in sip.conf that's generating this error "Received SIP subscribe for peer without mailbox" |
| 08:53.12 | Strom_M | marc7: take out the mailbox line? |
| 08:53.44 | marc7 | Strom_M: there is no mailbox line, I'm thinking I need to add one for that friend even though there's no entry in voicemail.conf for them |
| 08:53.58 | marc7 | this is recent, from an update from 1.4.18 to 1.4.20.1 |
| 08:54.15 | Strom_M | *shrug* |
| 08:56.49 | OcterA | kams_brb : with the command top, I see 96% cpu idle, with a call to the asterisk and asterisk generating DTMF (but I ear nothing) |
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| 09:08.49 | bootc | hey folks, I'm doing a Dial() with multiple phones and want to put in CDR which phone picked up |
| 09:09.12 | bootc | I can't just look at dstchannel though since I need to translate that into a user (hot desking) and put it in the accountcode field |
| 09:09.43 | bootc | I've tried using M() but that can't seem to change the CDR, and G() prevents the call from being bridged |
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| 09:18.32 | Strom_M | bootc: why do you want to know who answered? |
| 09:19.35 | bootc | so that when I record the call I can tell who dealt with the caller |
| 09:20.11 | Strom_M | perhaps you should investigate using Queue() instead of Dial() |
| 09:22.14 | bootc | I tried that before but ran into trouble when trying to time out dials in the queues |
| 09:22.29 | Strom_M | what do you mean "time out dials in the queues"? |
| 09:22.31 | bootc | I need to have variable timeouts in some situations |
| 09:22.52 | bootc | sometimes I need to ring the phones for 6 seconds, and sometimes for 12 seconds depending on how the call comes in |
| 09:23.15 | bootc | for example, if you ring the main number it rings queue1 for 12s, queue2 for 6s, then queue3 forever |
| 09:23.36 | bootc | it you ring a DDI it rings the phone for 12s, queue1 for 6s, queue2 for 6s, queue3 indefinitely |
| 09:24.38 | Strom_M | that shouldn't be difficult to do with queues at all |
| 09:25.03 | bootc | I did try to do this using queues before, but I just couldn't achieve that goal |
| 09:25.26 | bootc | I would get into the situation where the phones would ring for 6s and then just not ring at all for the remaining 6s |
| 09:27.14 | Strom_M | what about using the h extension to manipulate the CDR while the call is being torn down? |
| 09:27.53 | bootc | I had thought of that but I can't seem to get it to work |
| 09:27.57 | Shotygun | Quick one, is it possible that ext 'h' is not executed at a hangup inside a macro even if it's defined at the calling context? 1.4 branch |
| 09:28.21 | bootc | what I'm doing is in from_pstn calling a call_group_timeout macro which does the Dial |
| 09:28.38 | bootc | and strangely Shotygun's question is something that came up when I was testing this |
| 09:29.02 | bootc | h is not called in from_pstn, and I can't add it to the macro because I'm using AEL2 |
| 09:29.44 | Shotygun | I didn't follow your conversation bootc, but when using the legacy extensions (not ael) I noticed that during a Record() the 'h' extension is not called when there is a hangup. neither 'h' at the macro or 'h' at the calling context. |
| 09:30.10 | Shotygun | I had to replace the Macro with a regular context for this to work. |
| 09:30.33 | bootc | Shotygun: I'm using MixMonitor so this is probably what happens here |
| 09:31.24 | Shotygun | bootc: During macro or context? |
| 09:33.19 | bootc | context |
| 09:33.44 | bootc | well, the context calls a different macro to start recording |
| 09:33.46 | bootc | :-) |
| 09:34.00 | Shotygun | Then it's macro, the question is where MixMonitor is called from, macro or context |
| 09:34.08 | Shotygun | If it's macro then it's the same issue that I have, and maybe it's a bug |
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| 09:36.50 | tuxx- | hmz |
| 09:37.22 | tuxx- | i got a couple of companys configured on my asterisk server. They all have their own voicemailbox, but when i call to the numbr and get the voicemail it always plays the |
| 09:37.25 | tuxx- | vm-intro.gsm file |
| 09:38.02 | tuxx- | anyone have any idea how i can make the voicemail play the recorded sounds? (recorded sounds are in the /var/lib/asterisk/voicemail/ dir). |
| 09:38.26 | Shotygun | tuxx-: It means that the extension that is calling the VoiceMail() function is not giving the right extension as an argument. Are you using FreePBX? |
| 09:38.48 | tuxx- | nope, just asterisk ;p |
| 09:39.18 | Shotygun | Then make sure you are calling VoiceMail() with the proper voicemail number. |
| 09:39.28 | Shotygun | You need to provide the box number as an argument |
| 09:39.40 | tuxx- | exten => s,2,Voicemail(306@mb_360) |
| 09:39.48 | tuxx- | [mb_360] |
| 09:39.48 | tuxx- | 306 => 0000,360 ICT,meldingen@360ict.nl,,|attach=yes| |
| 09:39.50 | tuxx- | :P |
| 09:39.54 | tuxx- | looks correct, right? :) |
| 09:39.58 | Strom_M | what's the text of vm-intro.gsm again? |
| 09:40.22 | tuxx- | someone put some own recorded message into vm-intro. so i don't know :P |
| 09:40.42 | Shotygun | "Please leave your message after the tone, when done hang up, or press the pound key" |
| 09:41.17 | Strom_M | that recording will always play after the user-recorded message unless you call voicemail with the "s" option |
| 09:41.19 | Shotygun | I like to put nbdy-avail-to-take-call.gsm and afterwards vm-intro.gsm |
| 09:41.53 | tuxx- | hmkay |
| 09:42.07 | tuxx- | but i recorded an unavail message, but somehow i dont hear that one when i call... |
| 09:42.23 | tuxx- | very strange |
| 09:42.32 | Strom_M | tuxx-: you have to call voicemail with the "u" option to play the unavailable message |
| 09:42.38 | tuxx- | hmmmmkay |
| 09:42.43 | tuxx- | lemme try it out :) |
| 09:42.45 | Strom_M | hmmmmmmmmmmkaaaaayyyyy |
| 09:45.56 | tuxx- | yay, it works. Thanks alot! :) |
| 09:46.04 | Strom_M | you're welcome |
| 09:46.27 | tuxx- | \o/ |
| 09:46.34 | Strom_M | please pass nine pizzas |
| 09:46.38 | tuxx- | hehe |
| 09:46.43 | Strom_M | in a floorwise fashion! |
| 09:46.43 | tuxx- | nypizza.com ? |
| 09:46.43 | tuxx- | ;x |
| 09:47.05 | Strom_M | new york is kind of far away from los angeles |
| 09:47.25 | tuxx- | they probably have an nyp in LA |
| 09:47.26 | tuxx- | i guess |
| 09:47.26 | tuxx- | ;x |
| 09:47.38 | Strom_M | heh, um, nope. :P |
| 09:47.45 | tuxx- | damn >_> |
| 09:48.02 | Strom_M | there is at least one independent pizzeria here, though, that makes absolutely amazing new york style pizza |
| 09:48.15 | tuxx- | :-) |
| 09:48.48 | Shotygun | Can anyone explain to me briefly what is the function pbx_builtin_getvar_helper() and is there is a cli debugging command its settings? |
| 09:48.58 | bootc | Shotygun: so your thinking is if I call MixMonitor from the context I should get the h extension called in that context? |
| 09:49.17 | Shotygun | bootc: Yeah |
| 09:49.41 | Shotygun | Let me know if it does, I'm very curious |
| 09:51.45 | bootc | nope not seeing it |
| 09:52.07 | bootc | I set up MixMonitor in from_pstn, called my call_group macro, and h doesn't get called in from_pstn |
| 09:53.39 | Shotygun | What is call_group ? |
| 09:54.12 | bootc | it's a macro of mine |
| 09:54.49 | Shotygun | Try to avoid the macros, just put mixmonitor inside a context and call that context with no macros in it, just mixmonitor, noops & 'h' |
| 09:56.02 | bootc | aha, that did work |
| 09:56.21 | Shotygun | I'm telling ya, it's bug in app_macro I think |
| 09:56.37 | bootc | hmm, but I'm using AEL2 which doesn't use app_macro |
| 09:56.47 | bootc | hmm, actually, yes it does |
| 09:56.51 | Shotygun | It does |
| 09:56.53 | Shotygun | while is also app |
| 09:56.55 | bootc | is very confused now |
| 09:57.13 | bootc | I wonder if Gosub works then |
| 09:57.58 | Shotygun | It's kinda part of the same code from what I've seen but I could be wrong of course, let me know if it works for ya. |
| 09:59.18 | bootc | gonna try it |
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| 10:20.29 | Great_Anta_Baka | hi |
| 10:20.38 | Great_Anta_Baka | i have polycom phones in my setup |
| 10:20.51 | Great_Anta_Baka | if i transfer using normal transfer |
| 10:20.58 | Great_Anta_Baka | it does not generate a ringtone |
| 10:21.02 | Great_Anta_Baka | to the caller |
| 10:21.13 | Great_Anta_Baka | but if i transfer using transfer blind there is a ring tone |
| 10:21.29 | Great_Anta_Baka | to the calling party that is. |
| 10:21.46 | Great_Anta_Baka | what could be the possible reasons for it doing this? |
| 10:21.55 | Strom_M | Great_Anta_Baka: well that's because attended transfer doesn't connect the calling party to the destination party until after the transferring party releases from the call |
| 10:22.13 | Strom_M | also, could you please not press "enter" after every phrase? because that's really irritating. |
| 10:22.24 | Great_Anta_Baka | sorry |
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| 10:23.17 | Great_Anta_Baka | is there anyway to generate a ringtone while the caller is waiting for the callee to pickup? |
| 10:23.42 | Strom_M | Great_Anta_Baka: forget this "caller" and "callee" nonsense -- it's "calling party" and "called party" |
| 10:23.56 | Great_Anta_Baka | k |
| 10:23.59 | Strom_M | also, at what point do you expect the calling party to hear alerting tone? |
| 10:24.37 | Great_Anta_Baka | well i press blind, extension number and then dial |
| 10:24.56 | Great_Anta_Baka | immediately afterthis a ringtone is generated but it does not generate one at all when i am doing a normal transfer |
| 10:25.25 | Strom_M | you don't hear alerting tone from the station doing an attended transfer? |
| 10:25.36 | Great_Anta_Baka | no |
| 10:25.53 | Strom_M | are you transferring to another polycom set? |
| 10:26.04 | Great_Anta_Baka | i am using freepbx on that box btw. and music on hold does work when you transfer between calls |
| 10:26.08 | Great_Anta_Baka | yes |
| 10:26.13 | Strom_M | oh god, not freepbx |
| 10:26.20 | Great_Anta_Baka | yeah i know |
| 10:26.41 | Great_Anta_Baka | i didnt install it but they want to keep everything as it is, so i have to kinda figure my way around it |
| 10:26.50 | Strom_M | sigh |
| 10:27.16 | Great_Anta_Baka | could this be freepbx related? |
| 10:27.25 | Strom_M | I would suspect so |
| 10:27.42 | Strom_M | it's such a convoluted jumble of useless crap that it's difficult to tell where the problem lies |
| 10:28.11 | Great_Anta_Baka | yeah after using it at just that one site i have never looked at it. asterisknowgui fTW or scopserve |
| 10:30.18 | Great_Anta_Baka | i thought it was weird that when i tell it to use music on hold for internal calls it works and the calling party hears music but the ceo didnt like the fact that when he called his secretary he heard it :( |
| 10:30.28 | Great_Anta_Baka | thanks for the help btw |
| 10:30.41 | Strom_M | you're welcome |
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| 10:31.13 | Strom_M | with regular asterisk i'd know where to look, but i've tried chasing down problems like this with freepbx and it's like trying to find a needle in a football stadium |
| 10:31.14 | DarKnesS_WolF | i want to make some modifcation on the CDR after the call is done like change the dst number, any idea how ? |
| 10:31.24 | Strom_M | DarKnesS_WolF: "h" extension |
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| 10:33.29 | DarKnesS_WolF | Strom_M: yes but then ? i'll use normal bash or PHP thing to change the dst in database ? |
| 10:33.32 | DarKnesS_WolF | or what ? |
| 10:34.01 | Strom_M | why are you modifying the CDR post-call? |
| 10:34.13 | bootc | Shotygun: indeed using gosub works as expected |
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| 10:37.38 | bootc | how do people here rate 1.6's stability? |
| 10:40.13 | loompek | i was kinda dissapointed... |
| 10:40.15 | *** join/#asterisk FrankPiffleHeart (n=boffvert@81-178-81-165.dsl.pipex.com) |
| 10:40.48 | loompek | disappointed |
| 10:40.49 | loompek | even |
| 10:41.03 | *** join/#asterisk FrankPiffleHeart (n=boffvert@81-178-81-165.dsl.pipex.com) |
| 10:42.10 | Strom_M | oh for god's sake, it's not even out of beta yet |
| 10:42.23 | Strom_M | give it a couple of bugfix releases |
| 10:43.51 | FrankPiffleHeart | Looking for help setting up tdm410 in the uk, could anyone point to some decent documentation, any help appreciated! |
| 10:44.21 | Strom_M | FrankPiffleHeart: well, the documentation included with asterisk or in the book should be sufficient; what specific problem are you having? |
| 10:45.25 | FrankPiffleHeart | unfortunately the book aint helping; it wont pick up wether it be an automated pickup to a voice box or transferred to an fxs and physically picked up |
| 10:46.18 | Strom_M | what do you mean "it wont pick up"? |
| 10:46.47 | Strom_M | you're going to have to provide a more detailed explanation of the problem if you expect me to be able to troubleshoot it |
| 10:47.00 | FrankPiffleHeart | appreciate that |
| 10:47.42 | gr0mit | FrankPiffleHeart, can you give us a bit more background please |
| 10:47.53 | gr0mit | we will do what we can to assist |
| 10:47.54 | FrankPiffleHeart | Incoming call to an fxo, rings fine, but if asterisk or handset is set to recieve a call it fails. |
| 10:48.31 | gr0mit | still not quite sure what you mean..... |
| 10:48.52 | gr0mit | so, you plug your BT line into an FXO, right? |
| 10:48.57 | gr0mit | then you call the BT line |
| 10:48.59 | FrankPiffleHeart | yep pluggedin |
| 10:49.03 | gr0mit | and what happens? |
| 10:49.05 | FrankPiffleHeart | <PROTECTED> |
| 10:49.07 | FrankPiffleHeart | rings |
| 10:49.23 | gr0mit | what rings? the BT line? |
| 10:49.48 | FrankPiffleHeart | fxs if routed to or can see the status in the cli if routed to a voicebox |
| 10:50.06 | Strom_M | FrankPiffleHeart: please don't turn this into one of those "like pulling teeth from a lion" troubleshooting sessions... |
| 10:50.12 | FrankPiffleHeart | sorry |
| 10:50.35 | gr0mit | ok, so does Asterisk ever answer the call? |
| 10:50.51 | FrankPiffleHeart | nope breaks fails |
| 10:51.14 | FrankPiffleHeart | goes down a couple of tasks but its failed at "pick up" |
| 10:51.45 | FrankPiffleHeart | if at alll possible direct me to documentation because i'll just be an irratation |
| 10:52.32 | gr0mit | well if you have looked at voip-info.org and the Asterisk book to no avail, it is probably a config issue or a concept problem |
| 10:53.28 | gr0mit | asterisk has a few concepts which are hard to grasp to start with |
