| 00:00.58 | dlynes | coppice: is that what it was? I would think it was they forgot to turn some things on for most builds...those devices are supposed to support it, according to the product documentation |
| 00:01.19 | dlynes | coppice: are you interested in which hardware version and firmware version I've got where it actually works? |
| 00:01.49 | coppice | works as in "has config options" or works as in "runs all day with no trouble"? |
| 00:02.11 | coppice | dlynes: which devices are supposed to support it? |
| 00:02.45 | dlynes | coppice: all pap2 devices, according to their product packaging, manuals, and not to mention their web configuration |
| 00:02.57 | drmessano | <PROTECTED> |
| 00:03.20 | coppice | the PAP2 and PAP2T according to all documentation do NOT support T.38 |
| 00:03.28 | hesco | there was a question last night from someone seeking a syntax highlighter for writing an asterisk dialplan. Has anyone written one for vim, yet? |
| 00:03.47 | dlynes | coppice: what documentation are you referring to? |
| 00:03.57 | dlynes | hesco: vim already has one |
| 00:04.13 | hesco | and thanks dlynes and [TK]D-Fender for the feedback on the perl -wc equivalent |
| 00:04.17 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 00:04.25 | hesco | oh yeah? where would I find it, any idea? |
| 00:04.36 | coppice | anything. the manual, the ads, the general documentation for the linksys product line which lists the PAP2 and PAP2T as exceptions every time T.38 is mentioned |
| 00:04.40 | dlynes | hesco: assuming your version of vim is recent enough to have it, it should be included |
| 00:05.09 | dlynes | coppice: is that for the newer hardware versions then? |
| 00:05.17 | hesco | I'm using whatever came with Debian Lenny a few months ago when I rebuilt this desktop |
| 00:05.19 | dlynes | coppice: afaik, the pap2t doesn't support it |
| 00:05.41 | hesco | no syntax highlighting in my /etc/asterisk/* files, though |
| 00:06.03 | coppice | the PAP2T is a cost reduction of the PAP2. it looks like they added no functionality |
| 00:06.07 | dlynes | hesco: it has one for general asterisk config files, and another one for asterisk voicemail |
| 00:06.25 | dlynes | hesco: /usr/share/vim/vim71/syntax/asterisk.vim and /usr/share/vim/vim71/syntax/asteriskvm.vim |
| 00:06.37 | dlynes | hesco: that's in debian lenny |
| 00:06.44 | dlynes | hesco: assuming you installed vim-full |
| 00:06.47 | coppice | dlynes: can you point to any piece of documentation which says a PAP2 or PAP2T supports T.38? |
| 00:07.01 | dlynes | coppice: not for pap2t, but for pap2...gimme a few |
| 00:07.30 | hesco | I also just found this on google: http://www.voip-info.org/wiki/view/vim+syntax+highlighting |
| 00:07.30 | hesco | thanks, I'll look for those. |
| 00:07.33 | drmessano | The PAP2 most certainly does not.. |
| 00:08.19 | *** join/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 00:08.35 | coppice | T.38 support in the PAP2 and PAP2T should be listed as an urban myth |
| 00:08.40 | drmessano | I've seen mixed documentation on the PAP2T, but what coppice said earlier about the common source, firmware version or two with incomplete options, that probably explains the confusion on that one |
| 00:09.34 | hesco | I have this: /usr/share/vim/vim71, but no syntax directory there, wonder if I'm missing a piece of vim in my install. |
| 00:09.36 | manxpower | Even my SPA-2100 lists T.38 support, but I don't believe it. |
| 00:10.06 | coppice | There seem to be two things fueling this. One is people confusing "FAX" in the documentation with "T.38". The other is people muddling the PAP2 with the SPA2102 |
| 00:10.16 | drmessano | I think the association is that the PAP2T uses the v5 firmware.. the v5 firmware supports T.38.. But the PAP2T does not |
| 00:11.00 | drmessano | They use common documentation and release notes |
| 00:11.11 | coppice | manxpower: the SPA2100 was launched with "supports T.38" in big letters, and "to be added later by software update" in small letters. I don't think they ever got a stable release out |
| 00:12.01 | drmessano | Just like the SPA-941 having all these cool features of the later v5 and v6 firmware, but they never released firmware beyond 5.1.8 or (or was it 5.2.8) for it |
| 00:12.17 | drmessano | But the docs list the SPA-941 along with 8 other phones |
| 00:12.31 | coppice | linksys are terrible for software control. The SPA2102 and SPA3102 use common source, but they are so bad about releasing bug fixes the two boxes have completely separate sets of bugs in the most recently available firmwares |
| 00:13.05 | drmessano | Yeah, the SPA-3102 hook flash is complete blown |
| 00:13.13 | drmessano | Never to be fixed, i would guess |
| 00:13.24 | manxpower | coppice: *nod* That's one of my biggest problems with Linksys. That and half their products don't actually work under load. |
| 00:13.27 | coppice | and these days its pretty hard to even get at the firmware |
| 00:13.59 | manxpower | 1) fire up utorrent 2) router reboot 3) rinse. repeat |
| 00:14.19 | manxpower | Thank dog my actual real cisco (1721) will be here tomorrow |
| 00:14.20 | coppice | manxpower: I suspect that has been a key problem with their T.38 support. I think there isn't enough CPU time for the second port if the first one runs T.38 |
| 00:14.45 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 00:14.58 | KavanS | thank dog! |
| 00:15.31 | manxpower | My SPA-2100 also had the similar problem, but instead of crashing at 45 TCP streams, it didn't reboot until about 150 TCP streams |
| 00:15.34 | drmessano | Linksys routers are fine, just need current or alternate firmware.. Older firmware bombs on BT big time |
| 00:15.46 | drmessano | I had to toss a Linksys Cable modem due to bittorrent problems |
| 00:15.49 | hesco | dlynes: thanks, found them! |
| 00:15.55 | manxpower | drmessano: I have the latest firmware available for the model I have. |
| 00:16.07 | drmessano | Only because comcast wouldn't push the firmware out |
| 00:16.12 | drmessano | manxpower: Which one is it? |
| 00:16.41 | coppice | the cisco web site seems designed to never accept a registration from outside the US :-) |
| 00:16.59 | manxpower | BEFSR41 |
| 00:17.07 | drmessano | Oh god |
| 00:17.34 | drmessano | Which version? |
| 00:17.52 | manxpower | drmessano: the one where the most recent firmware is like 5 years old |
| 00:18.07 | manxpower | one rev before the latest hardware rev, I think. |
| 00:18.27 | manxpower | In any case I'll give it to someone I hate |
| 00:18.49 | drmessano | That is pretty much why.. I dont think you can make a proper assessment based on a router with an 8 yr old hardware design and 5 yr old firmware |
| 00:19.10 | manxpower | drmessano: why not? What has changed in IP protocols in that time? |
| 00:20.06 | *** join/#asterisk Faithful (n=Faithful@124.217.119.183) |
| 00:20.19 | manxpower | I don't expect it to do IPSex or IPv6, I just expect it to not reboot when I start pushing traffic thru it. |
| 00:20.27 | drmessano | Ability to handle the number of concurrent connections associated with Bittorrent is a problem that was shared by many older routers |
| 00:20.28 | manxpower | IPSex! That and IPSec |
| 00:21.15 | manxpower | as I said, my real cisco should be here tomorrw |
| 00:21.54 | drmessano | So 5 yr old firmware was probably untouched by any optimizations made.. Just like my 6 yr old Linksys cable modem |
| 00:21.55 | rue_mohr | if a system had a FXO card with a echo canceler, and a SIP phone with echo cancelation, like a polycom 601, what are the odds the two could conflict and cause audio levels that almost randomly fluctuate between calls, in some cases causing calls with echo |
| 00:21.57 | rue_mohr | has anyone ever used polycom phones with a digium pots card? |
| 00:22.01 | rue_mohr | is "upgrade" the only thing digium support ever says? |
| 00:22.06 | rue_mohr | anyone know how long it takes polycom to answer a tech support question? |
| 00:22.13 | rue_mohr | anyone know how to measure audio levels on a rtp stream? |
| 00:22.42 | manxpower | rue_mohr: the phone EC only applies to speakerphone |
| 00:22.55 | rue_mohr | ? |
| 00:22.58 | rue_mohr | k... |
| 00:23.15 | manxpower | rue_mohr: don't expect any IP phones to have regular EC. The endpath delay is far, far too long. |
| 00:23.31 | dlynes | coppice: I've been trying to find our manual that was kicking around on the fileserver, but it seems to have disappeared |
| 00:23.31 | drmessano | You're still looking for anything that may be causing your echo problems but the obvious that we've beat into you for months |
| 00:23.39 | manxpower | rue_mohr: As far as I can tell, Digium's hardware EC simply does not work well in some situations |
| 00:23.43 | rue_mohr | I'm trying to work out why the audio is so drastically different every day |
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| 00:23.47 | Sier | hi |
| 00:23.56 | coppice | dlynes: this is the normal pattern when I ask that question :-) |
| 00:24.00 | dlynes | coppice: but perhaps you're right...maybe i confused 'fax' with t.38...how else would it be fax support, unless there's t.38? |
| 00:24.35 | manxpower | dlynes: um, detect fax tone, switch to ulaw codec is what "traidional non-t.38" fax support is done |
| 00:24.41 | rue_mohr | drmessano, you guys said to make digium replace the card, and they wont till their tech support has had their fill of the problem, and all they EVER seem to do is push a different software version at me |
| 00:24.41 | coppice | most ATAs have a section for FAX support. It usually sets a fixed jitter buffer length, and forces G.711 as the codec |
| 00:24.56 | dlynes | manxpower: that's not supporting fax...that's just autoswitching over to another codec, that still won't work |
| 00:25.06 | manxpower | dlynes: welcome to marketing. |
| 00:25.32 | drmessano | rue_mohr: and to check the line, which you still claim is fine, based on measurements that are highly questionable |
| 00:25.35 | rue_mohr | maybe I shoudl get a dual T1 card and hook it up to one of the T1 echo cancelers I have, run all the phone audio thru it |
| 00:25.51 | rue_mohr | drmessano, what do you think is wrong with the line that I should look for? |
| 00:25.56 | manxpower | I think it's kind of funny seeing people waste DAYS on trying to get VoiceOverIPOverFax working. I've always used the same solution and never had a problem. |
| 00:25.56 | rue_mohr | excessive echo? |
| 00:26.01 | rue_mohr | bad level? |
| 00:26.02 | coppice | dlynes: its not perfect, but changing the codec and the buffer changes FAX from "cannot work" to "works if the wind is favourable" |
| 00:26.07 | dlynes | manxpower: except for one thing...the one firmware version we've got on this particular hardware version is actually sending out t.38 information in the sip packets |
| 00:26.29 | dlynes | manxpower: I've had it independently verified at the other end by our sip provider, navigata |
| 00:26.33 | manxpower | dlynes: welcome to the world of "most T.38 devices don't work well with any other vendor" |
| 00:27.12 | dlynes | manxpower: but i haven't had them test this one, because asterisk doesn't even believe that it's sending out t.38 on this one |
| 00:27.16 | manxpower | Even my employer uses the same solution to FaxOverVoiceOverIP as I always used when I was a consultant. |
| 00:27.43 | rue_mohr | brb |
| 00:27.46 | dlynes | manxpower: just don't do it, period? |
| 00:27.47 | manxpower | That solution? Install a fsxking POTS line. |
| 00:27.56 | dlynes | manxpower: that's what i meant :) |
| 00:28.00 | manxpower | We have a %100 success rate. |
| 00:28.10 | manxpower | (or as close as you can get with fax machines) |
| 00:28.23 | dlynes | manxpower: really? I've never had 100% success with an analog line and fax :) |
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| 00:29.08 | coppice | its possible to get very close to 100%, but only under controlled conditions |
| 00:32.39 | dlynes | manxpower: but the linksys 2102 works with t.38 for sure? |
| 00:34.05 | manxpower | dlynes: Huh? |
| 00:34.14 | manxpower | I never said that. |
| 00:34.29 | dlynes | manxpower: oh...thought it was you that said that...maybe it was coppice then |
| 00:34.41 | manxpower | I said the config option is on the config web page |
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| 00:43.02 | rue_mohr | I miss anything |
| 00:43.04 | rue_mohr | ? |
| 00:44.26 | dlynes | rue_mohr: no |
| 00:45.30 | dlynes | hesco: makes it a little easier with the syntax files installed, doesn't it? |
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| 01:39.10 | MarcWeber | I'd like to setup a cron job which calls me in the morning so that I wake up. Is asterisk the tool to use? |
| 01:44.57 | [TK]D-Fender | MarcWeber: Certainly can be |
| 01:45.06 | manxpower | MarcWeber: Asterisk is a toolkit that allows you to build a PBX. |
| 01:46.03 | MarcWeber | [TK]D-Fender, manxpower Do you know another project I could use which would make me reach my goal faster? |
| 01:47.57 | MarcWeber | I read that asterisk can connect to sipgate. Then I need a client I pipe a sound file into which will use asterisk to connect to sipgate and I'm done |
| 01:49.01 | [TK]D-Fender | MarcWeber: * can automate that without another client |
| 01:49.31 | [TK]D-Fender | MarcWeber: * IS the client that will call out via sipgate to phone you and play a message |
| 01:51.00 | MarcWeber | [TK]D-Fender: Can you estimate the time it would require you to do such a setup? * documentation is quite long .. |
| 01:51.47 | [TK]D-Fender | MarcWeber: from scratch, about an hour |
| 01:52.22 | MarcWeber | [TK]D-Fender: What's your hourly rate? Can I hire you ? |
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| 02:01.06 | [TK]D-Fender | steps out for a few hours |
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| 03:17.28 | brunner | When I interact with the Manager API using PHP, my script hangs unless do Action: Logoff before trying to read output |
| 03:17.33 | *** join/#asterisk Samodelkin (n=fjoe@host89-251-107-22.hnet.ru) |
| 03:17.45 | brunner | However, I need to read the output of the first command in order to form the second command |
| 03:18.11 | brunner | how can I get the output of "show channels concise" before logging off? |
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| 03:32.22 | AeroCloud | brunner: I am no expert at asterisk, but you can run an exec(); and parse the output from the exec |
| 03:32.44 | AeroCloud | run the asterisk command from command line |
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| 03:53.39 | hesco | I have a dialer which is dialing phone numbers for a call agent. They agent is already on the phone as the dialer goes out and starts looking for someone at home for the agent to speak with. When we find them, we'll bridge the calls. But what do we do with agent, while we're waiting for a connection? Park() them, Wait() with them? How do I get the dialplan to appropriately pause until I have a call for them to take? |
| 03:54.30 | hesco | not go off and Play(goodbye) && Hangup() ?? |
| 04:05.56 | Gokee2 | Hello all, I am working on getting asterisk to talk to sipgate. I have got outgoing calls working however incoming calls don't work. SO was my sip.conf looks like http://pastebin.com/d4a1ec1c8 and my dial plan is http://pastebin.com/d1747bc49 . Any idea what is wrong? Thanks |
| 04:21.10 | Kobaz | do de do |
| 04:21.19 | Kobaz | crashed asterisk again |
| 04:22.17 | MarcWeber | Gokee2: Does sipgate show that you're connected ? |
| 04:22.30 | MarcWeber | I'm trying the same as you. |
| 04:23.50 | Gokee2 | MarcWeber, No, it does not |
| 04:24.36 | MarcWeber | Gokee2: How did you test that calling out works? |
| 04:25.07 | Gokee2 | MarcWeber, I used SFLphone and dialed 10005 |
| 04:25.20 | Gokee2 | I also called my cell phone and that worked too :) |
| 04:25.39 | MarcWeber | I have to compile those apps first.. |
| 04:26.35 | Gokee2 | ended up using the ubuntu packages on debian and its working fine |
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| 04:55.43 | dandate2 | damn kayako.freepbx.org ripped me off, could not homogenize my moh to end transcoding, charged me for 2 hours, and tried to sell me a faster computer to transcode .wav to g729. i solved the whole problem simply and easily by doing this http://pastebin.ca/1565004 |
| 05:00.01 | dandate2 | i don't know if they were incompetant or just trying to sell me the faster computer lol |
| 05:01.17 | dandate2 | told me that since we couldn't homogenize the moh to a natural codec i would need a quad core pbx with 4 disk raid array that would have to be configured by them since i'm out of country |
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| 05:05.27 | dandate2 | <dandate2> i don't know if they were incompetant or just trying to sell me the faster computer lol |
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| 05:50.15 | [TK]D-Fender | Checkout time, later all |
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| 06:50.06 | hesco | I'd like to ask folks advise again on this, please? Any thoughts would be helpful. I have a dialer which is dialing phone numbers for a call agent. They agent is already on the phone as the dialer goes out and starts looking for someone at home for the agent to speak with. When we find them, we'll bridge the calls. But what do we do with agent, while we're waiting for a connection? Park() them, Wait() with them? How do I get the dialplan |
| 06:50.06 | hesco | to appropriately pause until I have a call for them to take? not go off and Play(goodbye) && Hangup() ?? |
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| 07:08.46 | fiddur | hesco: you can check dialstatus on the next priority, and do what you want if dialstatus is busy for example... |
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| 07:10.37 | fiddur | hesco: sorry, I read to quickly... now I understand your question :) ... |
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| 07:12.55 | fiddur | hesco: you could probably let the agent sit in a queue, and let the called person 'answer' the queue... |
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| 07:20.46 | Polysics | hello |
| 07:21.02 | Polysics | i need to store a large (1000+) number of sip accounts somewhere |
| 07:21.09 | Polysics | do you recommend using mysql? |
| 07:21.27 | Polysics | i am on 1.6.1.6, working correctly off flat file configs |
| 07:22.03 | Polysics | i was thinking that i could store only the sip accounts in mysql, while using flatfile extensions to reach them |