| 10:53.32 | DarKnesS_WolF | Strom_M: i am doing something like programing a button when i call it it dial extension and i want to replace that extension with something else |
| 10:54.04 | gr0mit | firstly, just checking- you are running Asterisk, not one of the clones which forces you to use a web interface, right? |
| 10:54.10 | Strom_M | DarKnesS_WolF: huh? that makes no sense. |
| 10:55.17 | FrankPiffleHeart | tried compiling with centos, problem arose with voice box, so i switched to asterisknow to limit my potential to do it wrong but seems to have the same problem either way. |
| 10:55.33 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 10:56.30 | Strom_M | FrankPiffleHeart: have you tried calling Digium support? They'll help you get your hardware installed and working properly for no extra charge. |
| 10:57.10 | gr0mit | indeed |
| 10:57.24 | gr0mit | FrankPiffleHeart, are you fairly new at this? |
| 10:57.28 | FrankPiffleHeart | ow ok, i'll give that a go, appreciate you efforts sorry for the irratation. |
| 10:57.41 | FrankPiffleHeart | Yeah first attempt. |
| 10:57.56 | gr0mit | ok, well please don't think you are irritating anyone |
| 10:58.04 | gr0mit | we all went through the pain barrier |
| 10:58.17 | FrankPiffleHeart | i see. |
| 10:58.30 | gr0mit | there are a lot of concepts which are hard to grasp initially |
| 10:58.39 | FrankPiffleHeart | I presumed on purchase that this is a common card |
| 10:58.46 | gr0mit | but when you do get the hang of it you will be fine |
| 10:58.52 | gr0mit | which card again? |
| 10:58.56 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:58.59 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 10:59.00 | FrankPiffleHeart | So people in the tdm410p |
| 10:59.00 | gr0mit | tdm410? |
| 10:59.02 | gr0mit | ok |
| 10:59.05 | FrankPiffleHeart | yep |
| 10:59.16 | gr0mit | btw where in uk are you? |
| 10:59.34 | FrankPiffleHeart | Bt line Leicester. |
| 10:59.39 | gr0mit | ah ok. |
| 10:59.52 | gr0mit | too far from me ;-) |
| 11:00.15 | FrankPiffleHeart | people will have this card up and working in the uk, won't they? |
| 11:00.26 | gr0mit | I'm sure of that |
| 11:00.32 | gr0mit | however, |
| 11:00.42 | gr0mit | ISDN will give you better performance |
| 11:00.45 | FrankPiffleHeart | ok i ll go harass digium support |
| 11:00.50 | gr0mit | on a £15 card |
| 11:00.56 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 11:01.01 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 11:01.12 | FrankPiffleHeart | i just wanted it to pick up and handle landline to begin with |
| 11:01.29 | Rico29 | does somebody knows where I can find a good TTS soft (or web-based application) which can make .wav (or .gsm) with good quality ? |
| 11:01.35 | Rico29 | (sorryfor my english...) |
| 11:01.47 | gr0mit | well, make sure you enable ssh access to your asterisk box, FrankPiffleHeart |
| 11:01.50 | Strom_M | gsm and good quality don't belong in the same sentence |
| 11:01.55 | gr0mit | then calls Digium |
| 11:01.58 | Rico29 | yes |
| 11:02.05 | FrankPiffleHeart | ok will do thanks for your help/direction bb |
| 11:02.07 | gr0mit | they will ssh into your box and get the card running for you |
| 11:02.24 | gr0mit | if you get stuck ask here |
| 11:02.38 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 11:02.38 | Strom_M | Rico29: "Cepstral" is generally considered to be fairly decent for TTS |
| 11:02.45 | Rico29 | ok thanks |
| 11:02.46 | FrankPiffleHeart | yeah np, will prepare, thanks again bb |
| 11:02.52 | gr0mit | offers consultancy if you want it too, FrankPiffleHeart |
| 11:03.12 | gr0mit | but better to try get your solution for free here ;-) |
| 11:03.29 | Strom_M | not only do I offer consulting, but I take all major credit cards too! |
| 11:03.29 | Strom_M | :) |
| 11:03.32 | FrankPiffleHeart | <PROTECTED> |
| 11:03.39 | FrankPiffleHeart | bb |
| 11:03.42 | *** part/#asterisk FrankPiffleHeart (n=boffvert@81-178-81-165.dsl.pipex.com) |
| 11:03.47 | gr0mit | likes bank payments! |
| 11:04.27 | gr0mit | Strom_M, where abouts are you then? |
| 11:04.55 | Strom_M | Los Angeles |
| 11:04.56 | *** join/#asterisk Sniper_mich (n=michofr@mail.splendor.net) |
| 11:05.09 | gr0mit | bit early for you isn;t it!?! |
| 11:05.15 | gr0mit | or late |
| 11:05.34 | Strom_M | I took an extended nap in the late afternoon |
| 11:11.29 | jblack | He also guarantees his answers, so if he gives you shit, you can get a refund. =) |
| 11:11.52 | Strom_M | haha yes |
| 11:12.22 | jblack | speaking of money, I need to find a new client. |
| 11:14.03 | Strom_M | yeah, as do I. |
| 11:14.30 | Strom_M | actually, I need my potential client to hurry up and make a decision :) |
| 11:14.48 | Sniper_mich | Hi all...I would like to ask you please aboutthe configuration that I should add in order to register asterisk server on the provider as static gateway and not an endpoint by adding a trunk |
| 11:15.18 | jblack | My current one is ok, and have paid on time so far, but the onsite tech is slowing things up. |
| 11:15.31 | Strom_M | Sniper_mich: so you want asterisk to function as a SIP proxy? |
| 11:16.00 | Strom_M | jblack: well, if the onsite tech slows you down, that's just more consulting income for less work |
| 11:16.16 | Strom_M | let the client tell the tech to get out of your way if it's costing them too much |
| 11:16.35 | jblack | I'm not costing them much, but it's costing them. |
| 11:16.37 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:17.15 | Sniper_mich | Strom_M, In order to send calls from an extension registered on my asterisk server, I added a trunk in order to register my asterisk server on my provider as an endpoint(With username and password) |
| 11:17.17 | jblack | they've had two pri's for over a month, because he can't bother to get all 24 phones plugged in. |
| 11:17.51 | jblack | Doing nothing isn't something I'm very good at. |
| 11:17.54 | Sniper_mich | I would like to remove the trunk and send traffic from asterisk server to the provider by registering Asterisk server as static gateway on my provider |
| 11:18.12 | Strom_M | Sniper_mich: that's not how asterisk works. asterisk is not a SIP proxy. |
| 11:18.20 | Strom_M | asterisk is a back-to-back user agent. |
| 11:18.30 | jblack | Sniper_mich: Check out siproxd |
| 11:18.44 | Strom_M | or have a look at SER |
| 11:19.50 | Strom_M | jblack: yeah, my philosopy is to always provide reliable, useful, carefully and thoroughly thought-out advice, but let the client make the final decision |
| 11:19.54 | Sniper_mich | Strom_M, I found a solution on google but did't work |
| 11:19.55 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:20.03 | Sniper_mich | it says to add the following lines to sip.con |
| 11:20.05 | Sniper_mich | [provider1.sip] |
| 11:20.06 | Sniper_mich | <PROTECTED> |
| 11:20.06 | Sniper_mich | type=peer |
| 11:20.06 | Sniper_mich | fromdomain=asterisk IP address |
| 11:20.06 | Sniper_mich | host=provider IP address |
| 11:20.07 | Sniper_mich | exten => _X.,n,Dial(SIP/${EXTEN}@provider1.sip|40) |
| 11:20.14 | Strom_M | it's more stressful for you in the short term but makes the relationship easier |
| 11:20.21 | Strom_M | Sniper_mich: please use pastebin |
| 11:20.22 | Strom_M | ~pb |
| 11:20.23 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 11:20.42 | Strom_M | Sniper_mich: also, that exten line doesn't go in sip.conf |
| 11:22.17 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 11:22.40 | Strom_M | jblack: there's a very good reason my warranty explicitly doesn't cover anything the client does against my recommendations |
| 11:23.06 | Sniper_mich | Strom_M,Please check the following |
| 11:23.08 | Sniper_mich | http://pastebin.com/m74d83ed6 |
| 11:23.27 | Strom_M | Sniper_mich: that tells me nothing |
| 11:24.36 | jblack | Oh, I'm covered |
| 11:25.50 | Strom_M | damnit, I had wonderful inspiration to write a new landing page for my website, and writing the final sentence is taking just as long as it took to write an entire screenfull of text. |
| 11:26.02 | Sniper_mich | Strom_M,Please check this...It'll be more clear |
| 11:26.05 | Sniper_mich | http://pastebin.com/m91a3b92 |
| 11:26.41 | Strom_M | there is nothing more frustrating than knowing exactly the impression you want to convey and not being able to find the combination of words that conveys it quite the way you want. |
| 11:27.14 | Strom_M | Sniper_mich: that still tells me nothing. What errors are you receiving? What does your CLI look like? etc etc etc etc etc |
| 11:27.25 | Sniper_mich | ok |
| 11:27.33 | Strom_M | I may be an experienced consultant, but I'm not a mind-reader. |
| 11:28.29 | bootc | Shotygun: I can start the MixMonitor in a macro, but I have to Gosub for the h extension to work |
| 11:33.04 | Sniper_mich | Strom_M,I'm getting the following error |
| 11:33.05 | Sniper_mich | TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack |
| 11:33.26 | Strom_M | alright...well, the channel is unavailable |
| 11:33.41 | Strom_M | did you configure it in accordance with your ITSP's instructions? |
| 11:34.45 | Sniper_mich | Strom_M,I think we should add an outbound route |
| 11:34.57 | Strom_M | "outbound route"> |
| 11:34.59 | Strom_M | ? |
| 11:35.01 | Strom_M | are you using a GUI? |
| 11:40.37 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 11:41.49 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 11:41.57 | Sniper_mich | Strom_M,Thanks a lot...I found it and it's working fine |
| 11:46.03 | *** join/#asterisk viperdude (i=viperdud@87-127-248-92.no-dns-yet.enta.net) |
| 11:46.26 | Strom_M | you're welcome |
| 11:46.51 | viperdude | hi all, is it possible to use asterisk with alaw and 10ms packets? |
| 11:47.09 | Strom_M | alaw yes....I don't know about being able to make the packets 10ms |
| 11:47.19 | Strom_M | why mus the packets be 10ms? |
| 11:47.39 | viperdude | this is because the incoming calls will be encoded at 10ms |
| 11:47.43 | Strom_M | s/mus/must/ |
| 11:47.52 | Strom_M | viperdude: ok...why? |
| 11:48.11 | viperdude | it is a tech trial BT are doing in the UK and that is what they are mandating |
| 11:48.43 | Strom_M | ok |
| 11:48.51 | Strom_M | using SIP? |
| 11:49.01 | viperdude | yes, but openser in front of asterisk |
| 11:49.21 | Strom_M | *shrug* might as well just try it |
| 11:49.39 | viperdude | but do you know if it is likely to work? |
| 11:50.06 | Strom_M | I would guess that it will work, but I don't really know for certain |
| 11:50.46 | viperdude | ok thanks |
| 12:00.03 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:01.47 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 12:05.22 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:12.47 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:12.47 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:15.14 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:20.55 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 12:26.33 | Great_Anta_Baka | does anyone here know what the polycom status icons mean |
| 12:26.46 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-70-221-48.myvzw.com) |
| 12:26.51 | [TK]D-Fender | Great_Anta_Baka: All in the admin & user guides. |
| 12:26.59 | Great_Anta_Baka | k thanks |
| 12:27.09 | [TK]D-Fender | Great_Anta_Baka: And they are largely obvious |
| 12:29.44 | Strom_M | [TK]D-Fender: I think that one that looks like a loudspeaker has something to do with poutine |
| 12:29.51 | Great_Anta_Baka | they are but there is an arrow going up and down arrow on one phone where there is supposed to be a solid phone handset |
| 12:29.52 | [TK]D-Fender | mmmmmm |
| 12:30.25 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:30.25 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:30.31 | Great_Anta_Baka | and it was working perfectly till a second ago |
| 12:30.38 | [TK]D-Fender | Great_Anta_Baka: And what else do you see on display? |
| 12:31.07 | Great_Anta_Baka | it shows the extension number everything looks standard |
| 12:31.14 | Great_Anta_Baka | expect for the icon next to the extension number |
| 12:32.05 | [TK]D-Fender | Great_Anta_Baka: Well check the guide on that one... I'd be wondering if its a reg timeout or something |
| 12:32.15 | Strom_M | maybe it means you forwarded the line appearance |
| 12:32.58 | Great_Anta_Baka | what does that mean? cos when switchboard transfers calls to that extension it reverts directly back to switchboard |
| 12:33.25 | ManxPower | Great_Anta_Baka: The phone is configured to forward calls. |
| 12:33.28 | Strom_M | yup, you forwarded the line appearance |
| 12:34.28 | Great_Anta_Baka | ah thanks for the help |
| 12:34.30 | [TK]D-Fender | Great_Anta_Baka: when you say up/down, I'd be guessing more like 45 deg down, then back up |
| 12:34.38 | [TK]D-Fender | Great_Anta_Baka: Which would indeed be forwarding |
| 12:34.38 | Great_Anta_Baka | she put the phone on call forward |
| 12:34.57 | [TK]D-Fender | Great_Anta_Baka: go download the user's guides for your phones and hand them out and have them read them. |
| 12:36.07 | coppice | can anyone access www.linksys.com right now? |
| 12:36.08 | Great_Anta_Baka | i wish it was that simple but people here in this company are very stubborn. I am just going to start charging them i guess until they learn :( |
| 12:36.29 | Great_Anta_Baka | yes |
| 12:36.30 | ManxPower | You can't accidental forward calls on polycoms |
| 12:37.10 | ManxPower | Great_Anta_Baka: thing of the users as dogs and you won't be disappointed. |
| 12:37.14 | Great_Anta_Baka | seriously these people do strange things.. i wish there is a way to disable it completely but alas no such feature in the config files |
| 12:37.25 | Great_Anta_Baka | hahaha will do |
| 12:37.36 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 12:38.27 | ManxPower | Great_Anta_Baka: the polycom admin guide should tell you how to disable call forward features on the phone |
| 12:44.00 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 12:46.15 | *** join/#asterisk Sniper_micho (n=michofr@mail.splendor.net) |
| 12:47.07 | Sniper_micho | Hi ALL,I need to know please how can I force the asterisk server to send the extension number as ANI instead of "asterisk" as ANI? |
| 12:47.43 | Strom_M | ANI? or calling party number? |
| 12:47.48 | Strom_M | and over what kind of interface? |
| 12:47.51 | ManxPower | Sniper_micho: Set(CALLERID(ani)=5045551212) does not work? |
| 12:47.59 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 12:48.18 | hsv-al | hello all - are we all looking forward to another long & glorious week of irc addiction? |
| 12:48.32 | Strom_M | hsv-al: oh yes |