| 07:22.20 | Polysics | as the wiki isn't that keen on realtiming extensions |
| 07:23.17 | agel | hi, anyone can help me with that res_odbc problem? the complete asterisk crashes after that message ... [Sep 14 08:02:18] WARNING[7371] func_odbc.c: SQL Exec Direct failed![SELECT `x` FROM `y` WHERE `a`='z'] |
| 07:23.17 | agel | [Sep 14 08:02:18] WARNING[7371] res_odbc.c: SQL Exec Direct failed. Attempting a reconnect... |
| 07:23.17 | agel | [Sep 14 08:02:23] WARNING[7376] func_odbc.c: SQL Alloc Handle failed! |
| 07:23.17 | agel | [Sep 14 08:02:23] NOTICE[7371] res_odbc.c: Connecting mysql_asterisk |
| 07:23.17 | agel | [Sep 14 08:02:23] WARNING[7376] res_odbc.c: SQL Exec Direct failed. Attempting a reconnect... |
| 07:23.19 | agel | [Sep 14 08:02:23] NOTICE[7371] res_odbc.c: res_odbc: Connected to mysql_asterisk [mysql_asterisk] |
| 07:25.41 | fiddur | Polysics: I use realtime extensions, working very well... pgsql, but that wouldn't make much difference from mysql... |
| 07:25.57 | Polysics | fiddur, does that involve much work? |
| 07:26.06 | fiddur | Polysics: Realtime extensions work well at least with direct extensions... pattern matching you can keep in flatfile perhaps... |
| 07:26.13 | Polysics | i need to do it anyway, just looking at what awaits me :-) |
| 07:26.34 | Polysics | i got this page: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 07:26.44 | Polysics | looks easy enough if that is all |
| 07:26.55 | fiddur | Polysics: No, it's as easy as any other realtime... you just insert exten, priority, app and appdata into a table... only difference from file is you can't use priority 'n' :) |
| 07:27.01 | *** join/#asterisk ming_zym (n=ming_zym@124.127.101.0) |
| 07:27.30 | Polysics | i will start by realtiming sip users |
| 07:27.39 | Polysics | then, if i can call one, i can call all :-) |
| 07:29.41 | fiddur | realtime sip can make one confused in the beginning in regards to 'sip show peer 123' not working without adding ' load' etc... and there's a bug regarding unregister, at least in my setup, that requires me to use ignoreregexpire=yes in sip.conf.... |
| 07:30.51 | fiddur | The bug shows up when the registration is renewed but the original timeout is triggered anyway, unregistering it... I haven't placed a bug report though, since trunk segfaulted on me when I investigated if it was fixed, and I haden't time to trace that :P |
| 07:31.21 | *** join/#asterisk af_ (n=getsmart@88-149-240-140.dynamic.ngi.it) |
| 07:34.18 | Polysics | what do you mean with "adding load", please? |
| 07:35.00 | fiddur | Polysics: To see a realtime sip peer (without caching) you need to write 'sip show peer 123 load' |
| 07:35.19 | fiddur | ...at least in 1.6.1.... |
| 07:35.27 | Polysics | oh ok, sounds easy enough :-) |
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| 08:13.45 | hesco | I'm using a Local Channel to initiate a call to a RECIPIENT channel variable, but when I answer the calls to my test recipient, It gives me a ring tone instead of doing an immediate Bridge($waiting_extension). What might that be about? |
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| 08:39.56 | joobie | hey guys.. everytime my moh plays, it seems to start from the start of the track.. if user(a) is on hold already, how can i make any new users that join the moh to pick up at the same point user(a)'s onhold is at? trying to stop a seperate mp3 stream being rendered for each user on hold |
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| 08:48.22 | dandate2 | odesk.com |
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| 09:00.27 | genin | mornin |
| 09:00.47 | genin | anyone familiar with this error |
| 09:00.48 | genin | 500 Server Internal Error |
| 09:04.32 | Polysics | that's not a * error, it's an HTTP error |
| 09:04.41 | Polysics | it is a web server that has some problem |
| 09:06.24 | genin | uhm |
| 09:06.27 | genin | n,o |
| 09:06.34 | genin | SIP/2.0 500 Server Internal Error. |
| 09:06.43 | genin | SIP error |
| 09:07.01 | Gugge | you look in the log files on the device giving the error |
| 09:07.01 | genin | rfc3261 |
| 09:07.07 | Gugge | and it should be there |
| 09:07.25 | genin | i was looking on the actual asterisk server and it seems like it comes from the providers side |
| 09:07.37 | genin | the problem is i only get it with a certain destination |
| 09:07.53 | genin | one audiocode of a client configured the same way as mine |
| 09:08.04 | genin | cant pass one sigle call to a certain prefix |
| 09:08.08 | genin | but then my audiocode can |
| 09:08.16 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
| 09:08.18 | genin | it is driving me insane |
| 09:08.19 | genin | heh |
| 09:08.42 | Gugge | thats easy then, ask the client to read the logs from the audiocode :) |
| 09:09.10 | genin | so you think this error is being created from the audiocode? |
| 09:09.28 | genin | because i have his ini file of his account that was having the issues |
| 09:09.36 | Gugge | if you pasted the sip debug on pastebin i would know |
| 09:09.43 | Gugge | right now i can only guess |
| 09:09.43 | genin | it is odd because i made another account for this client |
| 09:09.46 | genin | cool give me a sec |
| 09:09.56 | genin | i have to reconfig this audiocode |
| 09:10.16 | genin | and ill do some tests and paste the results |
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| 09:20.58 | phix | hey |
| 09:21.28 | phix | :D |
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| 09:37.57 | AL-Hadi | Hi |
| 09:39.22 | AL-Hadi | is panasonic TDA 200 supported in asterik ? |
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| 09:49.23 | lipek | how to check why am i receiving 401 Unauthorized? |
| 09:49.28 | lipek | one asterisk is connecting to another |
| 09:49.40 | lipek | here is sip debug: http://pastebin.com/m38dc8f05 |
| 09:50.03 | lipek | how to debug it deeper? (i have only response code 401 without explanation) |
| 09:51.02 | lipek | on 194.181.xxx.xxx machine when i call sip show registry i have: mrg3.xxxxxxxx.xx:5060 warszawa 120 Request Sent |
| 09:51.13 | AL-Hadi | hello |
| 09:51.36 | AL-Hadi | does asterisk work with Panasonic TDA 200 ? |
| 09:51.38 | AL-Hadi | PABX |
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| 10:14.33 | flohack | Hi! I'm trying to track down a crash with the 1.6.1 branch where chan.exten = "?\00027" which results in a failed cdr DB query and then a crash because of a race condition while trying to reconnect to the DB. |
| 10:15.16 | flohack | Can someone think of a case which results in chan.exten never being set (it is malloced to 1, which explains the weird string). |
| 10:15.26 | flohack | ? |
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| 10:19.04 | *** join/#asterisk wathek (n=wathek@41.224.152.207) |
| 10:19.13 | flohack | Looking at the queue_log the call was connected to an agent and the caller hung up. |
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| 11:09.18 | wathek | is it possible to limit the duration of a single call ? |
| 11:09.19 | wathek | I mean is it possible to make a call duration doesn't go more than 10 minutes ? |
| 11:09.32 | voipmonk | yes |
| 11:09.38 | wathek | cool |
| 11:10.04 | wathek | voipmonk, I'm trying to configure for the first time a VoIP Server using Debian lenny |
| 11:10.10 | voipmonk | search for absolutetimeout or similar |
| 11:10.26 | wathek | ok thank you buddy |
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| 11:16.38 | voipmonk | wathek you'll need to refer to http://www.voip-info.org/wiki/view/Asterisk+func+timeout |
| 11:16.58 | wathek | voipmonk, ok let me have a look at it |
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| 12:01.38 | Druken | has anyone here had problem receiving calls on 1.6 from freeswitch? |
| 12:01.52 | garymc | anyone know how i can terst if UDP port 5060 is open to my asterisk server. As my sip phone is dialing to extensions at the office from home but no voice is heard either end |
| 12:02.51 | Druken | RTP = voice doesn't travel over 5060, it would be in the range of 10,000 - 20,000 |
| 12:02.54 | voipmonk | if the phone rings you're good |
| 12:02.58 | voipmonk | if you hear no audio its rtp |
| 12:03.12 | garymc | right so how do I test this? |
| 12:03.14 | Druken | 5060 is the "control" port |
| 12:03.18 | voipmonk | which means u need to open 10,000 ( or 10,001 if u have webmin set for 10,000 ) - 20,000 UDP |
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| 12:03.33 | garymc | cos my Zoiper softphone registers to the server. From my laptop at home. |
| 12:04.02 | garymc | When i take my laptop to the offcie the phone registers and all voice is heard etc, but im sure thats because we are on the same router |
| 12:04.09 | wathek | I got a problem I've configured my first voip server (here's the sip.conf file :http://pastebin.com/m39ca4499) but when I try to call the linux user using the wathek user I'm getting : [Sep 14 14:59:20] NOTICE[18578]: chan_sip.c:14847 handle_request_invite: Call from 'wathek' to extension 'linux' rejected because extension not found. |
| 12:05.10 | Druken | garymc: did you read what voipmonk told you to do? i'm guessing not |
| 12:05.14 | garymc | ok so it could be the port on my Macintosh Airport isnt opening the 10001-20000 range as the Airport doesnt have a clear way of putting port ranges in |
| 12:05.19 | dandre | hello, |
| 12:05.26 | garymc | Druken : Where? |
| 12:05.37 | dandre | How can I place a call on hold from the manager interface? |
| 12:05.47 | Chainsaw | garymc: You could always do a DMZ forward from the Airport Express to the Asterisk server IP. |
| 12:06.01 | garymc | i dont want to DMZ it |
| 12:06.15 | Chainsaw | garymc: Not permanently, no, but you could do that as a test. |
| 12:06.16 | Druken | then suffer :) |
| 12:06.36 | Chainsaw | garymc: If you still don't want to, then yes, you're on your own there. |
| 12:06.44 | garymc | I phone Apple and they said to put a port range in you put 10001-20000 in the port box |
| 12:06.52 | Druken | goes off an a rant about stupid people putting servers behind a nat |
| 12:07.07 | garymc | Chainsaw I will test it yes |
| 12:07.11 | Chainsaw | garymc: So phone Apple for your Asterisk server. You obviously take their word over ours. |
| 12:07.23 | garymc | What are you on about? |
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| 12:25.21 | wathek | I got a problem I've configured my first voip server (here's the sip.conf file :http://pastebin.com/m39ca4499) but when I try to call the linux user using the wathek user I'm getting : [Sep 14 14:59:20] NOTICE[18578]: chan_sip.c:14847 handle_request_invite: Call from 'wathek' to extension 'linux' rejected because extension not found. |
| 12:26.05 | [TK]D-Fender | wathek: Means what it says. It is looking for an extension to match "linux" in the context it is looking in. |
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| 12:30.02 | Druken | morning [TK]D-Fender |
| 12:30.57 | wathek | [TK]D-Fender, so do I have to add something to the extensions.conf ? |
| 12:31.19 | [TK]D-Fender | wathek: well that is telling you what the call is LOOKING for in your extensions.conf, so yes |
| 12:31.45 | Druken | hands [TK]D-Fender the imfamous cluebat |
| 12:32.00 | [TK]D-Fender | Druken: Thank you good sir... |
| 12:32.29 | wathek | [TK]D-Fender, ok thank you |
| 12:32.55 | Druken | hey [TK]D-Fender, have you heard of problems receiving calls on 1.6 from freeswitch? |
| 12:33.12 | [TK]D-Fender | Druken: No |
| 12:33.22 | Druken | k |
| 12:33.28 | Druken | thanks :) |
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| 12:40.41 | garymc | right is there a way to test if UDP ports are open to my server? |
| 12:41.24 | Chainsaw | garymc: Sure, nmap from the outside. |
| 12:41.54 | garymc | ok, should i ssh to my Asterisk server install nmap on it then run a command? |
| 12:42.09 | Chainsaw | garymc: nmap from the *OUTSIDE* |
| 12:42.25 | kaldemar | or using netcat for example. |
| 12:42.40 | Chainsaw | kaldemar: From the outside, yes. |
| 12:43.29 | garymc | Chainsaw : Im on a windows machine on the outside so not sure how to do this? |
| 12:43.43 | [TK]D-Fender | garymc: INSTALL NMAP |
| 12:43.47 | kaldemar | Chainsaw: yes, from the outside |
| 12:43.48 | Chainsaw | garymc: I'm sure there's a Windows port of nmap these days. |
| 12:43.59 | [TK]D-Fender | Chainsaw: For many years.... |
| 12:44.18 | garymc | right ok ill take a look |
| 12:44.21 | Chainsaw | [TK]D-Fender: *nod* I haven't used Windows in 8 years, I can't be sure of these things. |
| 12:44.22 | kaldemar | next he's asking what nmap output means |
| 12:45.09 | *** join/#asterisk dymaxion (n=dymaxion@host86-148-60-186.range86-148.btcentralplus.com) |
| 12:46.18 | dymaxion | Hi, I'm having difficulty deciding whether or not I should use SIP or IAX trunking. I've read various threads/forums, some pro SIP some pro IAX. I'm more interested in this debate from an Asterisk perspective. Is it better to use IAX for asterisk ? |
| 12:46.19 | garymc | Its great in here: people talk about you as if your not here kaldemar |
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| 12:46.57 | Chainsaw | dymaxion: If you're connecting two Asterisk hosts, it makes sense to use IAX. That's exactly what it's written for. |
| 12:47.31 | dymaxion | we'll be connecting our 2nd asterisk box later (via IAX) but in terms of trunking to ITSP shoudl I use SIP or IAX? |
| 12:47.43 | Chainsaw | ITSP? |
| 12:48.00 | dymaxion | <PROTECTED> |
| 12:48.08 | kaldemar | garymc: that was not my intent. IMO it's easier to put a netcat to listen to a port in UDP mode and then connect to it with another and type something. then if you see something on the listening end, it's open and works. |
| 12:48.16 | Chainsaw | dymaxion: Right. Well it depends on what they run. |
| 12:48.27 | dymaxion | they run both :-) just to complicae the matter ! |
| 12:48.36 | garymc | im not that technicaly minded Kaldemar |
| 12:48.50 | garymc | im downloading NMAP for windows now |
| 12:48.57 | dymaxion | I'm thinkign IAX cos it's open source, and also less ports to open in our firewall |
| 12:49.05 | Chainsaw | dymaxion: If they run Asterisk in their back-end servers... it makes sense to use IAX. |
| 12:49.20 | kaldemar | that's why i said it. with nmap you may need knowledge on port states. |
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| 12:49.35 | garymc | right |
| 12:49.44 | Chainsaw | kaldemar: It's nothing pastebin.ca and a kaldemar frontend to the result can't solve though. |
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| 12:50.34 | dymaxion | Chainsaw, thanks.. i'm asking the company now (voiptalk.co.uk) |
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| 12:51.08 | dymaxion | aha they use Asterisk in the back-end so IAX it is then :-) cheers |
| 12:51.38 | kaldemar | Chainsaw: i tend to be buggy, occasionally :) |
| 12:51.40 | Chainsaw | dymaxion: *nod* You could always say you have a preference for IAX, and ask if they could think of any scenario where SIP trunking would be a better idea. |
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| 12:51.52 | dexthageek | Good Morning |
| 12:51.57 | Chainsaw | Hi there. |
| 12:52.09 | dexthageek | I am running into a really strange problem |
| 12:52.27 | Chainsaw | My crystal ball is in the shop. So you'll have to actually tell me about it now. Sorry about the inconvenience. |
| 12:52.42 | dexthageek | i have multiple asterisk servers one of them over the weekend stopped recording the A leg |
| 12:53.04 | garymc | right anyone wanna help me out with my problem, as Myports are set as open on my router and on my Airport. But I cant get voice to work over the sip connection. It works when i test in the office on the same network conection |
| 12:53.19 | Chainsaw | garymc: What does nmap think? |
| 12:53.36 | garymc | namp is thinking it still downloading |
| 12:53.37 | *** join/#asterisk gabri-shatana (n=shatana@95.235.120.253) |
| 12:53.40 | gabri-shatana | hi |
| 12:53.42 | garymc | nmap |
| 12:53.49 | Chainsaw | garymc: I can't answer you until you show me nmap results. |
| 12:53.53 | voipmonk | dexthageek: rebooted already? or is reboot not an option? :) Restart asterisk? |
| 12:53.53 | garymc | ok |
| 12:53.57 | dexthageek | we have asterisk record each leg seperate and we mix them manually later |
| 12:54.01 | gabri-shatana | i want to configure asterisk on my server with euteliavoip |
| 12:54.17 | dexthageek | I completely stopped asterisk |
| 12:54.46 | gabri-shatana | how can i check if it work propely ? |
| 12:55.02 | *** join/#asterisk oej_ (n=olle@132.177.253.250) |
| 12:55.07 | dexthageek | My sip proxy handles load balancing so I was able to pull this server out of the loop |
| 12:55.44 | gabri-shatana | anyon? |
| 12:57.00 | Chainsaw | dexthageek: It's worth checking dmesg, to see whether the underlying OS ran out of memory or any hardware glitches occured. |
| 12:57.21 | Chainsaw | dexthageek: Always suspicious to me when you have a cluster of several machines and only one malfunctions. If it's software, you'd expect it to hit all your nodes equally. |
| 12:57.27 | dexthageek | Chainsaw: yeah I was monitoring it and everything was happy |
| 12:57.31 | [TK]D-Fender | gabri-shatana: PLACE A CALL |
| 12:57.35 | dexthageek | exactly |
| 12:57.37 | voipmonk | LOL |
| 12:57.53 | gabri-shatana | i have no money i want use it only for in-call |
| 12:58.28 | Chainsaw | dexthageek: Hm, okay. Anything in the asterisk logs themselves at around that timeframe? |
| 12:58.33 | [TK]D-Fender | "How do I know my car is working?" , "START THE FUCKING ENGINE!" |
| 12:58.34 | dexthageek | Chainsaw: at first I thought I had an issue in the IVR but the IVR is identical on all servers |
| 12:59.16 | gabri-shatana | in my server i haven't a sound card |
| 12:59.21 | *** join/#asterisk redax (i=redax@r6.hu) |
| 12:59.23 | gabri-shatana | i want use asterisk for remote control... |
| 12:59.24 | redax | hi, |
| 12:59.30 | Chainsaw | Hello redax. |
| 12:59.43 | dexthageek | Chainsaw: if I was only that lucky. Asterisk says it is recording the call but then when the call finishes uniqueid-in.wav is only 44 bytes |
| 13:00.02 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 13:00.08 | Chainsaw | dexthageek: Okay, so you have a standard WAV header but no PCM data. |
| 13:00.15 | redax | if an asterisk box is behind a firewall and ports 5060-5080, 10000-10999 are nat'd to the asterisk box. what makes oneway audio? |