| 12:48.48 | Sniper_micho | ManxPower, I don't need a static ANI...The ANI in each call should be the extension number of the extension that trying to make the call |
| 12:49.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:49.12 | Strom_M | Sniper_micho: how about setting callerid number in the channel driver config file? |
| 12:49.16 | sysadmin-lb22 | hi all |
| 12:49.20 | ManxPower | Sniper_micho: then modify my example to do so, it's not rocket science. Set(CALLERID(ani)=${CALLERID(num)}) |
| 12:49.29 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 12:49.35 | hsv-al | sysadmin, you're not the only one guilty of it |
| 12:49.44 | hsv-al | I've always wondered why people irc from mailservers, heh. |
| 12:49.55 | Sniper_micho | ManxPower, I should add this line in extensions.conf right? |
| 12:50.01 | sysadmin-lb22 | is it possible for me to setup asterisk on two different locations so that if I change settings on the first server the settings get replicated on the second server |
| 12:50.13 | ManxPower | Sniper_micho: I suggest reading The Good Book first. |
| 12:50.14 | ManxPower | ~book |
| 12:50.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 12:50.15 | [TK]D-Fender | Sniper_micho: Do something like that before you dial. |
| 12:50.20 | Strom_M | sysadmin-lb22: yes -- rsync or realtime architecture |
| 12:51.00 | ManxPower | Sniper_micho: How are you determining the ANI infor? |
| 12:51.15 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 12:51.16 | Strom_M | Sniper_micho: are you aware of the difference between ANI and Caller ID? |
| 12:51.31 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 12:51.33 | Great_Anta_Baka | please explain if you can |
| 12:51.42 | sysadmin-lb22 | Strom_M, what about using NFS and using one shared configuration dir |
| 12:51.54 | Strom_M | sysadmin-lb22: that could work too |
| 12:52.00 | Sniper_micho | ManxPower, I'm trying to send a call from asterisk server to another SIP Proxy...If I check the ANI sent by the asterisk server thenI find that is "asterisk" |
| 12:52.19 | Sniper_micho | I need to change it in order to have extension number as ANI instead of "asterisk" word |
| 12:52.31 | Strom_M | Sniper_micho: are you aware of the difference between ANI and Caller ID? |
| 12:52.57 | Sniper_micho | Strom_M,They are the same no? |
| 12:52.58 | ManxPower | Sniper_micho: It should be impossible to have an ANI of "asterisk", as ANI is supposed to be all numbers. Asterisk probably does not enforce that rule. |
| 12:53.07 | hsv-al | strom, has the directory structures changed that much |
| 12:53.08 | Strom_M | Sniper_micho: far from it |
| 12:53.10 | ManxPower | Sniper_micho: NO THEY ARE NOT THE SAME |
| 12:53.14 | hsv-al | in centos 4.6 from 4.4? |
| 12:53.26 | Strom_M | hsv-al: I have no clue. I don't run centos |
| 12:53.53 | Sniper_micho | what is the suggestions? |
| 12:54.33 | Sniper_micho | ManxPower, I got the difference |
| 12:55.09 | Sniper_micho | ManxPower, Is there any place where I can ask asterisk to send ANI? |
| 12:55.37 | ManxPower | Sniper_micho: we told you how to do several times. |
| 12:56.18 | ManxPower | Sniper_micho: perhaps your real problem is that the CALLERID number is not correct. |
| 12:57.12 | Sniper_micho | ManxPower, o you know from where the asterisk server is giving that caller ID "asterisk" |
| 12:57.14 | Sniper_micho | ? |
| 12:57.49 | ManxPower | Sniper_micho: It will do that if it does not receive callerid information |
| 12:58.30 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-181-11.dsl.teksavvy.com) |
| 12:58.39 | [TK]D-Fender | Sniper_micho: Where is the originating call coming from? |
| 12:59.01 | Sniper_micho | [TK]D-Fender, From an extension registerd on the asterisk server |
| 12:59.17 | ManxPower | Maybe you need to put a Noop(CALLERID(num) is ${CALLERID(num)} and CALLERID(ani) is ${CALLERID(ani)}) just before the Dial line to see what ASTERISK thinks those settings are. |
| 12:59.46 | [TK]D-Fender | Sniper_micho: How do you figure * should come up with some sort of "default ANI"? |
| 12:59.54 | ManxPower | Sniper_micho: your lack of understanding telecom and Asterisk is slowing you down in trying to fix this problem. |
| 13:00.15 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 13:00.16 | [TK]D-Fender | Sniper_micho: Sounds like something that it is your job to set manually. * can't know what service you are going to dial out from or who you should appear as. |
| 13:00.26 | Sniper_micho | ManxPower, That's why I'm asking |
| 13:00.43 | Strom_M | Sniper_micho: read this |
| 13:00.44 | Strom_M | ~101 |
| 13:00.44 | jbot | i guess 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 13:00.46 | ManxPower | Sniper_micho: we are not here to teach people telecom |
| 13:01.06 | Sniper_micho | ManxPower, No one asked you |
| 13:01.23 | ManxPower | You asked everyone |
| 13:01.42 | Sniper_micho | ManxPower, Please don't aswer the next time |
| 13:01.48 | ManxPower | I shall no. |
| 13:01.59 | Strom_M | Sniper_micho: good job biting the hand that feeds. |
| 13:02.01 | Sniper_micho | ManxPower, thanks |
| 13:02.15 | ManxPower | I really with Pidgin had a perm /ignore list. |
| 13:02.35 | [TK]D-Fender | ManxPower: Talks does funny Yoda hhmmmmmMMMMMM!?!?! |
| 13:02.57 | [TK]D-Fender | ManxPower: You need to go caffeinate :p |
| 13:03.00 | coppice | Makes ATAs, Yoda does |
| 13:04.29 | ManxPower | [TK]D-Fender: working on it. |
| 13:05.34 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:05.49 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 13:06.16 | Great_Anta_Baka | is there a command i can type to get help from jbot? |
| 13:06.26 | Great_Anta_Baka | or a list of all jbots commands? |
| 13:06.50 | ManxPower | Great_Anta_Baka: yes. |
| 13:07.21 | Great_Anta_Baka | could you give it to me please? |
| 13:07.31 | [TK]D-Fender | Great_Anta_Baka: PM "help" to jbot |
| 13:07.32 | ManxPower | <PROTECTED> |
| 13:07.51 | Great_Anta_Baka | ah thanks |
| 13:08.08 | Great_Anta_Baka | people are more helpful on here than what i am used to.. really weird feeling |
| 13:08.32 | mvanbaak | that's only today |
| 13:08.35 | ManxPower | Great_Anta_Baka: only if you are not asking stupid questions |
| 13:08.38 | mvanbaak | tomorrow everything will be back to normal |
| 13:08.38 | Great_Anta_Baka | waaa "jbot: No such nickname online." |
| 13:08.43 | Great_Anta_Baka | hahhaa |
| 13:09.03 | Great_Anta_Baka | ah |
| 13:09.04 | Great_Anta_Baka | i see why |
| 13:09.11 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:09.56 | [TK]D-Fender | Strom_M: Looks like your domain expired or got hijacked |
| 13:10.14 | ManxPower | If you come in here asking the Asterisk equiv of "I'm installing Apache and need help! What is HTML?" you'll see our wrath. |
| 13:10.21 | sysadmin-lb22 | Strom_M, about the rsync and nfs you just adviced me about...I need to setup two asterisk servers..in a way that if one users is registered on server A and another user is registered on server B and both servers read from the same configuraiton file ...server A knows what extensions are registered on server B..in another meaning both servers shoudl be aware of each other |
| 13:10.27 | Strom_M | ooh, expired |
| 13:10.34 | Strom_M | time to throw money at people |
| 13:10.53 | Great_Anta_Baka | yeah figured that.. being very careful what i ask... as the saying goes rtfm |
| 13:11.20 | ManxPower | sysadmin-lb22: what you want to do is impossible without lots of work. See the HA part of the Wiki |
| 13:11.33 | OcterA | Hi there |
| 13:11.35 | ManxPower | or "clustering" on the Wiki |
| 13:11.46 | *** join/#asterisk mbranca (n=matteo@81.208.92.210) |
| 13:12.06 | OcterA | How can I know if the DTMF that asterisk has received is inband or out of band? |
| 13:12.15 | ManxPower | OcterA: however you set it |
| 13:12.42 | OcterA | i can't read this in the asterisk console? |
| 13:13.15 | ManxPower | OcterA: SIP debug will show RFC2833 DTMF on the console |
| 13:14.07 | ManxPower | I think 1.4 has a DTMF debug option, but I've not used it. You just need to have faith that what you configure Asterisk for is what Asterisk will do, what you configure the phone for is what the phone will do. |
| 13:14.10 | Strom_M | OcterA: I thought you were certain you were receiving inband dtmf a few hours ago |
| 13:14.26 | Strom_M | even after I asked you to double-check |
| 13:15.00 | Strom_M | [TK]D-Fender: domain renewed...should be back to normal in a bit |
| 13:15.34 | OcterA | i'm certain, but i don't understand why a SIP phone that i have, send the both DTMF when I say to send only inband |
| 13:15.39 | [TK]D-Fender | Strom_M: ok, will check up on it as I'm adding the jbot info-let to me protected aqrchivewhen it is |
| 13:15.50 | ManxPower | OcterA: that is a setting on the phone. |
| 13:16.13 | Strom_M | ManxPower: ooh ooh, what do you want to bet that it's a grandstream |
| 13:16.22 | ManxPower | On phones I've used you set "auto mute audio when sending DTMF" setting or something like that. |
| 13:16.46 | ManxPower | or you just tell Asterisk to use rfc2833 DTMF and it will ignore the inband dtmf |
| 13:17.32 | ManxPower | Strom_M: I never understood why people have DTMF problems. We have not had DTMF problems in YEARS at my client locations. |
| 13:17.32 | OcterA | In the phone I have set DTMF inband, but in asterisk I got this : Receiving INFO! |
| 13:17.33 | OcterA | * DTMF-relay event received: 1 |
| 13:17.52 | ManxPower | OcterA: In what way do you think this is an Asterisk problem?? |
| 13:18.32 | OcterA | because with the same phone, the same config, on a x86 arch, I have no problem, but on a ARM the DTMF inband don't work |
| 13:18.32 | Strom_M | magic incantations and dead chickens |
| 13:18.54 | Strom_M | OcterA: can't you just set the phone to rfc2833 and be done with it? |
| 13:18.54 | ManxPower | Strom_M: none of those, just good design, I think. |
| 13:20.38 | OcterA | with the RFC2833 it's works fine, but I need the inband because we can phone to the server by an analog line via a analog/sip provider |
| 13:20.50 | hsv-al | hello d-fender |
| 13:21.19 | Strom_M | OcterA: the sip provider will handle the analog to rfc2833 conversion |
| 13:23.16 | OcterA | Strom_M : The provider only give me a analog number to be able to phone my server with a analog line |
| 13:23.33 | ManxPower | OcterA: You are not listening! |
| 13:23.37 | Strom_M | OcterA: you're completely missing the point |
| 13:23.59 | ManxPower | OcterA: The DTMF will be converted to inband whenever the call is sent over analog. It does not matter what format the DTMF starts out as. |
| 13:24.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:24.55 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:24.57 | ManxPower | If that is not happening then there is a bug or config issue. |
| 13:25.09 | Strom_M | or the provider blows |
| 13:26.01 | OcterA | ManxPower : sorry i don't understand what you say (i'm french) |
| 13:27.30 | sysadmin-lb22 | ManxPower, is there a way to store extensions and logged in users etc in a database ? |
| 13:28.13 | [TK]D-Fender | sysadmin-lb22: Yes there is. its all int he book. Go read. |
| 13:28.27 | coppice | anyone using a PAP2T for T.38? |
| 13:28.46 | sysadmin-lb22 | [TK]D-Fender, I will read about it but I am just getting my thoughts straight..just to know where i have to look..thanks for the advice |
| 13:28.53 | ManxPower | sysadmin-lb22: I have no idea, as I have never needed to use a database with Asterisk. There is really no point unless you are a service provider. |
| 13:29.22 | [TK]D-Fender | sysadmin-lb22: Read the book first, then ask for whats missing or not working quite right. |
| 13:29.23 | ManxPower | sysadmin-lb22: The Good Book does not mention this topic? |
| 13:29.30 | [TK]D-Fender | ManxPower: It does. |
| 13:29.45 | ManxPower | [TK]D-Fender: maybe skipped that part? |
| 13:30.04 | *** part/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 13:30.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:30.49 | sysadmin-lb22 | [TK]D-Fender, ManxPower thx..let me go for the book for the time being |
| 13:31.15 | ManxPower | Sad how many people come here asking questions about things covered in The Good Book |
| 13:31.57 | *** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku) |
| 13:32.22 | Great_Anta_Baka | by the good book are you referring to the orielly one? |
| 13:32.38 | [TK]D-Fender | Great_Anta_Baka: Yes |
| 13:33.49 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
| 13:35.52 | hsv-al | i have the PDF, but i got sick and tired of being glued to a computer |
| 13:35.58 | hsv-al | to get specifics/info |
| 13:36.02 | hsv-al | so I payed $45 at B&N heh |
| 13:37.10 | Strom_M | maybe you should have bought a dictionary too, so that you'd know that the past tense of "pay" is "paid" |
| 13:37.21 | Strom_M | but perhaps I'm just tilting at windmills here |
| 13:37.41 | hsv-al | typos are genurully utilized by eye are cee addicts |
| 13:37.59 | Strom_M | that's no typo :) |
| 13:38.27 | hsv-al | Just finished a large coffee, now it's time for 2 sugar-free red bulls |
| 13:38.34 | hsv-al | its a cheap/dangerous way to mask hunger when you dont eat solids :) |
| 13:38.57 | [TK]D-Fender | Strom_M: Keep charging "Don" ;) |
| 13:40.41 | hsv-al | just bought an N81 for $349 |
| 13:40.46 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.219) |
| 13:41.06 | hsv-al | http://www.plemix.com/phone-nokia-n81_8gb-phone |
| 13:42.03 | coppice | that's expensive |
| 13:42.08 | hsv-al | Theres a project community based to have it work in the u.s |
| 13:42.15 | hsv-al | hacking around to unlock it in the u.s |
| 13:42.28 | hsv-al | beautiful phone, puts the iphone/voyager to shame |
| 13:43.20 | hsv-al | coppice, specs: http://europe.nokia.com/A4494086 |
| 13:43.41 | coppice | i know what it is. that's a high price for it |
| 13:44.20 | *** join/#asterisk moy (n=moyhu@nat/ibm/x-d8f9936674df1cd1) |
| 13:44.32 | ManxPower | Can you make calls with it? |
| 13:44.33 | hsv-al | lots wont go below $450 |
| 13:44.40 | ManxPower | That's what I care about. |
| 13:44.42 | hsv-al | manxpower, when I get it in a few days |
| 13:45.00 | hsv-al | Same guys that wrote the dvd drive firmware hacks to play pirated Xbox 360 games |
| 13:45.02 | hsv-al | are working on this |
| 13:45.12 | coppice | it was about $270, the last time I looked in the shops here |
| 13:45.19 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 13:45.26 | ManxPower | I did finally find a use for the camera on my phone -- take pics of empty shelves at Walmant and send the pics to Walmart corporate |
| 13:47.02 | hsv-al | newer phones on nseries.com |