| 13:00.22 | dexthageek | Chainsaw: yes |
| 13:00.27 | redax | shall I use externalip= as well? |
| 13:00.40 | redax | hi Chainsaw |
| 13:00.41 | Chainsaw | redax: You're looking at an RTP failure due to NAT. |
| 13:00.56 | [TK]D-Fender | gabri-shatana: * doesn't need a sound card |
| 13:00.57 | dexthageek | Chainsaw: the b leg recording is fine |
| 13:01.02 | gabri-shatana | i know |
| 13:01.02 | Chainsaw | redax: Use nmap to doublecheck that this 10000-10999 pot range is open from the outside. |
| 13:01.09 | [TK]D-Fender | redax: READ > |
| 13:01.11 | [TK]D-Fender | ~sipnat |
| 13:01.12 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:01.45 | gabri-shatana | split server? |
| 13:01.46 | redax | hi D-Fender, |
| 13:01.50 | Chainsaw | dexthageek: Is this a SIP trunk or something more elaborate? PRI? BRI? |
| 13:02.21 | *** join/#asterisk flohack (n=fhackenb@lancelot.acoveo.com) [NETSPLIT VICTIM] |
| 13:02.21 | *** join/#asterisk digilink (n=digilink@c-76-123-245-221.hsd1.tn.comcast.net) [NETSPLIT VICTIM] |
| 13:02.21 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) [NETSPLIT VICTIM] |
| 13:02.21 | *** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1242444747.dsl.bell.ca) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk thansen (n=thansen@76.27.110.194) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk box2 (n=box2@DCC.SEND.startkeylogger.000.telephreak.org) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk ilteris (n=ozzz@ip67-155-145-199.z145-155-67.customer.algx.net) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk dwayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 13:02.22 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) [NETSPLIT VICTIM] |
| 13:02.22 | *** join/#asterisk keulin (n=cray@bne75-6-82-229-246-155.fbx.proxad.net) [NETSPLIT VICTIM] |
| 13:02.26 | redax | ok. so I need the externip.. thanks |
| 13:02.50 | garymc | Chainsaw : Its found the 3 TCP ports on the ip address, dont even know if its finding UDP just say initiating UDP scan |
| 13:03.07 | dexthageek | Chainsaw: SIP trunk |
| 13:03.16 | [TK]D-Fender | gabri-shatana: You won't knw that its right until you PLACE A CALL. |
| 13:03.47 | Chainsaw | dexthageek: Out of ideas for the moment, sorry. I'll pass you on to Fender. |
| 13:04.02 | dexthageek | Chainsaw: thanks for trying :) |
| 13:04.09 | redax | hides -- the old ip address was at the externip={ipaddr} |
| 13:04.12 | redax | ;/ |
| 13:04.16 | gabri-shatana | i havent set asterisk for outboud calls |
| 13:05.04 | *** join/#asterisk digilink (n=digilink@c-76-123-245-221.hsd1.tn.comcast.net) |
| 13:05.34 | *** join/#asterisk Greenbooger (n=Greenboo@vc-41-4-47-249.umts.vodacom.co.za) |
| 13:05.38 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 13:06.00 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:06.00 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:07.18 | garymc | Chainsaw : http://pastebin.ca/1565427 this is the Nmap output upto now |
| 13:07.32 | garymc | I dont think its finding the udp ports. |
| 13:07.41 | *** join/#asterisk nain (n=nain@119.154.79.140) |
| 13:07.45 | nain | Hello Everybody |
| 13:07.55 | garymc | But udp 5060 must be open otherwise i wouldnt beable to dial the office extension form home? |
| 13:08.01 | Chainsaw | garymc: Pastebin once done. |
| 13:08.04 | dexthageek | nain: hi |
| 13:08.11 | Chainsaw | garymc: This is just a progress bar, it doesn't say anything. |
| 13:08.38 | garymc | right, i thought it was just trying, then failing then retrying with a longer time limit? |
| 13:08.57 | Chainsaw | It is increasing the time between port probes, yes. |
| 13:09.09 | Chainsaw | Hopefully with success, if not it will tell you. |
| 13:09.11 | nain | Can i ask openser/kamalio related question here ? |
| 13:09.16 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:09.20 | garymc | Ok ill have to wait then |
| 13:09.47 | garymc | its gone up form 11 minutes remaining to 14 minutes remaining |
| 13:10.24 | Chainsaw | garymc: You may have to go with the netcat approach that kaldemar suggested. Please hold *music* |
| 13:10.42 | [TK]D-Fender | nain: Not for support of it |
| 13:10.44 | Chainsaw | kaldemar: Incoming call for you. UDP ports for RTP. Good luck. *hits transfer* |
| 13:10.45 | nain | I want to use OpenSer/Kamalio in front of Asterisk Server for the purpose of load balancing, My question is this does OpenSer/Kamalio re-write sip header/packet before it sends to asterisk server, so asterisk can authenticate sip peer based on it's orignal IP address? |
| 13:11.18 | [TK]D-Fender | nain: Yes it can rewrite the packet |
| 13:11.33 | garymc | Yo Chainsaw : I want you to look at this output once its finished in half hour |
| 13:11.48 | Chainsaw | garymc: Sure. |
| 13:11.58 | nain | <[TK]D-Fender>: is it the default behaviour of openser or we need to configure it specially for this purpose .. |
| 13:12.04 | garymc | cool , its just sitting there now though :( |
| 13:12.30 | [TK]D-Fender | nain: I don't believe there is such a thing as "default". |
| 13:12.42 | [TK]D-Fender | NaiMean you have to get off your ass and CONFIGURE it to do what you want |
| 13:13.17 | nain | [TK]D-Fender: any additioal plugin required for sip header re-write ? |
| 13:13.40 | [TK]D-Fender | nain: #opser <- |
| 13:13.44 | [TK]D-Fender | nain: #openser <- |
| 13:13.51 | kaldemar | Chainsaw: i'm out of here in in 15. use FollowMe |
| 13:13.59 | dexthageek | nain: in your kamialio script you can tell it to do anything to the header you want |
| 13:15.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.22 | dexthageek | nain: I am currently using Kamailio as our sip proxy in front of both asterisk and sems app servers. For load balancing take a look at the kamailio modules dialplan and dispatcher. Check out kamailios website as there is alot of documentation throughout the website and wiki, |
| 13:24.25 | dexthageek | Chainsaw: well at this point I am thinking of bouncing the host server to see if it clears up the A Leg recording issue. Just another one of the strange issues with * |
| 13:24.48 | Chainsaw | dexthageek: *nod* I hope that fixes it, can't think of a real explanation for it. |
| 13:24.57 | voipmonk | its about time |
| 13:24.58 | *** part/#asterisk RoyCrowder (n=roy@net-216-37-64-130.in-addr.worldspice.net) |
| 13:24.58 | voipmonk | :) |
| 13:25.38 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 13:27.28 | dexthageek | Chainsaw: i will let you know - thanks again for trying |
| 13:27.38 | Chainsaw | dexthageek: *nod* |
| 13:29.45 | nain | Thanks every body for your answers |
| 13:30.58 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:34.15 | *** join/#asterisk lowtek (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 13:40.12 | *** join/#asterisk ruben23 (n=RPL@122.55.48.243) |
| 13:40.18 | *** join/#asterisk saisoma (n=irchon@166.137.5.150) |
| 13:42.13 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:43.53 | gabri-shatana | hi |
| 13:44.10 | gabri-shatana | can i control asterisk trough a web-interface? |
| 13:45.33 | gabri-shatana | ?? |
| 13:46.03 | lowtek | gabri: try #freepbx, #asterisknow |
| 13:46.17 | gabri-shatana | ok |
| 13:46.29 | voipmonk | or the one you build |
| 13:46.39 | gabri-shatana | ? |
| 13:46.55 | voipmonk | build an interface |
| 13:46.56 | *** join/#asterisk moy (n=moy@74.12.131.104) |
| 13:46.59 | voipmonk | to do what you want |
| 13:47.21 | voipmonk | maybe yours will be closer to "perfect" |
| 13:49.19 | manxpower | ~answers |
| 13:49.20 | infobot | [answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 13:50.37 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 13:50.39 | *** join/#asterisk Freman (n=twitsrus@ppp178-75.static.internode.on.net) |
| 13:50.39 | b11d` | hello chaps! |
| 13:50.40 | *** join/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek) |
| 13:50.46 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 13:50.53 | Freman | greetings, I'm trying to get a lua dial plan working.... without any success |
| 13:50.54 | *** part/#asterisk beek_ (n=klinebl@pdpc/supporter/professional/beek) |
| 13:51.03 | b11d` | sorry I dont know anything about LUA |
| 13:51.13 | Freman | I've completely removed my extensions.conf and .ael |
| 13:51.51 | [TK]D-Fender | Freman: Good luck with that... |
| 13:51.58 | b11d` | so are you loading the lua module? |
| 13:52.07 | b11d` | greetings TK.. |
| 13:52.19 | Freman | http://pastebin.com/m728a647f yep, and it's even reading the lua file |
| 13:54.08 | *** join/#asterisk casnik (n=Nick@fw1-e0-2.dth.xiocom.net) |
| 13:54.18 | b11d` | does anyone know of a case study written that describes using asterisk in a district-wide setting? I'm trying to get my college district to accept Asterisk for use on six campuses, and the only thing in my way is not being able to show them an example of someone else who pulled this off |
| 13:54.49 | Freman | http://pastebin.com/m6276b87 there isn't a lot in the file, my incomming call says it can't find extension "nodephone" (the register line ends with /nodephone) |
| 13:55.26 | russellb | b11d`: http://www.digium.com/en/company/casestudies/ |
| 13:55.50 | b11d` | thanks russellb... thats exactly what I was hoping to find |
| 13:55.53 | b11d` | you're the best :) |
| 13:55.58 | russellb | you're welcome :-) |
| 13:56.13 | russellb | good luck! |
| 13:56.35 | b11d` | this is a done deal with these articles man.. everything else is OK to go |
| 13:56.35 | b11d` | :) |
| 13:56.54 | russellb | b11d`: If you go forward, our marketing department would love to talk to you to add you to the list :-) |
| 13:56.56 | Freman | Call from 'xxxyyyzzz' to extension 'nodephone' rejected because extension not found. |
| 13:58.10 | b11d` | that'd be just fine.. I'd love to help prove Asterisk can do the job! |
| 13:58.30 | *** join/#asterisk stimpie (n=stimpie@84-104-5-142.cable.quicknet.nl) |
| 13:59.28 | russellb | b11d`: what school do you represent, if you don't mind telling me |
| 13:59.52 | russellb | nevermind, I see it in your hostname, heh |
| 13:59.55 | b11d` | haha yeah |
| 14:00.09 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:00.12 | lowtek | b11d: There was a university that replaced a Cisco solution with asterisk (because they could still use their Cisco phones), google may help ... |
| 14:00.15 | b11d` | im trying to get the six area schools to convert to Asterisk to serve as a model for all of MnSCU |
| 14:00.16 | casnik | you don't need to have a digium card installed on a asterisk system just to start using it right? |
| 14:00.17 | b11d` | www.mnscu.edu |
| 14:00.20 | lowtek | b11d: I specifically remember reading the article |
| 14:00.30 | russellb | that was sam houston state university in Texas |
| 14:00.31 | lowtek | b11d: It was going to save them like $1.5M a year |
| 14:00.45 | b11d` | lowtek.. already dong that a little bit :) |
| 14:00.46 | russellb | casnik: that is correct |
| 14:00.50 | gabri-shatana | is possible to set asterisk as it redirects the call to a phone near my gps pos? |
| 14:00.51 | gabri-shatana | like |
| 14:01.04 | casnik | russellb, thanks |
| 14:01.08 | gabri-shatana | i'm in my house and asterisk redirect the call to house phone |
| 14:01.10 | [TK]D-Fender | gabi* will call whatever you tell it to call |
| 14:01.16 | gabri-shatana | yeah |
| 14:01.17 | [TK]D-Fender | gabri-shatana: * will call whatever you tell it to call |
| 14:01.25 | *** join/#asterisk tbic (n=tbic@24-236-204-27.static.aldl.mi.charter.com) |
| 14:01.44 | gabri-shatana | can it call a phone checking my gps pos? |
| 14:02.20 | lowtek | gabri-shatana: It can if you script or program it to is the answer |
| 14:02.27 | Freman | hmmm, I'm suspecting that incoming calls can't be dumped streigth to lua |
| 14:02.27 | tbic | how can I hangup a SIP channel from cli? |
| 14:02.30 | [TK]D-Fender | gabri-shatana: If you write the script to do so |
| 14:02.33 | lowtek | gabri-shatana: But there is no native functionality for that |
| 14:02.47 | *** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com) |
| 14:02.51 | [TK]D-Fender | tbic: "sfot hangup [channel]" |
| 14:03.15 | [TK]D-Fender | s/sfot/soft/ |
| 14:03.30 | gabri-shatana | when it recive a call it send a request to my phone with gps it reply with the coord |
| 14:03.42 | tbic | thank that was it |
| 14:03.50 | gabri-shatana | and asterisk select the propely number... |
| 14:04.01 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:04.19 | *** part/#asterisk tbic (n=tbic@24-236-204-27.static.aldl.mi.charter.com) |
| 14:04.24 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 14:04.37 | [TK]D-Fender | gabri-shatana: * has nothing to do with getting the coordinates. this is up to you with some sort of scripting you'll have to code yourself froms cratch |
| 14:05.04 | gabri-shatana | i know |
| 14:05.17 | gabri-shatana | but now my problem is... |
| 14:05.33 | gabri-shatana | i want to asterisk control my server like |
| 14:05.41 | gabri-shatana | i call i press a "2" and the server reboot |
| 14:05.55 | [TK]D-Fender | gabri-shatana: "core show application system" |
| 14:06.18 | *** join/#asterisk rj45 (n=rj45@c-24-0-166-121.hsd1.pa.comcast.net) |
| 14:06.18 | gabri-shatana | ... |
| 14:06.34 | *** join/#asterisk pbxuser911 (n=pbxuser9@75.99.9.170) |
| 14:06.42 | pbxuser911 | anyone ever set up teleyapper? |
| 14:09.41 | gabri-shatana | [TK]D-Fender, so.. |
| 14:12.26 | p3nguin | pbxuser911: What does it do? |
| 14:12.54 | Freman | sighs well... guess I'll give up on the thought of using lua - it would have been nice, I'll port the ael over from my old box |
| 14:14.40 | Katty | sighs. |
| 14:15.14 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 14:15.28 | dlynes | coppice: So the 2102 is guaranteed to work with t.38 then? |
| 14:15.32 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:15.32 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:15.36 | *** part/#asterisk Freman (n=twitsrus@ppp178-75.static.internode.on.net) |
| 14:16.18 | coppice | its guaranteed to support T.38, but how well is another matter entirely |
| 14:16.27 | dlynes | coppice: ah, ok |
| 14:16.29 | dlynes | coppice: thanks |
| 14:17.16 | dlynes | coppice: is it any better at supporting it than the mediatrix, though? |
| 14:17.34 | Gugge | can i somehow replace +45 in a variabel with nothing, or do i have to test if ${var:0:3} is +45 and then use ${var:4} ? |
| 14:18.09 | coppice | the mediatrix has some funky bugs, but most systems seem to tolerate it (spandsp has some code to work around them) |
| 14:18.56 | *** join/#asterisk lowtek (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 14:19.13 | dlynes | coppice: ah...so the t.38 on the mediatrix will work with spandsp, but not necessarily asterisk's passthrough? |
| 14:19.20 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 14:19.50 | *** part/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 14:22.24 | pbxuser911 | anyone evr used TeleYapper 4.0 for Asterisk 1.4 for wakeup call service? |
| 14:22.45 | jaytee | I know that Asterisk 1.4 can only do T38 passthrough but is there a way in 1.6 to take a call from the PSTN and then relay it to a T38 endpoint and preserve the number that was dialed? I want to use a block of DID's to route faxes to user's Outlook inboxes through UM. I can get it to work from fax on ATA to UM internally but not from PSTN through Asterisk to my Exchange UM system. |
| 14:24.31 | *** join/#asterisk spck (n=spck@216.170.229.86) |
| 14:26.23 | *** part/#asterisk manxpower (n=EWieling@69.73.94.162) |
| 14:26.53 | coppice | won't fax to e-mail throw the faxes at their mailboxes? |
| 14:28.54 | jaytee | coppice, not sure what you mean |
| 14:31.13 | *** join/#asterisk dajhorn (n=dajhorn@transmisor.vanadac.com) |
| 14:31.53 | *** join/#asterisk Daviey (n=Daviey@ubuntu/member/pdpc.gold.Daviey) |
| 14:32.42 | *** join/#asterisk beniwtv (n=beniwtv@124.Red-83-36-62.staticIP.rima-tde.net) |
| 14:35.50 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:36.12 | beniwtv | Hi all... Im trying to debug a strange problem... Sometimes when I do a call on my * box, music on hold suddenly starts in the middle of the call, lasts for a few seconds (2-4), then stops and all goes normally. I have checked features.conf, but there's everything on default. I also found a forum thread, here: http://www.trixbox.org/forums/trixbox-forums/help/music-hold-suddenly-jumps-out-and-interrup-conversation. Any ideas on |
| 14:36.12 | beniwtv | <PROTECTED> |
| 14:36.59 | tzafrir_laptop | beniwtv, what type of call is it? SIP? DAHDI? |
| 14:37.10 | *** join/#asterisk raden_work (n=tanning@69-179-99-17.stat.centurytel.net) |
| 14:38.18 | [TK]D-Fender | beniwtv: show us the call <- |
| 14:38.39 | *** join/#asterisk Wimme (n=Wim@83.101.79.230) |
| 14:39.57 | beniwtv | tzafrir_laptop, [TK]D-Fender: It's a normal SIP call, will post messages in a second |
| 14:40.05 | tzafrir_laptop | wow. We finally have the gnu OS: http://kongoni.co.za/ |
| 14:40.23 | Chainsaw | With a HERD kernel? |
| 14:40.39 | tzafrir_laptop | HURD, and no, it is a sort of gnu/linux |
| 14:40.46 | Wimme | anyone got skype f or asterisk to work? when i do "skype show users" its says Connection Error" and skype login returns "active skype user 'username' not found. |
| 14:40.59 | Wimme | i tried logging in with the user in a regular skype client and that works |
| 14:41.03 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 14:41.20 | Chainsaw | tzafrir_laptop: Site fails to load here. |
| 14:41.50 | Chainsaw | tzafrir_laptop: It's not a real GNU OS to me until they run on their own HURD kernel. |
| 14:42.00 | tzafrir_laptop | Fails to reload here as well |
| 14:42.05 | Chainsaw | It would put an end to this GNU/Linux silliness as well. |
| 14:42.16 | mog | Chainsaw, or until gnu says that x kernel is the gnu kernel |
| 14:42.35 | tzafrir_laptop | "Kongoni is the Shona word for a Gnu " |
| 14:42.37 | mog | just like how X11 is part of gnu os |
| 14:42.37 | Chainsaw | mog: They won't, because they can't claim copyright on it. |
| 14:42.51 | mog | they dont have copyright of all thats in gnu os to begin with |
| 14:43.22 | mog | if fsf or linus had been more agreable the linux kernel would have just been the gnu kernel |
| 14:43.32 | mog | but stallman and linus are both blowhards |
| 14:43.43 | gabri-shatana | what's the default database's user and pwd ? |
| 14:43.49 | Chainsaw | mog: And I like it that way :) |
| 14:43.54 | mog | heh |
| 14:44.12 | tzafrir_laptop | gabri-shatana, what default database? the astdb? |