| 13:47.06 | hsv-al | N96 is sick, but to much |
| 13:47.17 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 13:47.41 | coppice | is the N96 the replacement for the N95? |
| 13:48.04 | hsv-al | dunno, havent looked at above N91 |
| 13:48.17 | hsv-al | 5megapixel in N96 |
| 13:48.33 | coppice | I know there is supposed to be a replacement in the pipeline |
| 13:52.58 | *** join/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 13:54.01 | hsv-al | coppice, N96 |
| 13:54.01 | hsv-al | EDGE Class 32, 296 kbps; DTM Class 11, 177 kbps |
| 13:54.01 | hsv-al | 3G HSDPA, 3.6 Mbps |
| 13:54.33 | coppice | only 3.6M? not the new 7.2M? tut, tut |
| 13:55.51 | hsv-al | well, your in hong kong heh |
| 13:55.57 | hsv-al | now I see why you think anything is slow |
| 13:56.02 | hsv-al | don't they have all the bleeding edge stuff there? |
| 14:00.31 | ManxPower | At least it's not satellite internet |
| 14:00.40 | ManxPower | I'm still in therapy after using that for 2 years. |
| 14:01.16 | Kobaz | anyone know if there is an equivalent of wanpipemon for analog sangoma cards |
| 14:01.33 | Kobaz | to see loop current status and etc |
| 14:02.10 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 14:04.15 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:04.53 | *** join/#asterisk xenonex (n=xenonex@89.218.235.221) |
| 14:06.08 | *** join/#asterisk Titanous (n=titanous@unaffiliated/titanous) |
| 14:06.16 | Titanous | recommendations for a Toll Free DID in Canada? |
| 14:09.52 | *** join/#asterisk Sniper_mich (n=michofr@mail.splendor.net) |
| 14:11.36 | *** join/#asterisk Torikun (n=386426@75-171-135-108.hlrn.qwest.net) |
| 14:11.38 | x86 | how long does it take a polycom IP330 to update the BootROM? |
| 14:11.40 | Torikun | Hello |
| 14:12.19 | x86 | I booted up the phone with the new sip.ld and bootrom.ld in the TFTP root, it grabbed them, and it's been saying "Updating BootROM. Please wait." for about 12 minutes now |
| 14:12.26 | x86 | is that normal? |
| 14:12.38 | Strom_M | x86: it doesnt sound completely out of the ordinary to me |
| 14:12.59 | x86 | ok |
| 14:16.12 | Strom_M | of course, just for reference, there's usually very little reason to update the bootrom |
| 14:18.02 | [TK]D-Fender | yup |
| 14:18.08 | [TK]D-Fender | SIP sure, but not bootROM |
| 14:19.44 | x86 | Polycom told me to update it |
| 14:20.00 | x86 | (both application and bootrom) |
| 14:20.17 | Strom_M | they told you specifically, or did they make a general announcement of some sort? |
| 14:20.47 | [TK]D-Fender | And what else are your Rice Crispies saying to you? ;) |
| 14:21.14 | Strom_M | Snap, Crackle, and Pop: a handbook of VoIP problems |
| 14:21.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:23.57 | x86 | well I have (3) IP330's that kept locking up for no apparant reason |
| 14:24.09 | x86 | so I called my reseller (VoIP Supply) and gave them all of the logs |
| 14:24.42 | x86 | they forwarded it on to Polycom, who told them to tell me to update both the bootrom as well as the application, as there is a known issue with the version the 330's ship with |
| 14:24.50 | Strom_M | ah, ok |
| 14:26.08 | _ShrikE | x86: what firmware did they have you go to? |
| 14:28.14 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 14:31.09 | x86 | spip_ssip_3_0_2RevC_release_sig.zip |
| 14:31.16 | x86 | and BootRom_4_1_0_release_sig.zip |
| 14:31.32 | x86 | bootrom is still updating |
| 14:31.36 | x86 | how long does it take? |
| 14:31.42 | x86 | been ~30 minutes now |
| 14:34.12 | coppice | anyone here using a PAP2T? |
| 14:35.52 | Torikun | I got linlksys pap2 |
| 14:35.59 | Torikun | using it with gsipphone |
| 14:36.08 | _ShrikE | coppice: I have a pap2t, but it shows no signs of T.38 support. |
| 14:36.14 | Torikun | same thing coppice ? |
| 14:37.18 | coppice | I was assured if you have a PAP2T and not a PAP2, and you have the latest firmware there is a T.38 option. I wonder if some kind of flush of the config is required after upgrading |
| 14:37.25 | krdian | is back. "Linux: Saving us from the Gates from Hell" |
| 14:37.27 | *** join/#asterisk ajohnstone (n=ajohnsto@host81-137-217-96.in-addr.btopenworld.com) |
| 14:37.36 | _ShrikE | I am running 3.1.15(LS) |
| 14:38.02 | krdian | hi |
| 14:40.33 | _ShrikE | looks like the latest firmware is 5.1.6, I'll load it up and see what I get. |
| 14:41.51 | krdian | anybody playing with conexant musycc cards ? |
| 14:42.51 | x86 | ok seriously how long does it take to update the firmware on a IP330? |
| 14:43.15 | _ShrikE | x86: a while, but not this long. |
| 14:44.07 | x86 | should I restart it? |
| 14:45.41 | Great_Anta_Baka | no |
| 14:45.48 | Great_Anta_Baka | i have updated the firmware on it |
| 14:45.53 | Great_Anta_Baka | but it didnt take that long |
| 14:46.03 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:46.16 | Great_Anta_Baka | what is it saying on the screen now x86? |
| 14:48.10 | x86 | _ShrikE: is it prov.polling.enabled that makes the polycom periodically check for new configs / firmware? |
| 14:48.40 | *** part/#asterisk Torikun (n=386426@75-171-135-108.hlrn.qwest.net) |
| 14:48.58 | *** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net) |
| 14:49.46 | Great_Anta_Baka | that should be in the user guide |
| 14:56.38 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.fr) |
| 14:59.01 | x86 | _ShrikE: I unplugged the power and now I'm going to try this again |
| 14:59.10 | x86 | this time, only the sip.ld |
| 15:04.20 | x86 | it loaded sip.ld just fine |
| 15:04.44 | x86 | sure is staying on the "IP address, Enet, Rev" screen for a while |
| 15:04.55 | *** join/#asterisk LuisTorres (n=chatzill@bl6-200-239.dsl.telepac.pt) |
| 15:04.58 | *** join/#asterisk draygon (n=draygon@gateway5-pnap.exigo.com) |
| 15:04.59 | x86 | but it shows Rev as 3.0.2.0972, which looks correct |
| 15:06.25 | *** join/#asterisk xacatecas (n=jkroon@dsl-242-253-90.telkomadsl.co.za) |
| 15:06.29 | x86 | still just sitting there at that screen |
| 15:06.36 | x86 | gah I bet it's locked up |
| 15:06.51 | x86 | IP330's are absolute crap imho |
| 15:07.31 | x86 | I'd much prefer an IP501 or even an IP301 over the IP330 |
| 15:08.05 | xacatecas | hi guys, when I've got two sip trunks to an upstream provider I start getting authentication errors between them. I need host= lines in both sip declarations and asterisk then forces all inbound sip requests from that IP into the one declaration. I need to either not require auth on those sip users, or i need to get asterisk to go through the normal username based authentication processes for both. any ideas? |
| 15:09.09 | [TK]D-Fender | xacatecas: why would you have *2* for the same provider? |
| 15:09.47 | xacatecas | in short: a really messed up provider from whom i just took over a reseller and i need to "pull" all his existing sip accounts onto my server. |
| 15:10.17 | xacatecas | the same providers i moved away from about 4 or 5 months ago for similar reasons. |
| 15:10.30 | *** join/#asterisk golumn (n=golumn@201.220.132.138) |
| 15:11.03 | xacatecas | so yea, it's two today, and possibly 50 more in a week. |
| 15:11.09 | x86 | still locked up... |
| 15:11.16 | x86 | gah this sucks |
| 15:11.31 | xacatecas | lol ... yea, that's usually when people hang out here? what's locked up? |
| 15:11.44 | x86 | IP330 |
| 15:11.52 | x86 | these things are so unstable |
| 15:11.56 | x86 | crap imho |
| 15:11.57 | hsv-al | heh, did the 411e's go up in price? |
| 15:12.00 | hsv-al | Price: $544.39 |
| 15:12.06 | x86 | 411e? |
| 15:12.09 | x86 | wtf is that? |
| 15:12.14 | hsv-al | TDM411E |
| 15:12.20 | xacatecas | SIP330? i threw mine out. |
| 15:12.25 | xacatecas | *WIP330 ? |
| 15:12.46 | [TK]D-Fender | xacatecas: he's referring to the Polycom IP 330 |
| 15:12.47 | x86 | Polycom IP330 |
| 15:13.13 | x86 | WIP330 is junk too? |
| 15:13.15 | golumn | Hi guys. I update asterisk from 1.4.19 to 1.4.20 and now the agi-bin scripts doesn't execute |
| 15:13.16 | xacatecas | ah ok. i worked with a polycom _once_, long, long ago. can't remember much from it except that it took me a few hours just to get it going. |
| 15:13.18 | xacatecas | jip, avoid. |
| 15:13.27 | x86 | my buddy tried the WIP300 and said it sucked |
| 15:13.38 | xacatecas | battery life is maybe three hours if you're lucky. i've seen as low as 5 minutes. |
| 15:13.48 | x86 | wow |
| 15:13.59 | golumn | the error is verbose "Failed to execute '/var/lib/asterisk/agi-bin/file.agi':No such file or directory" 2 |
| 15:14.03 | golumn | even if the file is in there |
| 15:14.05 | xacatecas | my one client that bought one just leaves it on usb permanently to prevent it from dying. |
| 15:14.09 | [TK]D-Fender | x86: most WiFi phones are garbage |
| 15:14.25 | xacatecas | golumn, permissions? what's the first line from the file? |
| 15:14.47 | xacatecas | jip. the alternatives is equally annoying. |
| 15:14.52 | xacatecas | *are |
| 15:15.00 | *** join/#asterisk draygon (n=draygon@gateway5-pnap.exigo.com) |
| 15:15.16 | golumn | xacatecas #!/usr/bin/php -q |
| 15:15.27 | xacatecas | does /usr/bin/php exist? |
| 15:15.45 | golumn | yes |
| 15:15.46 | x86 | [TK]D-Fender: my buddy worked for a 40-acre car dealership, and they wanted all of their sales people to have SIP phones on them at all times.... they ended up pissing on WiFi after trying 5 vendors, and now they are completely satisfied with a Polycom DECT solution |
| 15:16.01 | xacatecas | dos2unix /var/lib/asterisk/agi-bin/file.agi and then retry. |
| 15:16.23 | [TK]D-Fender | golumn: ls -la /var/lib/asterisk/agi-bin/file.agi |
| 15:16.23 | x86 | xacatecas: doubt that's the problem, since it worked before he upgraded heh |
| 15:16.35 | golumn | [TK]D-Fender, chmod is 777 |
| 15:16.46 | [TK]D-Fender | golumn: SHOW is precisely |
| 15:16.52 | [TK]D-Fender | it* |
| 15:16.52 | xacatecas | x86, i wouldn't know :). i haven't been hanging out here for a while. |
| 15:16.59 | xacatecas | agrees with [TK]D-Fender |
| 15:17.45 | golumn | -rwxrwxrwx |
| 15:18.42 | [TK]D-Fender | golumn: pastebin all of the CLI output, but * and Linux. |
| 15:18.59 | [TK]D-Fender | golumn: I don't want to see some hack-job paste whree I don't see all the ahrd proof. |
| 15:19.01 | golumn | ok |
| 15:19.17 | [TK]D-Fender | both* |
| 15:19.25 | [TK]D-Fender | bleh, can't type today... |
| 15:21.42 | xacatecas | http://lists.digium.com/pipermail/asterisk-dev/2007-January/025625.html <-- NO! |
| 15:22.29 | *** join/#asterisk tristanbob_ (n=tristanr@ubuntu/member/tristanbob) |
| 15:24.00 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 15:25.58 | xacatecas | ok, the trick is this: create an entirely separate [messed-up] context for sip with host=that_providers_ip and insecure=invite. This _must_ be the last declaration with host=that_providers_ip |
| 15:29.02 | *** join/#asterisk icel (n=dan@65.200.26.85) |
| 15:29.02 | *** join/#asterisk af_ (n=getsmart@88-149-241-109.dynamic.ngi.it) |
| 15:30.11 | icel | hey, how do I park a call? I did include => parkedcalls in extensions.conf and have parkext, parkpos, and context set up in features.conf |
| 15:31.19 | [TK]D-Fender | icel: Do like the instructions say, and do an attended transfer to the parking ext # |
| 15:31.20 | lmadsen | transfer to 700 |
| 15:31.25 | icel | ahh |
| 15:31.30 | icel | thx, i will try |
| 15:31.34 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:31.34 | lmadsen | attended transfer because it will tell you the parking spot |
| 15:31.36 | *** join/#asterisk galeras (n=galeras@200.118.210.42) |
| 15:31.59 | anonymouz666 | blind works too |
| 15:32.01 | galeras | Besides audiocodes, is mediatrix a good choice? |
| 15:32.07 | *** join/#asterisk Strom_M (n=pocketir@m7f0e36d0.tmodns.net) |
| 15:32.10 | x86 | why don't people just transfer to a user's direct extension? or even a queue? |
| 15:32.17 | x86 | galeras: for what? |
| 15:32.27 | icel | working, thx |
| 15:32.50 | galeras | x86: to connect 200 analog extensions to an asterisk box |
| 15:32.52 | icel | x86: just trying to get it to work for kicks. |
| 15:33.38 | x86 | galeras: I personally would recommend Carrier Access or Adtran |
| 15:34.00 | x86 | galeras: I would not recommend Rhino, and I've not used audiocodes nor mediatrix |
| 15:34.11 | x86 | galeras: I have not heard anything good about mediatrix though |
| 15:34.30 | x86 | galeras: you also might be interested in Xorcom, as you don't need a T1 card |
| 15:35.00 | galeras | Xorcom uses usb ports right? |
| 15:35.12 | Strom_M | t1 is probably a more solid choice than ip or usb for this sort of thing |
| 15:35.13 | x86 | their Astribank line does, yes |
| 15:35.21 | x86 | Strom_M: indeed |
| 15:35.45 | [TK]D-Fender | galeras: I've tried MediaTrix 1124 & AudioCodes MP-124. Both work fine |
| 15:36.31 | galeras | i like mediatrix, has good prices |
| 15:36.37 | galeras | Thanjs TK |
| 15:37.31 | [TK]D-Fender | Strom_M: Debateable. SIP is more redundant capable and saves you T1 & PCI issues, and the added costs associated with. |
| 15:37.55 | [TK]D-Fender | Zaptel FXS = bleh |
| 15:37.55 | [TK]D-Fender | only good for faxes if needed |
| 15:38.48 | Strom_M | fender: perhaps, but any modern server nakes the t1 pci argument moot. it was a relevant argument four years ago. |
| 15:39.09 | galeras | thanks x86 |
| 15:39.43 | [TK]D-Fender | Strom_M: there are several server make out there that are still hostile, like HP's as we've seen crop up lately. |
| 15:40.22 | [TK]D-Fender | Strom_M: Just more things you don't need to deal with. SIP takes all the load off of *, and saves you the cost of the card in the first place and gives you backup capabilities |
| 15:40.26 | bootc | [TK]D-Fender: really? any details? |
| 15:40.52 | bootc | I'm looking at getting some HP servers for my * system... |
| 15:40.56 | [TK]D-Fender | bootc: look in the archives going back the past week or two. You'll see about guys with serious issues |
| 15:41.19 | hi365 | other than the randomly hosted and unmaintained ones, there arent really any rpm's for asterisk. how come? |
| 15:41.30 | [TK]D-Fender | bootc: With Digium cards anyways, not sure how Sangoma would pan out, and thats what I suggested he try next. |
| 15:41.38 | bootc | mail list archives or irc logs? |
| 15:41.49 | [TK]D-Fender | hi365: because you aren't making them. |
| 15:41.50 | hi365 | does it run better if its compiled? is it just a pain to maintain? |
| 15:42.02 | [TK]D-Fender | bootc: IRC for sure, probably some ML's out there as well |
| 15:42.09 | [TK]D-Fender | hi365: LOL! |
| 15:42.27 | [TK]D-Fender | hi365: jsut compile from source like the rest of us. |
| 15:42.48 | russellb | there are rpms for asterisk |
| 15:42.54 | russellb | they are in fedora now |
| 15:42.55 | x86 | bootc: I'm running HP ProLiant DL380 G5's in 7 different offices, and I've not had any problems with the hardware at all |
| 15:43.05 | hi365 | i have no problem with source - im just wondering if there are any benifits to not using rpm's |
| 15:43.07 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 15:43.16 | x86 | bootc: except for a failed power supply, but they have two in them so no biggie |
| 15:43.27 | x86 | hi365: RPM's are going to be ancient |
| 15:43.29 | bootc | x86: that's great to hear, are you using Sangomas or Digiums or something else? |
| 15:43.53 | x86 | hi365: source is always best... if you read the destructions, it's fairly straightforward to compile 1.4 yourself |
| 15:43.58 | hi365 | russellb: who maintains them? why arent there link form asterisk.org? |
| 15:44.01 | x86 | much easier now than 1.2 was |
| 15:44.06 | russellb | someone else maintains them ... |
| 15:44.17 | x86 | bootc: Sangoma |
| 15:44.26 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:44.41 | bootc | x86: very good news indeed, thanks, that's the exact setup I'm planning |
| 15:44.42 | russellb | you shouldn't have any problems with new digium cards, either. |
| 15:44.45 | hi365 | x86: for the second time - i have no problem with source. im just wondering WHY? mysql is in rpm, apatche is in rpm's so why not asterisk |
| 15:44.50 | x86 | bootc: most servers have a single A104D-x (A104 in a PCIe form factor with hardware EC) |
| 15:45.35 | x86 | hi365: they are out there, but generally not nearly up to date... same with mysql and apache, chances are you wont find the latest stable version in an RPM |
| 15:45.51 | hi365 | ah. ok |
| 15:47.09 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:47.59 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 15:48.13 | tzafrir_laptop | x86, your distro normally gives you an up-to-date apache. not bleeding edge, but with the critical fixes |
| 15:48.46 | tzafrir_laptop | In fact, doing this on your own would take much more time, and chances are you'd forget to do that |
| 15:49.02 | icel | is there something weird that you have to do to initiate a transfer on a snom 360? |
| 15:49.18 | icel | The transfer button works, but using # doesn't |
| 15:50.00 | Strom_M | use the transfer button |
| 15:50.10 | [TK]D-Fender | icel: You're supposed to use your phone's transfer button. "#" is something you have to enabel when you dial |
| 15:50.10 | Strom_M | # is a nasty hack |
| 15:50.20 | icel | k |
| 15:50.22 | x86 | tzafrir_laptop: that's not my point |
| 15:50.28 | icel | thx |
| 15:50.30 | x86 | tzafrir_laptop: not my point at all ;) |
| 15:50.45 | x86 | tzafrir_laptop: generally you'll not find anything close to being recent as an RPM |
| 15:51.02 | tzafrir_laptop | x86, sure you can. But it will never be stable |
| 15:51.12 | tzafrir_laptop | look for "backports" |
| 15:51.35 | tzafrir_laptop | Or rebuild yourself. That's easy |
| 15:51.47 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:52.03 | x86 | tzafrir_laptop: ugh, nevermind you're missing my point |
| 15:52.11 | tzafrir_laptop | But why exactly do you need bleeding edge on a production system? |
| 15:53.39 | tzafrir_laptop | In fact, the latest released tarball is not exactly bleeding edge. You can always get svn trunk |
| 15:54.17 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:54.17 | tzafrir_laptop | e.g.: you're not realy using latest stable release of speex |
| 15:56.55 | x86 | I said NOTHING about bleeding edge |
| 15:57.05 | x86 | in fact, I said explicitly "latest stable" |
| 15:58.39 | spokra | tzafrir_laptop: I think what x86 is saying is compile it!! if you want help here.. if you cant get * working from source you have no tight running it!! learn it!! |
| 15:58.59 | spokra | s/tight/right/ |
| 16:00.07 | tzafrir_laptop | spokra, that approch will cause damage in the long run, IMHO |
| 16:00.14 | spokra | hehehe jbot is correcting my typos!! |
| 16:00.43 | spokra | is this your first *? |
| 16:01.29 | tzafrir_laptop | Using mysql from source is plain silly |
| 16:01.42 | tzafrir_laptop | The distribution builds it nrmally better than you |
| 16:01.47 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:01.59 | tzafrir_laptop | Likewise apache |
| 16:02.05 | outtolunc | notes except for session duration <G> |
| 16:02.07 | tzafrir_laptop | At least with a proper distribution |
| 16:02.15 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 16:02.42 | spokra | if you take the class provided by digium thay tell you to compile it from source. You need to understand what is going on in * |
| 16:02.50 | jblack | Hmmm. I suspect that one of my clients is going to want to have a conversation about POE. |
| 16:02.59 | x86 | you talking about the dCap bootcamp? |
| 16:03.08 | spokra | yes |
| 16:03.13 | x86 | jblack: PoE rules |
| 16:03.22 | x86 | spokra: have you taken that? |
| 16:03.25 | spokra | yes |
| 16:03.40 | *** join/#asterisk inverted (n=codya@staff2.aus.datafoundry.com) |
| 16:03.41 | x86 | so you're dCap certified? |
| 16:04.01 | spokra | didn't take the test.. |
| 16:04.13 | coppice | isn't the term dCapitated? |
| 16:04.17 | hsv-al | I need some fried spokra for lunch |
| 16:04.23 | jblack | x86: Hopefully more than their current wall outlets. They're DOA this morning due to no power. |
| 16:04.39 | inverted | anyone know why I am getting the following error? the asterisk server is natted, in dmz, the phones are natted as well: chan_sip.c:11296 handle_request_register: Registration from '<sip:2006@192.168.1.172>' failed for '192.168.1.87' - ACL error (permit/deny) |
| 16:05.11 | tzafrir_laptop | spokra, that's because "building from source is more convinient for the poeple doing single installations |
| 16:05.15 | [TK]D-Fender | inverted: Clearly you have set a range in your sip peer and the call isn't coming in from an allowed place |
| 16:05.37 | tzafrir_laptop | It is something you have to know. But it does not mean you actually have to practice it. |
| 16:05.37 | hsv-al | "ACL error (permit/deny)" |
| 16:06.07 | tzafrir_laptop | You also take a course of assembly language as part of learning how to program |
| 16:06.19 | spokra | tzafrir_laptop: go ahead and use a package!! just don;t ask for help because the first thing you will be asked is if you compiled from source.. |
| 16:06.42 | inverted | i havent specified any acls or ranges, in fact i even went in and for each peer added a permit=x.x.x.x/255.255.255.0 |
| 16:06.45 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 16:06.49 | kannan | hello all |
| 16:06.52 | jblack | spokra: What is this you're talking about? |
| 16:07.36 | spokra | debating runnigng * for a package or compileing from source.. |
| 16:07.49 | tzafrir_laptop | spokra, I know well enough how to make packages. And if I have spare time I do support here people who use Asterisk from packages. Fair enough? |
| 16:07.54 | icel | Hey, is the Asterisk Bootcamp course worthwhile? |
| 16:08.22 | icel | how advanced does it get? |
| 16:08.24 | jblack | That's up to you. I stick to packages because I can easily replication identical versioning across machines. It also depend upon packaging to let me know when there's security updates and such. |
| 16:08.31 | spokra | tzafrir_laptop: then build yourself a package!! and reuse it!! |
| 16:09.45 | spokra | icel: if your just starting Dcap bootcamp is worth it.. if you have done it awhile.. ship the boot camp and take the 2nd course.. you will have to still take the bootcamp test to be able to take 2nd class |
| 16:09.56 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 16:10.08 | spokra | s/ship/skip/ |
| 16:10.45 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:11.08 | icel | spokra: ok. I can set up and configure asterisk although I haven't tested all of the features. I configured a voice T1 line just fine, but Im still a little sketch on data T1 usage and different codecs. |
| 16:11.17 | icel | spokra: so you think the 2nd class is better then for me? |
| 16:12.01 | x86 | Asterisk doesn't care about a data T1 |
| 16:12.18 | x86 | Asterisk does nothing with a data T1 at all |
| 16:12.22 | icel | x86: I just don't know how to configure everything in the grand scheme of things |
| 16:12.31 | x86 | ah |
| 16:12.37 | icel | x86: You can put a data T1 connection in tho, right? |
| 16:12.37 | spokra | 90% of the bootcamp would be waisted on someone that has built a asterisk system. if you can get a system running and do dialplans t1's dundi etc.. goto the 2nd class.. |
| 16:12.47 | icel | spokra: thx |
| 16:12.59 | icel | x86: but you fhave to find someone to terminate your lines |
| 16:13.13 | icel | x86: and that is where I get confused cuz I haven't done it before. |
| 16:13.38 | nny_1 | so... hints for BLF. I got them working when someone dials the extension, or calls in to it, but when i dial outwards from a phone, they blf on the sidecar stays green.. I have a [hints] context with (ex: exten => 10,hint,SIP/10).. any advice? |
| 16:15.59 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:21.53 | *** join/#asterisk southtel (n=southtel@24-240-24-20.dhcp.gwnt.ga.charter.com) |
| 16:22.54 | mort_gib | nny_1: What type do you have the devices as in sip.conf |
| 16:22.58 | southtel | Can anyone recommend some good windows network testing software? |
| 16:23.00 | *** join/#asterisk zenobic (n=zenobic@174-204-116-85.dsl.manitu.net) |
| 16:23.31 | southtel | We'rve recently changed our ISP at a remote office, and now we can't get those phones to connect to our * server. |
| 16:23.38 | [TK]D-Fender | southtel: "ping" |
| 16:23.44 | southtel | (I'm installing nmap at the moment) |
| 16:23.56 | mort_gib | TK : he he |
| 16:24.03 | southtel | Thanks. |
| 16:24.12 | icel | southtel: You could use wireshark or something to sniff the traffic if that is what you are after |
| 16:24.32 | southtel | Yeah, that did occur to me...I was hoping to not have to go to that level... |
| 16:24.47 | southtel | This feels like a NAT issue, but I just can't prove it. |
| 16:24.58 | nny_1 | mort_gib: friend |
| 16:25.22 | [TK]D-Fender | nny_1: "type=peer" , "call-limit=99" |
| 16:25.42 | nny_1 | peer would be the missing part, thanks [TK]D-Fender, mort_gib |
| 16:25.45 | icel | southtel: try traceroute then. You should be able to deduce a NAT error from that |
| 16:25.56 | zenobic | hi. is iax2 encryption "Packet Decrypt Failed" hungup a bandwith problem? |
| 16:27.05 | icel | southtel: i think it is tracert in windows. At least it will let you know if it can route. |
| 16:27.55 | southtel | Well, I know we can reach the * server from the office, because they're getting their configs from it via ftp. |
| 16:33.20 | icel | southtel: not sure what to tell you then since I don't what network hardware you have. IF it is routing between the remote site to your * server, then NAT seems like the most likely problem |
| 16:34.41 | southtel | Understandable...mostly I just would like to find some windows tools to do the sort of tests I'd normally do in linux. |
| 16:34.59 | golumn | are there changes in 1.4.20. My 1.4.19 scripts aren't working. I got them recognize but they are giving unrecognize command and stuff like that |
| 16:37.27 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 16:38.15 | DarKnesS_WolF | anyone using snom phones ? |
| 16:38.24 | tripps | anyone have a good SMS gateway service they can recommend? |
| 16:39.41 | hi365 | is it really nesesary to run ./configure and make menuselect befor compiling? (i.e. what happens if i dont?) |
| 16:40.11 | DarKnesS_WolF | hi365: configure is a must , make menuselect u can escpe it if u want the default make |
| 16:40.12 | [TK]D-Fender | hi365: Go ahead & try and see what happens |
| 16:40.14 | tripps | hi365, yes |
| 16:40.21 | DarKnesS_WolF | but u have to do ./configure make and make install |
| 16:40.43 | *** join/#asterisk JackEStorm (n=no@ip70-171-83-146.no.no.cox.net) |
| 16:40.48 | hi365 | cool. thanks to thoes that are HELPFULL |
| 16:47.29 | russellb | menuselect is option |
| 16:47.36 | russellb | s/n/nal/ |
| 16:47.41 | russellb | dangit |
| 16:47.43 | russellb | it's optional. |
| 16:47.44 | russellb | :) |
| 16:47.49 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:47.56 | DarKnesS_WolF | guys i real have a small question and very strange .... |
| 16:47.59 | DarKnesS_WolF | russellb: may be u can help :-) |
| 16:48.06 | russellb | i know nothing |
| 16:48.14 | DarKnesS_WolF | i didn't ask yet :P |
| 16:48.33 | DarKnesS_WolF | a client want to pressa buttn and get a tone from asterisk then he dials a number and go via zap channel |
| 16:48.47 | russellb | *CLI> core show application DISA |
| 16:48.49 | sysreq` | disa? |
| 16:48.50 | DarKnesS_WolF | the problem is i can't dial the zap directly cuz in that case the cdr will be missed up. |
| 16:49.02 | DarKnesS_WolF | russellb: i did it with waitexten and read |
| 16:49.29 | DarKnesS_WolF | russellb: but the porblem he has many zap channels and he want to assing each button to an channel with same way |
| 16:49.36 | *** join/#asterisk tiav (n=tiav@ivr94-4-82-227-121-53.fbx.proxad.net) |
| 16:49.42 | DarKnesS_WolF | too much messing around with cotexts and so |
| 16:49.46 | DarKnesS_WolF | any other better / lceaner idea ? |
| 16:50.04 | russellb | i don't understand the question |
| 16:52.49 | DarKnesS_WolF | russellb: mmmm ur right i'm so wasted :-) let me try to clear it up . is there a way that after i get the dialtome from the PSNT i still can get the DTMF i am sending to the PSTN line via the zap channel ? |
| 16:52.56 | russellb | ... |
| 16:53.28 | lmadsen | ... indeed |
| 16:53.31 | russellb | come back after you sober up |
| 16:53.35 | DarKnesS_WolF | lol |
| 16:53.36 | [TK]D-Fender | Grammar very is english good your |
| 16:53.39 | *** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net) |
| 16:53.45 | Juggie | hah |
| 16:53.52 | hsv-al | fender what ask you question him that for?? |
| 16:53.53 | DarKnesS_WolF | [TK]D-Fender: thx mate ;) |
| 16:54.01 | russellb | runs off for a bit |
| 16:54.09 | lmadsen | runs after russellb |
| 16:54.09 | [TK]D-Fender | runs in circles |
| 16:54.09 | DarKnesS_WolF | lmadsen: how ? |
| 16:54.16 | lmadsen | how what? |
| 16:54.23 | lmadsen | doesn't remember making a statement |
| 16:54.25 | [TK]D-Fender | Who's on first? |
| 16:54.33 | lmadsen | What's on second? |
| 16:54.35 | outtolunc | looks up the word run |
| 16:54.43 | Dabba | wonders if anyone from digium in watching, as he had a support consultancy paid up for 17:30BST but no one called |