| 14:44.19 | gabri-shatana | yes |
| 14:44.36 | tzafrir_laptop | It's an old version of Berkeley DB |
| 14:44.55 | [TK]D-Fender | gabri-shatana: There is no "default database", and there is not password |
| 14:45.35 | gabri-shatana | freepbx want a pwd |
| 14:46.39 | gabri-shatana | Connecting to database..FAILED |
| 14:46.39 | gabri-shatana | Try running ./install_amp --username=user --password=pass (using your own user and pass) |
| 14:46.39 | gabri-shatana | [FATAL] Cannot connect to database |
| 14:47.29 | *** join/#asterisk Polysics (n=luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 14:47.32 | Polysics | hello |
| 14:47.55 | Polysics | how do i get past this error: [Sep 14 16:38:26] WARNING[17538]: config.c:2010 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available? |
| 14:48.11 | Polysics | i have installed asterisk_addons and modified the config files as per wiki |
| 14:48.20 | Polysics | i suppose i am missing something in modules.conf |
| 14:48.20 | gabri-shatana | [TK]D-Fender, ???? |
| 14:48.32 | Polysics | i am trying to get the SIP users in a table |
| 14:48.55 | kaldemar | gabri-shatana: that's not asterisk, that's freepbx |
| 14:49.09 | gabri-shatana | yeah but it want |
| 14:49.23 | Polysics | which module names do i need to load? |
| 14:49.40 | kaldemar | Polysics: res_config_mysql.so is the mysql engine |
| 14:50.38 | Polysics | so i add "load => res_config_mysql.so" at the end of the file? |
| 14:51.09 | [TK]D-Fender | gabri-shatana: FreePBX is NOT supported here <- |
| 14:51.09 | [TK]D-Fender | ~freepbx |
| 14:51.10 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 14:51.10 | Polysics | still getting engine not available after restart |
| 14:51.14 | gabri-shatana | ok |
| 14:52.04 | kaldemar | Polysics: that won't necessarily help. what do you get in CLI with "module load res_config_mysql.so"? |
| 14:52.08 | beek | morning [TK]D-Fender , jaytee |
| 14:52.16 | [TK]D-Fender | beek: Mornin' |
| 14:52.21 | jaytee | morning |
| 14:52.54 | Polysics | kaldemar, no such file or directory |
| 14:52.57 | garymc | Chainsaw : You still here. Nmap finished its results stuff |
| 14:53.00 | Polysics | basically, it's not there |
| 14:53.44 | kaldemar | Polysics: then you didn't install it right. |
| 14:54.02 | Polysics | i just downloaded asterisk-addons, compiled and installed it |
| 14:54.12 | Chainsaw | garymc: Yes, I'm here. |
| 14:54.13 | garymc | heres is the nmap results. http://pastebin.ca/1565563 cant see anything relating to UDP ports or PORT 5060 anyway |
| 14:55.04 | *** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net) |
| 14:55.14 | Skeeter- | ~moh |
| 14:55.15 | infobot | i guess moh is Music On Hold. Good information about how to set it up in the various possible ways can be found at http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf, or originally from http://www.freeplaymusic.com |
| 14:55.18 | kaldemar | Polysics: you need to get the module in /usr/lib/asterisk/modules |
| 14:56.04 | beniwtv | tzafrir_laptop, [TK]D-Fender: Debug is here: http://pastebin.ca/1565566 |
| 14:56.30 | Skeeter- | anyone got a good site for MOH, asterisk ready |
| 14:56.58 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 14:57.01 | Chainsaw | Skeeter-: I use Blue Valley by Karsten Koch. Royalty free and it sounds nice. |
| 14:57.33 | Chainsaw | Skeeter-: Not to mention it doesn't do that white-noise-of-doom thing on a cellphone. |
| 14:57.40 | Polysics | kaldemar, i probably sound noobish, but how do i do that? |
| 14:57.52 | [TK]D-Fender | beniwtv: "... call goes on normally ..." <-- you are CUTTING OFF THE DAMN EVIDENCE |
| 14:58.20 | Chainsaw | garymc: You'll have to use netcat instead, sorry. |
| 14:58.26 | garymc | Right |
| 14:58.36 | [TK]D-Fender | Skeeter-: www.asterisk.org |
| 14:58.38 | garymc | netcat do i use that on this windows machine? |
| 14:58.39 | kaldemar | garymc: you only scanned 1000 ports, and the output doesn't say which ones were scanned. you need to scan the ports that you expect to be open. |
| 14:58.58 | Chainsaw | connects kaldemar to garymc |
| 14:59.03 | garymc | Kaldemar I cant find the option to do that on nmap |
| 14:59.07 | kaldemar | Polysics: for example by copying the file there once you locate it. :) |
| 14:59.07 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:59.10 | [TK]D-Fender | Chainsaw: This is a family channel! |
| 14:59.17 | kaldemar | Chainsaw: ... |
| 14:59.45 | Skeeter- | D-Fender, Chainsaw: Thanks |
| 14:59.47 | kaldemar | garymc: keep on looking |
| 15:00.04 | garymc | unless i need to put it in the command prompt bit. eg "nmap -sP ....etc etc |
| 15:00.21 | kaldemar | garymc: hint: -p |
| 15:01.02 | Polysics | shouldn't it be somewhere in the asterisk-addons source after make? |
| 15:01.30 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 15:01.53 | raden_work | i have a phone with no features can i set asterisk up so * or # will do a transfer or park the call or anything ? |
| 15:02.00 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 15:02.16 | beniwtv | [TK]D-Fender: There are no log messages between that :-/ |
| 15:02.40 | [TK]D-Fender | beniwtv: Sorry, I'm not buying that... |
| 15:02.41 | kaldemar | raden_work: yes |
| 15:03.33 | Polysics | the res_config_mysql.so module apparently does not get built by make |
| 15:04.01 | Polysics | make[1]: Entering directory `/opt/src/asterisk-addons-1.6.1.1/res' - make[1]: Nothing to be done for `all'. |
| 15:04.16 | kaldemar | Polysics: did you select to build it with make menuselect? |
| 15:04.25 | Polysics | erm :-P |
| 15:04.39 | beniwtv | [TK]D-Fender: huh? Well, if you want to see 500+ user logins rejected with bad password, be my guest. They are of type "[Sep 14 16:57:15] NOTICE[4891]: chan_sip.c:15055 handle_request_register: Registration from '<sip:myuser@mydomain.com>' failed for 'xxx.xxx.xxx.xxx' - No matching peer found" |
| 15:05.20 | kaldemar | Polysics: back to installing phase. |
| 15:05.38 | *** join/#asterisk KrisWillis (n=kris@host86-146-227-158.range86-146.btcentralplus.com) |
| 15:05.50 | *** join/#asterisk drichard (n=drichard@gw-123.euroconnect.fr) |
| 15:05.57 | *** join/#asterisk sercik (n=ciccio@host218-96-dynamic.53-79-r.retail.telecomitalia.it) |
| 15:06.02 | sercik | good day |
| 15:06.05 | drichard | Hi all |
| 15:06.05 | garymc | kaldemar iam inputting this into the command line but it doesnt seem to scan the udp port "nmap -p u:5060 81.***.***.** |
| 15:06.14 | [TK]D-Fender | beniwtv: We don't see the end of the call, and you shouldn't be having 100's of error messages flying around like that either |
| 15:06.29 | sercik | hi garymc |
| 15:06.30 | drichard | I have an issue in the conference room configuration |
| 15:06.38 | garymc | hi sercik |
| 15:06.39 | drichard | is it possible for somebody to help me ? |
| 15:06.57 | sercik | this is the best way to not obtain help... |
| 15:07.25 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 15:07.29 | drichard | really so how can I obtain help ? ;-) |
| 15:08.02 | sercik | i have a great doubt: which is the context of calls incoming from analog line? |
| 15:08.22 | beniwtv | [TK]D-Fender: I agree. But we have 30.000 customers, and many of them try to log-on even if their account is on hold. (Specially the bad guys trying to get free calls). Don't know how to stop that other than disabling the debug messages alltogether... |
| 15:08.53 | [TK]D-Fender | beniwtv: Sorry, we've got nothing to go on... |
| 15:08.55 | *** join/#asterisk ethicx (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 15:08.56 | kaldemar | garymc: nmap -sU -p U:5060 <ipaddr> <-- if the documentation has U, you don't use u. |
| 15:09.31 | garymc | yeah sorry tried both ways |
| 15:09.41 | beniwtv | [TK]D-Fender: I pretty much figured so, thanks anyway. |
| 15:09.45 | ethicx | hello everyone. |
| 15:10.01 | drichard | The error message is : app_meetme.c: Unable to open pseudo device |
| 15:10.02 | garymc | ok says its open sip |
| 15:10.10 | garymc | using your code there Kaldemar |
| 15:10.14 | Polysics | kaldemar, i suppose i solved it thanks to you :-) |
| 15:10.16 | garymc | 5060 is open/sip |
| 15:10.27 | kaldemar | Polysics: no problemo |
| 15:10.38 | drichard | I have verified and dahdi_dymmy module is loaded |
| 15:10.42 | Polysics | how can i tell if it is working? |
| 15:10.47 | garymc | so thats why i can ring the other extensions, but no speech got something to do with 10001-20000 |
| 15:10.59 | *** join/#asterisk neurosys (n=vinix@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 15:11.01 | garymc | so should i just test any port between those two? |
| 15:11.20 | kaldemar | garymc: test all ports your asterisk uses. |
| 15:11.34 | garymc | well i cant test 10000 ports can i? |
| 15:11.40 | kaldemar | garymc: why not? |
| 15:11.48 | garymc | its the RTP ones that arnt working |
| 15:11.56 | ethicx | Has anyone ever experienced a problem with "SIP response 440 "Cannot Authenticate Device" with any of your service providers? |
| 15:11.58 | garymc | the range 10001 to 20000 |
| 15:12.04 | kaldemar | garymc: what is your point? |
| 15:12.07 | sercik | ethicx i had |
| 15:12.16 | Katty | hummm |
| 15:12.22 | sercik | do you input port 5600? |
| 15:12.28 | Katty | Gingered Cheery Pear Cobbler or Pumpkin Pie? |