| 16:54.43 | DarKnesS_WolF | to get the DTMF to the zap channel i get it |
| 16:54.58 | lmadsen | you just press a button on the phone, which sends DTMF to the line... |
| 16:55.02 | DarKnesS_WolF | Dabba: yes lots od digium are around . |
| 16:55.35 | DarKnesS_WolF | lmadsen: :-) ok look i dial a button on the phone which is programed to dial exten => 111,1,Dial(ZAP/1) |
| 16:55.42 | lmadsen | right... |
| 16:55.52 | DarKnesS_WolF | so i get teh PSTN tone and i start to dial a normal number 345543345 |
| 16:55.57 | lmadsen | right... |
| 16:56.01 | DarKnesS_WolF | now in the CDR of asterisk teh dst is 111 |
| 16:56.05 | lmadsen | yes |
| 16:56.16 | DarKnesS_WolF | anyway i can get the DTMF that the user sent to replace the dst ? |
| 16:56.25 | jameswf-home | has a marshmellow gun and thinks productivity will suck today |
| 16:57.09 | lmadsen | exten => _XXX.,1,Dial(ZAP/1/${EXTEN}) |
| 16:57.36 | DarKnesS_WolF | mmmm |
| 16:57.43 | lmadsen | or capture with a Read() and send to that extension |
| 16:57.46 | kannan | if i build addons , will the cdr-csv stop automatically? |
| 16:57.49 | DarKnesS_WolF | will not work |
| 16:57.52 | DarKnesS_WolF | lmadsen: i did that already |
| 16:57.54 | lmadsen | if you need to simulate dialtone first |
| 16:58.02 | DarKnesS_WolF | the problem is they are like 12 PSTN lines |
| 16:58.05 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:58.09 | DarKnesS_WolF | each button should go to one zap channel |
| 16:58.40 | DarKnesS_WolF | lmadsen: i used WaitExten and it worked ,, and i used read and it worked , but i'm sure there is another more cleaner way to do it |
| 16:58.42 | lmadsen | create a group |
| 16:58.59 | lmadsen | Dial(Zap/g1/${EXTEN}) |
| 16:59.16 | DarKnesS_WolF | can't make group |
| 16:59.20 | DarKnesS_WolF | need each channel sperated |
| 16:59.21 | lmadsen | DarKnesS_WolF: you can't capture the DTMF on the zap line and replace it in the CDR |
| 16:59.29 | lmadsen | why? |
| 16:59.46 | DarKnesS_WolF | cuz actually each line spearted for some useing . like one line to call home , one for work one for mobiles and so on |
| 16:59.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 16:59.54 | lmadsen | and you could probably create a loop with a ChanIsAvail() or something in there to check the line and loop through until you find a free one |
| 16:59.59 | DarKnesS_WolF | lmadsen: yes that what i thought |
| 17:00.20 | lmadsen | you need to create some clever dialplan magic because you're trying to do something clever |
| 17:00.49 | DarKnesS_WolF | lmadsen: i know but the whole issue it is one of this distro thing :-s not something i build from scratch like i always do |
| 17:00.53 | DarKnesS_WolF | thats sucks |
| 17:01.05 | lmadsen | it totally does |
| 17:01.08 | lmadsen | good luck with that |
| 17:01.44 | DarKnesS_WolF | lmadsen: haha thx ;-) |
| 17:01.49 | DarKnesS_WolF | i thought about something |
| 17:01.54 | lmadsen | try not to |
| 17:01.57 | lmadsen | you'll just hurt yourself |
| 17:02.03 | [TK]D-Fender | lol |
| 17:02.13 | [TK]D-Fender | I was wondering what that burning smell was |
| 17:02.14 | DarKnesS_WolF | lmadsen: if i did somehting like 111,1,Dial(ZAP/1/${EXTEN}) |
| 17:02.20 | DarKnesS_WolF | here EXTEN == 111 right :D? |
| 17:02.25 | lmadsen | yes |
| 17:02.32 | [TK]D-Fender | DarKnesS_WolF: .... you want to dial "111" on the PSTN? |
| 17:02.35 | DarKnesS_WolF | lmadsen: see :D i need sleep :-) |
| 17:02.40 | [TK]D-Fender | DarKnesS_WolF: SOBER UP <----- |
| 17:02.42 | lmadsen | yep -- go to bed and try again tomorrow |
| 17:02.52 | lmadsen | you're just going to wreck something |
| 17:03.06 | lmadsen | think be to clearly for hurt not system |
| 17:03.31 | lmadsen | many sleeps customer happy yes system excited! |
| 17:03.41 | DarKnesS_WolF | [TK]D-Fender: no i don't want to do that .. |
| 17:03.49 | DarKnesS_WolF | lmadsen: true true :P |
| 17:03.59 | lmadsen | aha... well I don't want to help anyone either... so have fun :) |
| 17:04.04 | lmadsen | errr... anymore :) |
| 17:04.05 | DarKnesS_WolF | lmadsen: u think i need english lessons :P? |
| 17:04.20 | lmadsen | nope.. I think you need to go to sleep so you can think clearer |
| 17:04.51 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:05.02 | DarKnesS_WolF | lmadsen: yes true :-) thx for the help anyway and thx [TK]D-Fender , russellb and the rest :) |
| 17:05.35 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
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| 17:22.21 | *** join/#asterisk ruied_ (n=ruied@89-180-250-84.net.novis.pt) |
| 17:27.24 | *** join/#asterisk galeras (n=galeras@200.118.210.42) |
| 17:27.53 | galeras | Which is the maximum distance to connect an analog extension? |
| 17:29.10 | *** join/#asterisk AlexTO (n=alex@216.215.174.155.nw.nuvox.net) |
| 17:30.55 | [TK]D-Fender | galeras: depends on the equipment you're plugging it into. |
| 17:31.05 | anonymouz666 | let's say PAP2. |
| 17:31.17 | *** join/#asterisk nicox (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 17:31.20 | [TK]D-Fender | Go check that devices tech specs. |
| 17:31.24 | anonymouz666 | nice |
| 17:31.55 | anonymouz666 | PAP2 does not even have a decent manual. |
| 17:33.25 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 17:36.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
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| 17:39.34 | *** join/#asterisk wonderworld (n=service@dslb-088-076-088-238.pools.arcor-ip.net) |
| 17:40.45 | wonderworld | hi, i am trying to connect a Digium TE122 to a german E1. I am not able to get the card out of yellow alert. my zaptel.conf is here -> http://pastebin.com/m3b925722 . any ideas on how to debug this? thanks a lot. |
| 17:43.45 | jameswf-home | wonderworld: have you tried the proper cable |
| 17:44.03 | [TK]D-Fender | wonderworld: Is this a brand new card? |
| 17:44.10 | wonderworld | yes it is |
| 17:44.35 | [TK]D-Fender | wonderworld: Then make sure you have set the jumper on it to E1 as it defaults to T1 |
| 17:44.51 | wonderworld | i jumpered it to E1 (jumper is set) |
| 17:45.25 | jameswf-home | wonderworld: if your not using one try an RJ48c cable this is NOT the same as an RJ45 crossover |
| 17:45.35 | AlexTO | wonderworld: Check if the cable is a correct T1 Cross over |
| 17:45.49 | AlexTO | T1/E1 |
| 17:46.41 | wonderworld | ok thanks, ill check the cable first |
| 17:47.10 | jameswf-home | loopbacks are good for this too.. connect a loopback if the alarm clears.... |
| 17:48.47 | *** join/#asterisk CaRb0n^ (n=playa@203.81.221.240) |
| 17:50.45 | AlexTO | There is someone familiar with CDR on MySQL i want set my CDRs into a MySQL Database |
| 17:51.43 | jblack | Yes, there are many familiar |
| 17:51.48 | x86 | is that a question? |
| 17:53.15 | AlexTO | Great, well for any reason the CDRs are not recording the data into the BD, so I already set the cdr_mysql.conf file and also i have the DB with teh table ready with privilegies |
| 17:53.18 | *** join/#asterisk CVirus (n=GoD@196.205.192.192) |
| 17:53.46 | AlexTO | but for me looks lke there is something missing on linux to set up |
| 17:54.19 | jblack | If you had cdr_mysql right, the db and tables right, and priviledges right, it would work. |
| 17:54.31 | jblack | make sure you restart asterisk and mysql. |
| 17:55.00 | AlexTO | it should be, but obviusly there is something that i'm missing or i have wrong |
| 17:55.19 | AlexTO | my db is not onthe same machine |
| 17:55.45 | jblack | perhaps you forgot to get mysql to listen to the nic. |
| 17:56.02 | jblack | It's really a guessing game. Try making mysql-client work with the credentials you set up first. |
| 17:56.34 | AlexTO | I already test it and I can connect to tha BD using that Credentials |
| 17:57.17 | jblack | From the server that's running asterisk? |
| 17:57.19 | AlexTO | This is the error that i get ERROR[23369]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server |
| 17:57.57 | AlexTO | no, my asterisk is running in a different box |
| 17:58.25 | jblack | Go to the server running asterisk, run mysql-client there, and make connections work with mysql-client. |
| 17:58.38 | AlexTO | so there are some tips that you can tell me to dobble check? |
| 17:58.55 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 17:58.56 | jblack | I feel like I'm talking to a broken record. |
| 17:59.17 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 17:59.37 | jameswf-home | just run all mysql clients as root you only live once |
| 18:00.26 | *** join/#asterisk nicox_ (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 18:00.27 | jameswf-home | little johnny tables we call him |
| 18:00.28 | AlexTO | i'm ussing shell but i'm not familiar with mysql-client under shell commands |
| 18:01.07 | jameswf-home | mysql -u root -pSOOPERSEEkRET asterisk |
| 18:01.26 | jameswf-home | souperseekret |
| 18:02.08 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 18:02.58 | *** join/#asterisk csum77 (n=chatzill@c-71-195-222-192.hsd1.ut.comcast.net) |
| 18:03.13 | AlexTO | in my case will be: mysql [MySQL Server IP] -u root -pSupersecret asterisk ? |
| 18:03.26 | CaRb0n^ | my inbound recordings works fine but my outbound recordings are getting mixed up between both side voice, |
| 18:03.39 | CaRb0n^ | my asterisk version is 1.4.19 |
| 18:03.49 | CaRb0n^ | do i need to do something with somix? |
| 18:10.48 | *** join/#asterisk twelveca (n=twelve@cx-246.cx.voinetworks.net) |
| 18:11.42 | [TK]D-Fender | CaRb0n^: If you don't mix, how does each side sound? |
| 18:14.10 | CaRb0n^ | i mean |
| 18:14.18 | CaRb0n^ | what i hear in the recording is |
| 18:14.34 | CaRb0n^ | that my agent started talking and the other side is still ringing |
| 18:14.42 | CaRb0n^ | but in live calls its really not like that |
| 18:16.07 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 18:16.34 | AlexTO | jamesswf-home: that is what i get http://pastebin.com/m9cf0d45 |
| 18:16.42 | kannan | AlexTO , in your MySQL box, type mysql -p .. |
| 18:17.28 | *** part/#asterisk galeras (n=galeras@200.118.210.42) |
| 18:17.31 | AlexTO | Ask me for password |
| 18:17.37 | kannan | type the pwd |
| 18:17.57 | AlexTO | im in |
| 18:18.07 | AlexTO | I can see the databases too |
| 18:18.31 | kannan | kewl , is the root pwd the same as you used in mysql database, password for user root |
| 18:18.47 | AlexTO | with: show databases; |
| 18:20.05 | AlexTO | kannan, i'm ussing a different user to record the cdr into the BD asterisk |
| 18:20.25 | kannan | oh ok |
| 18:20.50 | kannan | AlexTo , then if you set the cdr_mysql params , you start logging the CDRs |
| 18:21.09 | kannan | you have to create the cdr table in the database first |
| 18:21.22 | AlexTO | i already have it set up |
| 18:21.27 | kannan | the samples config has a the create cmd i think |
| 18:22.04 | AlexTO | if i type mysql [Mysql server ip] -uroot -p |
| 18:22.13 | AlexTO | ask me for password too |
| 18:22.22 | AlexTO | but do not connect :-( |
| 18:22.28 | kannan | ok , but it asks for the root pwd i think |
| 18:22.47 | [TK]D-Fender | AlexTO: Go ask in #mysql |
| 18:23.22 | AlexTO | OK, i'll thanks anyway |
| 18:28.11 | ccvp | --13:27:50-- http://downloads.digium.com/pub/telephony/firmware/releases/zaptel-fw-tc400m-MR6.12.tar.gz |
| 18:28.17 | ccvp | when was that up'd? |
| 18:29.16 | DarKnesS_WolF | AlexTO: mysql -u root -ppassword -h server ip as i recall |
| 18:30.02 | *** join/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-c57c7ffab9b5c77c) |
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| 18:38.14 | *** part/#asterisk nicox_ (n=nicox@vie-nas-ge-0-2.onenet.at) |
| 18:42.31 | hsv-al | http://pastebin.com/m7a85bfeb |
| 18:46.04 | Qwell | hsv-al: "up'd".. that's a term I haven't heard in years |
| 18:46.23 | hsv-al | heh |
| 18:47.00 | SteveTotaro | can someone help me with trixswitch .2 ;-) |
| 18:47.15 | SteveTotaro | ~trixswitch |
| 18:48.10 | Qwell | SteveTotaro: I think you're looking for... |
| 18:48.13 | Qwell | ~happyclownpbx |
| 18:48.13 | jbot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for it's core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 18:48.46 | SteveTotaro | the damn thing keeps segfaulting at boot |
| 18:49.44 | SuPrSluG | start pulling cards |
| 18:50.13 | hsv-al | libcurl3-openssl-dev on ubuntu |
| 18:50.22 | hsv-al | acceptable replacement for openssl-dev? |
| 18:50.27 | Qwell | no |
| 18:50.38 | Qwell | you want libssl-dev |
| 18:51.30 | SteveTotaro | no cards to pull |
| 18:51.41 | SteveTotaro | it is bare hp dl380 |
| 18:51.55 | SteveTotaro | the iso installs selinux by default |
| 18:52.07 | SteveTotaro | maybe i alter the iso |
| 18:52.29 | SteveTotaro | it segfaults on selinux |
| 18:53.38 | *** join/#asterisk mike-ekim (n=mike@adsl-072-151-207-108.sip.mia.bellsouth.net) |
| 18:54.27 | mike-ekim | Hi, I have 3 asterisk servers, one incoming, and two outgoing |
| 18:54.40 | mike-ekim | what happens if in both outgoing servers I have the same outgoing sip providers for outgoing calls |
| 18:55.09 | mike-ekim | would they both cause registration issues with eachother? If so, what is the best way to load balance these boxes so that if outgoing-callbox01 would use outgoing-callbox02 if it cannot complete any calls? |
| 18:55.15 | [TK]D-Fender | mike-ekim: Shouldn't matter |
| 18:55.32 | [TK]D-Fender | mike-ekim: registering typically has nothing to do with outgoing calls |
| 18:55.35 | mike-ekim | no? what do you mean |
| 18:55.36 | mike-ekim | ooh |
| 18:55.44 | mike-ekim | so they both can have the saame outgoing sip providers in the sip.conf file? |
| 18:55.54 | [TK]D-Fender | mike-ekim: usually |
| 18:56.10 | mike-ekim | gotcha. of course as long as the IP matches, and/or user-pass are correct ? |
| 18:56.20 | [TK]D-Fender | mike-ekim: Clearly required. |
| 18:56.43 | mike-ekim | right on. |
| 18:56.45 | mike-ekim | one more thing... |
| 18:57.00 | jeev | Fender, i'm gonna move the phone to the 2mbit up line today to see if i still get static.. but i'll need to test for about a month ;< |
| 18:57.13 | mike-ekim | what would be the best way to configure asterisk to make a call out of another asterisk-outgoing box on the same network if one outgoing-callbox fails? |
| 18:58.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:58.26 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 18:59.30 | SteveTotaro | FYI selinux=0 on the kernel line in grub disables selinux at bootup |
| 19:03.26 | [TK]D-Fender | mike-ekim: dial them back to back |
| 19:04.53 | spokra | mike-ekim: setup dundi, use DNS to round robin the phone so the phones register with one or the other servers. when a server fails the phones on the failed system will re-register. you will have to setup ixa trunking between the servers so the phones can all connect internaly to each other |
| 19:05.57 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 19:06.41 | *** join/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 19:12.45 | Larisa | ~book |
| 19:12.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 19:24.32 | *** join/#asterisk ManxPower (n=manxpowe@86.sub-70-221-48.myvzw.com) |
| 19:27.17 | ManxPower | ~mailinglist |
| 19:27.18 | jbot | well, mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 19:29.03 | ManxPower | jbot: your zipper is down |
| 19:29.04 | jbot | ManxPower: okay |
| 19:30.35 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 19:32.35 | hsv-al | pbx_ael.so reminds me of Aeon Flux |
| 19:32.41 | hsv-al | O.O |
| 19:33.10 | russellb | that makes no sense. |
| 19:33.24 | russellb | jbot_: your zipper |
| 19:33.27 | hsv-al | i know, staring at a screen hours on hours makes one go insane |
| 19:33.28 | hsv-al | :) |
| 19:33.39 | russellb | ~your zipper |
| 19:33.39 | jbot | your zipper is probably down |
| 19:37.15 | jameswf-home | has killed 4 employees with my ninja marshmellow gun sniping skills |
| 19:37.46 | jameswf-home | jbot_: marshmellow guns are the coolest things ever |
| 19:37.48 | *** join/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 19:38.43 | nny_1 | any reason why a polycom would jump to context "default" instead of the posted "context=sip" in sip.conf when type is set to "peer" vs. "friend" ? |
| 19:39.06 | [TK]D-Fender | nny_1: No. Pastebin the complete call w/ SIP debug & your peer config. |
| 19:39.19 | nny_1 | k |
| 19:39.25 | Strom_C | and also the relevant section of extensions.cong |
| 19:39.27 | Strom_C | er, conf |
| 19:39.50 | [TK]D-Fender | Strom_C: Shouldn't actually need that. |
| 19:40.17 | Strom_C | no, but I'm curious if my initial theory is correct ;) |
| 19:46.17 | spokra | anyone know of any good docs on adhearsion? |
| 19:46.41 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:50.12 | *** join/#asterisk Psychobilly (n=moi@adsl89-118.kln.forthnet.gr) |
| 19:50.42 | patrick-- | Is it possible to perform an action as soon as a sip client logs onto the asterisk sip service? |
| 19:51.34 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 19:52.11 | *** join/#asterisk nicox (n=nicox@213-33-5-230.adsl.highway.telekom.at) |
| 19:54.03 | *** join/#asterisk Amun (n=Jonsi@c-71-205-221-174.hsd1.mi.comcast.net) |
| 19:55.31 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:55.35 | ManxPower | spokra: I think the people that make Super Glue have some |
| 19:55.54 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 19:56.08 | ManxPower | patrick--: Off the top of my head, I imagine you can use a hint priority and give it a Local/ channel. |
| 19:56.09 | spokra | manxPower: you should be a comidian.. :> |
| 19:56.49 | patrick-- | huh? |
| 19:57.01 | patrick-- | i want to perform an action when a SIP client signs on |
| 19:57.04 | ManxPower | patrick--: have you ever used hints for BLF? |
| 19:57.28 | patrick-- | whats BLF |
| 19:57.34 | ManxPower | patrick--: nevermind |
| 19:57.52 | hsv-al | progdocs error, paste bin url: |
| 19:57.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:58.00 | hsv-al | http://pastebin.com/m4b2d7976 |
| 19:58.02 | hsv-al | any idea? |
| 19:58.22 | patrick-- | ManxPower: care for a serious answer? |
| 19:58.33 | bbryant | hsv-al: install doxygen |
| 19:58.41 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 19:59.36 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 20:00.36 | russellb | and install graphviz, too |
| 20:00.41 | russellb | if you want sexy graphs in the docs :) |
| 20:00.49 | russellb | or ... just go to http://www.asterisk.org/doxygen/ |
| 20:02.09 | patrick-- | russellb: do you know if it's possible to perform a certain action when a sip client signs on? |
| 20:02.24 | russellb | there is a manager event when that happens |
| 20:02.26 | russellb | so, yes |
| 20:02.28 | [TK]D-Fender | ManxPower: Nifty approach, I like it... |
| 20:03.58 | patrick-- | nice ;) thx |
| 20:04.36 | BCS-Satori | Correct me if I am wrong, but Asterisk as a RTP gateway briding the SIP Trunk and Phone? Or does the Trunk connect directly to the Phone on RTP, and bypass Asterisk? |
| 20:04.53 | SteveTotaro | ~siptrunk |
| 20:05.06 | spokra | depends |
| 20:05.19 | patrick-- | siptrunk? |
| 20:05.38 | SteveTotaro | ~trunk |
| 20:05.38 | jbot | from memory, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 20:05.51 | [TK]D-Fender | BCS-Satori: * is a B2BUA <----- |
| 20:06.11 | [TK]D-Fender | BCS-Satori: it is not a proxy. |
| 20:06.14 | SteveTotaro | asterisk is a pbx |
| 20:07.00 | SteveTotaro | ~asterisk |
| 20:07.01 | jbot | methinks asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 20:07.30 | ManxPower | Asterisk is a PBX Toolkit, allowing to build a PBX. |
| 20:07.39 | ManxPower | Trixbox is a "PBX" |
| 20:08.12 | spokra | if you are looking for a sip proxy look at openser |
| 20:08.14 | SteveTotaro | ~asterisk |
| 20:08.15 | jbot | methinks asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org, or just like a mini-mall |
| 20:08.26 | SteveTotaro | jbot is always right |
| 20:08.46 | Qwell | ~lart SteveTotaro |
| 20:08.46 | jbot | slaps a compatible dib on SteveTotaro's head |
| 20:09.33 | BCS-Satori | So does a B2BUA, control RTP? Basically heres is my problem. Clients are complaining they can not hear people on the other end saying that they are too quiet. When they turn their phone up all the way it reverbs. I am curious if it is possible to amplify the RTP at asterisk and send it to the phone, yes i know it may distort, but I am curious if it is even possible. |
| 20:10.00 | SteveTotaro | are you using PSTN hardware? |
| 20:10.06 | SteveTotaro | are you using asterisk? |
| 20:10.16 | hsv-al | why is there a reference to microsoft*/leetspeak.mspx in the book HEH |
| 20:10.21 | SteveTotaro | do you have cheap a$$ phones? |
| 20:10.23 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-181-rrdg-esr-2.dynamic.isadsl.co.za) |
| 20:10.23 | ManxPower | BCS-Satori: Gain is done at the PSTN/VoIP interface. |
| 20:10.29 | Qwell | hsv-al: eh? |
| 20:10.37 | hsv-al | page 50, bottom left |
| 20:10.42 | BCS-Satori | SteveTotaro: just asterisk, registered to a voip account, and they are using Linksys SPA962, so yes cheap |
| 20:10.58 | ManxPower | BCS-Satori: you can change the gain on the SPA |
| 20:11.14 | disposable | what's a good usable gui/webui for a large scale asterisk installation? i don't care if it's commercial, it's a case study. (over 1000 extensions) |
| 20:11.18 | ManxPower | i.e/ at a pstn/voip interface, where this "pstn" is a single phone. |
| 20:11.23 | BCS-Satori | ManxPower: problem is when we do it reverbs in the microphone, and causes audio to be lost, its a know issue linksys tech's are working on |
| 20:11.28 | spokra | BCS-Satori: all calls that go to the sip provider are this way? is the server behind a nat firewall? |
| 20:11.30 | ManxPower | disposable: as we don't do GUIs here.... |
| 20:11.47 | ManxPower | BCS-Satori: Asterisk cannot change the volume in the rtp stream |
| 20:12.05 | spokra | make that ManxPower: |
| 20:12.08 | BCS-Satori | spokra: yes all the calls go via sip provider, and yes we are behind a nat |
| 20:12.30 | spokra | are all call missing audio to the provider or just some? |
| 20:12.48 | ManxPower | BCS-Satori: you are having a "volume too low" problem, not a "can't hear audio in one direction" problem, right? |
| 20:12.57 | BCS-Satori | spokra: it appears all external calls are quiet, this is an issue all clients complain about |
| 20:12.59 | SteveTotaro | the console is a gui of sorts |
| 20:13.08 | ManxPower | BCS-Satori: talk to your ITSP |
| 20:13.21 | BCS-Satori | ManxPower: yes volume is too low, but when we turn volume up on the phone it reverbs back into the phone |
| 20:13.37 | spokra | if it's no audio check your firewall.. if it's low audio talk to your itsp |
| 20:13.42 | SteveTotaro | if they are running asterisk, tell them to turn up the rxgain in their zap config |
| 20:13.42 | BCS-Satori | ManxPower: /laugh thats what i'm leaving them theres 3 providers infront of them i found out. |
| 20:13.54 | SteveTotaro | sorry PAHDI config |
| 20:14.49 | SteveTotaro | chan_pahdi.conf |
| 20:15.09 | *** join/#asterisk Geert (i=geert@irssi/staff/geert) |
| 20:15.47 | Geert | Does somebody know where I can get an up-to-date list of global dialplan? For example 0032 = belgium 00324* = belgium mobile |
| 20:16.53 | *** join/#asterisk asdx (n=diego@adsl-141-8.click.com.py) |
| 20:16.59 | asdx | hi guys |
| 20:17.21 | *** join/#asterisk deeperror (n=deeperro@adsl-76-226-148-247.dsl.sfldmi.sbcglobal.net) |
| 20:17.51 | asdx | is there a way i could extract some asterisk information, and put it somewhere (web, xml, database, etc), information like calls being made, etc |
| 20:18.13 | Geert | you can store CDR in mysql |
| 20:18.36 | Geert | and for real asterisk info, you can use the API |
| 20:18.44 | deeperror | [TK]D-Fender: Got the sangoma today, When connecting to channel banks should all ports be set to master for timing? |
| 20:19.13 | asdx | is there xml-rpc support? |
| 20:19.37 | asdx | what about agi/ami? |
| 20:19.48 | codefreeze-lap | Geert: yep, use the mysql in asterisk-addons |
| 20:19.54 | Geert | asdx: http://www.voip-info.org/wiki-Asterisk+manager+API |
| 20:20.07 | Qwell | asdx: use the manager interface |
| 20:20.08 | asdx | thanks |
| 20:20.11 | asdx | ok |
| 20:21.12 | Geert | codefreeze-lap: that logs cdr in mysql, that does not give me a current list of global dialplans/destinations |
| 20:21.23 | SteveTotaro | ~pb |
| 20:21.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:21.26 | *** part/#asterisk XnOSX (i=4de20eec@gateway/web/ajax/mibbit.com/x-c57c7ffab9b5c77c) |
| 20:21.55 | codefreeze-lap | Geert: true |
| 20:22.04 | *** join/#asterisk swiftkick (n=Miranda@mail.beanproducts.com) |
| 20:23.13 | SteveTotaro | http://pastebin.ca/1037189 |
| 20:23.17 | SteveTotaro | lol |
| 20:23.22 | SteveTotaro | I like it |
| 20:23.51 | SteveTotaro | gives asterisk more validity and i can undercut and give more features |
| 20:24.27 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 20:25.48 | hsv-al | crc_ccitt = ? |
| 20:25.52 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 20:26.52 | pigpen | Hi all, I know that my question is not a voip one, but I know that there are -so- many smart people. |
| 20:27.14 | pigpen | I need to solder a couple of wires together, but one of them is kinda greasy. |
| 20:27.24 | voxter | wow.. really? |
| 20:27.32 | voxter | I thought i'd seen it all.. haha |
| 20:27.38 | pigpen | and of course, the solder won't stick. How to I clean it up so solder will stick without destroying the wire? |
| 20:27.40 | patrick-- | http://phpfi.com/321487 why wont this show me all the sip peers conencted? |
| 20:27.43 | pigpen | voxter, heh, yeah. |
| 20:27.59 | voxter | pigpen: wash it in some detergent in a cup? |
| 20:28.10 | pigpen | hmm..line simple green? |
| 20:28.14 | voxter | sure. |
| 20:28.48 | pigpen | not a bad idea. I only have 1.5" left of wire, so I have very limited attempts. |
| 20:29.33 | patrick-- | Hey, i want to catch an event where a certain sipuser signs on and perform an action in that case. how can i possibly fix that? |
| 20:29.58 | *** join/#asterisk hsv-al (n=ccvp@66.0.46.210) |
| 20:30.00 | pigpen | patrick--, tail your logs. |
| 20:30.14 | ManxPower | patrick--: Your search of the mailing list archives was not helpful? |
| 20:30.24 | patrick-- | ManxPower: i did not find anything |
| 20:30.33 | patrick-- | pigpen: i have the feeling that tailing logs is not the thing to do |
| 20:30.56 | ManxPower | Speaking of the mailing list, there are at least 10,000 people on asterisk-users last I heard. Sounds like that would be a better pool then the 256 users here. |
| 20:31.57 | voxter | patrick--: I'll give you at least a lead to follow since everyone else seems to be against it. |
| 20:32.06 | voxter | voxter: You probably want to look at using the Manager API |
| 20:32.09 | voxter | oops |
| 20:32.12 | voxter | patrick |
| 20:32.13 | voxter | :) |
| 20:32.18 | patrick-- | voxter: i am already using it |
| 20:32.36 | voxter | patrick--: So you should be able to receive messages about SIP peers that become available/register |
| 20:32.44 | voxter | patrick--: when you receive those messages, "do something" |
| 20:33.11 | hsv-al | bleh, pulseaudio in infinite loop |
| 20:33.38 | patrick-- | but /q voxter |
| 20:34.51 | AlexTO | jamesswf-home u there? |
| 20:35.28 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 20:35.38 | AlexTO | Someone familiar with CDR on MySQL...?/ Im back |
| 20:36.16 | AlexTO | I already test the mysql-client connected to MySQL server and connects fine |
| 20:37.04 | AlexTO | but i'm still getting this error Jun 2 16:35:11 ERROR[2689]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server 64.25.XXX.XX |
| 20:40.41 | *** part/#asterisk nny_1 (n=Scott_My@64.203.239.83) |
| 20:42.11 | AlexTO | there someone familiar with CDRs on Mysql? i need a litle help to finish my setup |
| 20:42.45 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585011.dsl.bell.ca) |
| 20:44.07 | hsv-al | is it normal for asterisk -vvvc to go black console background? |
| 20:45.59 | _ShrikE | AlexTO: I am sure you already checked your settings in cdr_mysql.conf and they are correct. |
| 20:46.24 | _ShrikE | AlexTO: that said can you provide a tcpdump of a call attempt |
| 20:52.05 | *** join/#asterisk sp00k3y (n=chatzill@74.202.4.2) |
| 20:56.48 | ManxPower | hsv-al: I've never seen it do anything else |
| 20:57.10 | _ShrikE | loves it when folks ask for help and then dont respond |
| 20:58.34 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:00.49 | *** part/#asterisk hubguruJR (n=hubguruJ@mail.ntegratedsolutions.com) |
| 21:04.57 | AlexTO | Shireke, I make the mysql directly from shell and i could fime |
| 21:04.59 | AlexTO | fine |
| 21:06.51 | AlexTO | <PROTECTED> |
| 21:07.26 | AlexTO | what else should i take care to make the connection |
| 21:14.31 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 21:14.44 | _ShrikE | AlexTO: That is what the packet capture will tell us. |
| 21:18.36 | Larisa | hola |
| 21:19.04 | Larisa | who speak spanish ? |
| 21:19.15 | AlexTO | yo |
| 21:19.15 | Larisa | hola |
| 21:19.23 | AlexTO | ShrikeE |
| 21:19.33 | _ShrikE | ? |
| 21:20.01 | AlexTO | well basically the error said ERROR[6077]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql: cannot connect to database server 64.25.XX.XX |
| 21:20.09 | AlexTO | Hola larisa |
| 21:20.31 | Larisa | Hola AlexTO me puedes contestar en privado profavor ? |
| 21:20.50 | AlexTO | Ok.. |
| 21:24.59 | AlexTO | ShirikE u there? |
| 21:26.14 | _ShrikE | Yes, but that error alone is not enough for me to help you. |
| 21:27.10 | ManxPower | Golly, Beave, it sure would be nice if app_agi GARE YOU A FRICKEN ERROR MESSAGE when the script is not found. |
| 21:27.22 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:28.04 | AlexTO | let set highest debug level |
| 21:30.43 | AlexTO | ShirikE, i increased the Level but the error is the same |
| 21:31.17 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 21:32.03 | AlexTO | How can i make to debug the MySQL error? |
| 21:34.01 | _ShrikE | I told you. Provide a packet capture to file that we can look at. |
| 21:34.24 | *** join/#asterisk xnixan (n=xnixan@196.218.222.117) |
| 21:37.22 | *** join/#asterisk Mikeonline (i=Mike@p57A7E30E.dip.t-dialin.net) |
| 21:37.26 | Mikeonline | hi |
| 21:37.58 | Strom_M | hi |
| 21:38.00 | Mikeonline | is there a "powered by asterisk" logo or sth else out there for usage? |
| 21:38.15 | Strom_M | Mikeonline: talk to Digium's marketing department |
| 21:38.37 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:38.40 | Mikeonline | there is no site ready for use like at apache/php? |
| 21:38.53 | Strom_M | Mikeonline: have a look at www.digium.com |
| 21:39.02 | Mikeonline | iam looking :) |
| 21:39.26 | Strom_M | and no, I'm not sure there's anything like "powered by asterisk" |
| 21:39.40 | Strom_M | which is why I'm suggesting you talk to their marketing department |
| 21:41.35 | AlexTO | ShikeE, sorry but i dont know how do that, but if you explain to m that i can provie it |
| 21:42.19 | _ShrikE | AlexTO: something like... |
| 21:42.54 | _ShrikE | tcpdump -i eth0 -s 0 port 3306 and host 64.25.xx.xx -w /tmp/capture.cap |
| 21:43.26 | _ShrikE | substituting eth0 for your interface if needed |
| 21:44.09 | *** join/#asterisk ftoledo (n=ragnarok@201.250.49.67) |
| 21:46.41 | ftoledo | hi all |
| 21:46.59 | ftoledo | i have a my asterisk worning fine with a grandstream ata ht488 |
| 21:47.52 | ftoledo | but i have a little issue, when play the voicemail saved messages, isd sound with more speed that was recorded , is a little speed difference |
| 21:48.24 | ftoledo | any idea to begin my research? |
| 21:53.09 | *** join/#asterisk deeperror (n=deeperro@d149-67-253-63.try.wideopenwest.com) |
| 21:53.24 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
| 21:53.31 | *** part/#asterisk mprebello (n=marcel@c90696a5.static.spo.virtua.com.br) |
| 21:54.46 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:01.42 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 22:15.36 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 22:15.58 | seanbright | is there a way to: |
| 22:16.05 | seanbright | a) remotely reboot polycoms |
| 22:16.17 | seanbright | b) remotely force them to re-read configuration |
| 22:16.19 | seanbright | ? |
| 22:17.01 | _ShrikE | seanbright: sip notify polycom-check-cfg %peer% |
| 22:17.07 | _ShrikE | from the asterisk cli |
| 22:17.09 | seanbright | almost too easy |
| 22:17.11 | seanbright | thanks |
| 22:17.12 | seanbright | :) |
| 22:17.19 | _ShrikE | np |
| 22:18.07 | tripps | <PROTECTED> |
| 22:18.09 | tripps | heh |
| 22:18.13 | tripps | oops |
| 22:19.08 | seanbright | _ShrikE: worked like a charm. thanks again. |
| 22:21.18 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-82-206.vif.net) |
| 22:22.05 | *** join/#asterisk Dovid (n=Dovid@bzq-79-177-102-237.red.bezeqint.net) |
| 22:26.57 | Mikeonline | Strom_M: btw thanks for your response. i havent found anything than contacting as you said but that will waste too much time. |
| 22:27.03 | Mikeonline | good night all |
| 22:27.09 | Strom_M | waste time? |
| 22:27.17 | Strom_M | it'll take you...what...15 minutes? |
| 22:27.18 | Mikeonline | yes |
| 22:27.25 | Strom_M | you've spent more than that just poking around online |
| 22:27.27 | Mikeonline | but the response will take hours |
| 22:27.35 | Mikeonline | and i have to get work done today :) |
| 22:27.41 | Strom_M | god forbid you should plan or anything |
| 22:28.06 | Mikeonline | if i get the response some hours later ill have to start again.. |
| 22:28.17 | voxter | Mikeonline: you could... make your own, you know. |
| 22:28.18 | Mikeonline | and i havent searched all the time btw :P |
| 22:28.24 | Mikeonline | voxter? |
| 22:28.32 | voxter | Mikeonline: you need a powered by asterisk logo, right? |
| 22:28.48 | Mikeonline | i need a permission to use the logo |
| 22:28.53 | voxter | Mikeonline: take the asterisk logo itself, open it in an image editor, and add the words "powered by" to it |
| 22:29.13 | Strom_M | Mikeonline: digium publishes their trademark policy |
| 22:29.14 | voxter | you're free to use the asterisk logo wherever you want, as long as you give the appropriate credit |
| 22:29.23 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 22:29.26 | [TK]D-Fender | ~asterisktrademark |
| 22:29.30 | jbot | [~asterisktrademark] See http://www.digium.com/en/company/view-policy/5 for Digium's trademark policy. |
| 22:29.30 | Strom_M | Mikeonline: perhaps you should read the trademark policy |
| 22:29.43 | Mikeonline | i have read.. and i think there is no way to use |
| 22:29.57 | Strom_M | Mikeonline: so CALL THEM for fuck's sake |
| 22:30.01 | [TK]D-Fender | Mikeonline, then stop whining and live with it :p |
| 22:30.07 | Mikeonline | i already do :P |
| 22:30.12 | Strom_M | seriously, the whining is annoying |
| 22:30.15 | Strom_M | do something |
| 22:30.24 | Mikeonline | ya ya ya |
| 22:30.30 | Mikeonline | slaps Strom_M around a bit with a large trout |
| 22:30.32 | Mikeonline | ;-) |
| 22:31.08 | Mikeonline | Strom_M: voxter.com google maps api key is registered for another website. |
| 22:31.22 | voxter | Mikeonline: ...... |
| 22:31.25 | Strom_M | huh? |
| 22:31.25 | voxter | Mikeonline: so? |
| 22:31.25 | Mikeonline | if you belong to.. |
| 22:31.37 | Mikeonline | so the map wont load |
| 22:31.37 | voxter | Yes, its my site. Thank you for pointing out a random known fact. |
| 22:31.46 | Strom_M | how is this relevant to me? |
| 22:31.50 | voxter | Strom_M: its not. |
| 22:31.53 | Mikeonline | maybe you want to correct it? |
| 22:31.59 | voxter | Mikeonline: not really. thanks though. |
| 22:32.10 | Strom_M | i thought you didn't have time to screw around on the internet, Mikeonline. I thought you had to get work done today !!!! |
| 22:32.50 | Mikeonline | i need a creative minute *lol* |
| 22:32.58 | Mikeonline | and you give me inspiration |
| 22:33.08 | voxter | wanders off |
| 22:33.41 | Qwell | voxter: the reason that was pointed out, is because maps enabled sites don't work correctly when used with an improper key |
| 22:33.56 | Qwell | s/correctly/at all/ |
| 22:35.04 | voxter | Qwell: <- [hC] |
| 22:35.17 | Qwell | just sayin' :p |
| 22:35.27 | voxter | Qwell: i know why he pointed it out.... i just dont understand why he went to my website to begin with :) |
| 22:35.40 | *** join/#asterisk erojasv (n=erojasv@190.40.53.84) |
| 22:37.18 | Strom_M | maybe he was looking for the digium marketing department |
| 22:37.40 | Mikeonline | voxter: i even dont know it too :) |
| 22:48.51 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
| 22:49.01 | teknoprep | hey all... i am getting very odd random 1-way audio problems |
| 22:49.35 | teknoprep | here's the thing its not to the PBX.. becuase if it was a 1-way audio to the PBX which is behind a NAT... the user that is calling would never be able to press 1 or 2 to get passed the IVR |
| 22:49.55 | teknoprep | but.. here is the thing... when the call gets to the phone.. ANY phone.. sometimes i get 1-way audio |
| 22:50.24 | [TK]D-Fender | teknoprep, No.... usually DTMF is passed through SIP, not RTP, and NAT can still be the issue |
| 22:50.26 | drmessano | I prefer the "Powered by Most Popular Open Source PBX Engine" logo |
| 22:50.37 | teknoprep | [TK]D-Fender, yeah it feel's like a NAT issue |
| 22:50.46 | teknoprep | [TK]D-Fender, i have multiple subnets i need NOT to nat for |
| 22:50.54 | [TK]D-Fender | "* : The PBX Toolkit formerly known as Asterisk" |
| 22:50.55 | teknoprep | [TK]D-Fender, i have each one listed in its own ... localnet = |
| 22:50.56 | drmessano | So use localnet |
| 22:51.11 | teknoprep | i did |
| 22:51.13 | drmessano | Then your masks are wrong or something |
| 22:51.17 | teknoprep | dude |
| 22:51.23 | teknoprep | localnet = 10.0.0.0/255.0.0.0 |
| 22:51.26 | [TK]D-Fender | teknoprep, pastebin it all. |
| 22:51.34 | teknoprep | localnet = 192.168.0.0/255.255.0.0 |
| 22:51.34 | teknoprep | ok |
| 22:52.51 | teknoprep | http://pastebin.com/m15631869 |
| 22:53.24 | [TK]D-Fender | Oh god, freePBX... |
| 22:53.56 | teknoprep | heh |
| 22:54.05 | [TK]D-Fender | teknoprep, What have you got forwarded to *, and pastbin THEIR entries as well. |
| 22:54.22 | teknoprep | 1:1 NAT -> externip -> 10.100.100.97 |
| 22:54.29 | teknoprep | i have everything forwarded for now |
| 22:54.30 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 22:54.34 | teknoprep | its a 1:1 nat |
| 22:54.41 | teknoprep | through m0n0wall |
| 22:55.20 | teknoprep | i have a specific IP just for the PBX. which is forwarded to a private VLAN ... VLAN 100 which is the 10.100.100.0/24 network |
| 22:55.40 | teknoprep | do my subnets need to match actual subnets in the sip_nat.conf ? |
| 22:55.41 | [TK]D-Fender | teknoprep, .... PASTEBIN <----- |
| 22:55.48 | teknoprep | PASTEBIN WHAT ? |
| 22:55.54 | teknoprep | its a 1:1 nat |
| 22:55.59 | [TK]D-Fender | <[TK]D-Fender> teknoprep, What have you got forwarded to *, and pastbin THEIR entries as well. <------- |
| 22:56.10 | [TK]D-Fender | teknoprep, the PEER ENTRY. |
| 22:56.15 | [TK]D-Fender | sheesh |
| 22:56.26 | teknoprep | oh i thought you meant port entry inside of the firewall it was behind |
| 22:57.20 | teknoprep | i don't understand the question... what i have forwarded to asterisk ? |
| 22:57.23 | [TK]D-Fender | teknoprep, And while you're at it, add a call attempt with sip debug enabled. |
| 22:57.36 | teknoprep | yeah i tryed that.. the thing is to catch it when it happens |
| 22:57.54 | [TK]D-Fender | teknoprep, I asked 2 things. 1 : what ports do you have forwarded to *. 2 : show me the PEER entry for the account whose calls are screwing up. |
| 22:58.24 | teknoprep | i mixed up the 2 questions... ok i will show you the PEER info ... and i have already said EVERYTHING is forwarded.. brb |
| 22:58.24 | [TK]D-Fender | teknoprep, and now, 3. A call against that peer. |
| 22:58.37 | *** part/#asterisk disposable (i=disposab@blackhole.sk) |
| 22:58.41 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 23:00.30 | *** join/#asterisk nicox (n=nicox@213-33-5-230.adsl.highway.telekom.at) |
| 23:01.43 | teknoprep | [TK]D-Fender, http://pastebin.com/m90932fd |
| 23:02.24 | [TK]D-Fender | teknoprep, your VP peers should be "nat=NO" <-- |
| 23:02.40 | teknoprep | ahh |
| 23:02.48 | teknoprep | may i ask why ? |
| 23:02.58 | [TK]D-Fender | teknoprep, because THEY are not behind NAT. |
| 23:03.05 | teknoprep | ahh |
| 23:03.08 | teknoprep | i am .. not them |
| 23:03.12 | [TK]D-Fender | teknoprep, let them do their own thing. |
| 23:03.44 | teknoprep | could that cause this problem ? |
| 23:05.50 | [TK]D-Fender | teknoprep, Could be when you are mangling IP's you're not supposed to. |
| 23:06.59 | teknoprep | [TK]D-Fender, ? |
| 23:07.06 | teknoprep | [TK]D-Fender, it is random |
| 23:07.10 | teknoprep | [TK]D-Fender, very random |
| 23:07.40 | teknoprep | [TK]D-Fender, well lets see if this fixes it.. if it does.. thanx man |
| 23:07.42 | teknoprep | again |
| 23:07.51 | *** join/#asterisk RoyK (n=Roy@ip-105-50-149-91.dialup.ice.no) |
| 23:08.01 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal) |
| 23:10.28 | *** join/#asterisk LiNeTuX (n=LiNeTuX@171.117.8.67.cfl.res.rr.com) |
| 23:16.05 | *** part/#asterisk ftoledo (n=ragnarok@201.250.49.67) |
| 23:22.04 | Mikeonline | what the hell |
| 23:22.13 | Mikeonline | i saw the lightning |
| 23:22.18 | Mikeonline | after many seconds the thunder came.. |
| 23:22.28 | Mikeonline | lasting for many seconds wow :) |
| 23:22.53 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 23:23.10 | voxter | Qwell: i dont suppose that your chan_skinny supports conferencing (3 way calling) via a "conf" soft button on the phone, does it? |
| 23:23.23 | drmessano | Mikeonline: If I was an op, I would kick you |
| 23:23.33 | drmessano | Mikeonline: Reason? NO Haiku! |
| 23:26.30 | Mikeonline | whats a haiku? a japanese tale? |
| 23:27.07 | Mikeonline | @drmessano |
| 23:27.19 | drmessano | What is a haiku? |
| 23:27.28 | Mikeonline | yes? |
| 23:27.32 | drmessano | It is a poem of a sort |
| 23:27.41 | Mikeonline | but its the truth :) |
| 23:27.49 | drmessano | If you must know this |
| 23:28.16 | Mikeonline | so now its time to say goodbye |
| 23:28.34 | Mikeonline | bye bye |
| 23:28.45 | drmessano | Are you 12? |
| 23:29.13 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 23:30.06 | drmessano | What a retard |
| 23:30.40 | drmessano | He needs a "Powered by Special Olympics" banner for his page |
| 23:31.59 | CVirus | that sounds mean |
| 23:32.10 | drmessano | It probably is |
| 23:32.28 | CVirus | you're making fun of people with physical disablities |
| 23:32.38 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 23:33.04 | CVirus | I would have kicked you if I had the privilege to do so |
| 23:34.21 | drmessano | I would have kicked you for exhibiting a total lack of backbone and taking the internet seriously.. But i'll let Darwinism take care of that.. |
| 23:35.51 | [TK]D-Fender | Funny... I thought the "Special Olympics" were for the MENTALLY challenged, and the Para-lympics for the "differently-abled" |
| 23:36.49 | CVirus | http://madduck.net/blog/2008.01.31:linux-for-visually-impaired/ ... the blind Debian developer |
| 23:39.23 | LiNeTuX | What's the best Music-On-Hold (either from mp3's or streams) setup I should look into? |
| 23:40.25 | drmessano | [TK]D-Fender: Ok, then what is the "Internet" for? Both? |
| 23:42.19 | [TK]D-Fender | linuxmaniac_away, recordings in the native codec of your channels |
| 23:43.25 | *** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk) |
| 23:54.40 | disposable | can i run on SIPp on the same asterisk box i want to test? |
| 23:54.58 | disposable | s/run on SIP/run SIP/ |