| 15:12.30 | Polysics | ok, now |
| 15:12.35 | garymc | what would i put in the command line to test all ports in that range? |
| 15:12.37 | kaldemar | garymc: RTP runs on top of UDP. |
| 15:12.39 | Polysics | chan_sip.c:22330 build_peer: Qualify is incompatible with dynamic uncached realtime. Please either turn rtcachefriends on or turn qualify off on peer '1000' |
| 15:12.42 | [TK]D-Fender | Katty: YES |
| 15:12.46 | ethicx | sercik: Don't know what you mean by input port 5600 |
| 15:12.54 | kaldemar | garymc: nmap -sU -p U:5060,10001-20000 |
| 15:13.04 | garymc | ahh ok ill give it a whirl |
| 15:13.07 | casnik | ethicx, I think he means 5060? |
| 15:13.10 | Polysics | afaik, my setup wasn't working until i added qualify |
| 15:13.12 | sercik | in the sip client configuration you should input not only ip address of asterisk server but also port |
| 15:13.18 | Katty | Pot Roast with Mushroom Gravy or Traditional Beef Stew? |
| 15:13.37 | ethicx | sercik: yeah port 5060 is there. |
| 15:13.51 | sercik | casnik i use 5600 is wrong? |
| 15:14.10 | sercik | my sip softphone works altogether |
| 15:14.18 | Polysics | does rtcachefriends go in the general sip.conf, or do i need it on each peer? |
| 15:14.30 | casnik | sercik, I beleive 5060 is the correct port for sip by default ,... I might be wrong though .. |
| 15:14.54 | garymc | sercik : 5060 is one of the ports ;) |
| 15:15.00 | *** join/#asterisk momelod (n=smelo@CPE001f3a8fe859-CM0012c91df0bc.cpe.net.cable.rogers.com) |
| 15:15.06 | momelod | greetings channel |
| 15:15.07 | sercik | i'm not sure, only my experience with xlite and sjphone |
| 15:15.14 | ethicx | sercik: so adding port 5600 in your sip.conf solved the issue with SIP 440 response? |
| 15:15.14 | sercik | together works with 5600 :) |
| 15:15.15 | garymc | and i think 10001-20000 are the others it needs |
| 15:15.35 | sercik | ethicx i was spekaing about client configuration |
| 15:15.39 | sercik | not server side |
| 15:15.50 | casnik | depends on what the sip server is set to listen on (they have to match) |
| 15:16.04 | Polysics | ok, solved that too |
| 15:16.07 | Polysics | one more hurdle |
| 15:16.07 | sercik | •casnik• extacly |
| 15:16.11 | casnik | whther it's asterisk or opensips or whatever |
| 15:16.17 | ethicx | sercik: I'm running ip phones not softphones, so I guess you mean extensions by saying client? |
| 15:16.17 | Polysics | handle_request_register: Registration from '<sip:1000@192.168.1.7>' failed for '192.168.1.60' - Wrong password |
| 15:16.25 | Polysics | the password is not wrong :-( |
| 15:16.29 | casnik | if it's a "default" then 5060 is the common |
| 15:16.29 | drichard | I am using asterisk 1.4.26.2 under ansterisknow and I have got an issue configuring a conference room |
| 15:16.47 | drichard | I have the following message app_meetme.c: Unable to open pseudo device |
| 15:16.48 | momelod | i have echo when calling over my pri line which are connected via a Wildcard TE122 w/ hw echo cancel. If i tune down the volume via tx/rx gains the echo goes away at about -15 but then the call is too quite. |
| 15:17.16 | sercik | ethicx so you should check your sip.conf |
| 15:17.16 | drichard | I have already veryfied and dahdi is started and dahdi_dummy module is loaded |
| 15:17.31 | sercik | are you sure that ip phones are correctly connected to asterisk?? |
| 15:17.45 | ethicx | sercik: I did, and port 5060 is what I use for all my phones and they all work fine, but my problem resides with the service provider. |
| 15:17.51 | voipmonk | but who's your dahdi? ( what timing device are you using? ) |
| 15:18.27 | drichard | I dont have any device, I read that it is possible to use dummy module if we dont have one |
| 15:18.32 | drichard | is it wrong ? |
| 15:18.39 | momelod | w/ the hardware echo canceller enabled, why am i still getting echo? |
| 15:18.54 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 15:19.00 | ethicx | sercik: I get that SIP response 440 when I try to make outbound calls with my voip provider. |
| 15:19.01 | sercik | please someone can tell me which it the context ftom calls incoming from analog line? |
| 15:19.21 | sercik | ah |
| 15:19.42 | sercik | i got that error during registration phase of phones to asterisk |
| 15:19.43 | Polysics | kaldemar, ever seen that? i get register rejected even though the password is correct |
| 15:19.52 | sercik | so you have a different problem... |
| 15:19.58 | momelod | sercik: from-trunk? |
| 15:20.13 | p3nguin | polysics: I would venture to say that asterisk is right and you're mistaken. |
| 15:20.20 | sercik | pci card. |
| 15:20.55 | Polysics | p3nguin, i checked about 10 times, unless there is something i don't know, Ekiga has the same password i have in the "secret" column in the table |
| 15:21.02 | p3nguin | sercik: Only you can know what your contexts are. |
| 15:21.06 | drichard | voipmonk: did you see my answer ? |
| 15:21.16 | kaldemar | Polysics: never with matching credentials. :) |
| 15:21.37 | *** join/#asterisk wathek (n=wathek@41.224.156.190) |
| 15:21.44 | Polysics | password is plaintext by default, correct? |
| 15:21.59 | momelod | sercik: in zapata.conf or chan_dahdi.conf there is an entry for the context |
| 15:22.01 | sercik | i don't need to enter some context to extensions.conf to make my asterisk answer a call on analog line? |
| 15:22.25 | p3nguin | polysics: Looks like you're trying to register a user named '1000'. Is that the right user? |
| 15:22.31 | Polysics | yes, it is |
| 15:22.36 | *** part/#asterisk smtx (n=smtx@p50998557.dip0.t-ipconnect.de) |
| 15:22.51 | sercik | 1000?!? are you reading O'really book? |
| 15:23.04 | p3nguin | polysics: I would change the password in the context for that phone as well as in the client/softphone. |
| 15:23.08 | Polysics | the account name goes in the "name" field in the db, i suppose |
| 15:23.43 | Polysics | or would it be the "username" column? |
| 15:23.46 | voipmonk | who summoned me? |
| 15:24.08 | voipmonk | drichard: http://docs.tzafrir.org.il/dahdi-linux/#_dahdi_timing |
| 15:24.14 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 15:24.14 | p3nguin | sercik: Many people start out that way. Nothing wrong with using 1000 as long as you realize that you might want to rethink the extension numbers before going into production. |
| 15:24.46 | sercik | sure p3nguin |
| 15:24.51 | Polysics | i'm just learning at the moment :-) |
| 15:24.53 | sercik | i have not told is bad |
| 15:25.03 | sercik | Polysics i know less than you trust me |
| 15:25.07 | Polysics | and yes, i carried it over from the o'reilly book |
| 15:25.12 | sercik | i'm reading that book also |
| 15:25.13 | kaldemar | Polysics: name |
| 15:25.16 | p3nguin | polysics: In my sip.conf, the username is an arbitrary name (but I like to put the name of the person who will be using that phone). |
| 15:25.17 | momelod | can anyone help me out w/ these echo issues... im sure echo is a topic thats been beaten to death, but i don't know where else to turn |
| 15:25.22 | Polysics | it's not half bad |
| 15:25.29 | Polysics | p3nguin, i am using realtime |
| 15:25.47 | wathek | would any one help me to test my asterisk config please ? any one got ekiga and could try to call 600@wathek.homelinux.org and tell me if the echo test works ? Thank you |
| 15:25.50 | Polysics | and i at least hope asterisk reports the correct error :-) |
| 15:26.06 | Polysics | what is username for? |
| 15:26.11 | sercik | wathek i can try but i have no microphone |
| 15:26.23 | p3nguin | wathek: We would have to have the user and secret before we could even call 600. |
| 15:26.26 | wathek | sercik, no problem you should hear a femal voice |
| 15:26.34 | sercik | are you? |
| 15:26.37 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:26.41 | wathek | p3nguin, there's no user and secret ! |
| 15:26.48 | p3nguin | Then how's that work? |
| 15:27.05 | garymc | Kaldemar its is showing ports 10000-20000 closed |
| 15:27.11 | wathek | p3nguin, just call sip:600@wathek.homelinux.org |
| 15:27.13 | kaldemar | wathek: why are you not calling yourself? |
| 15:27.23 | Polysics | i wonder if the problem is actually the wrong password... |
| 15:27.26 | momelod | wathek: i called it said it answered but i didnt hear anything |
| 15:27.28 | sercik | i obtain person is unavailable |
| 15:27.31 | garymc | but i opened them up in the routers :S same as i did for 5060 |
| 15:27.38 | Polysics | i changed it to "a", no way i can type that wrong :-) |
| 15:27.58 | wathek | momelod, no voice ? |
| 15:28.03 | wathek | kaldemar, what µ? |
| 15:28.17 | momelod | wathek: just dead air |
| 15:28.37 | p3nguin | Without having a system to register to, ekiga won't do anything for me. |
| 15:28.47 | wathek | ok |
| 15:29.07 | wathek | p3nguin, I'm a newbie I'm trying to configure a VoIP server for the first time |
| 15:29.14 | p3nguin | So good luck on providing services with no users and corresponding secrets. |
| 15:29.16 | kaldemar | garymc: that only means that no software is listening to the port |
| 15:29.31 | garymc | like asterisk? |
| 15:29.47 | p3nguin | exactly like asterisk. :) |
| 15:29.55 | garymc | so how do i fix this? |
| 15:30.11 | kaldemar | garymc: filtered would mean that there's a firewall blocking the ports. no reason to assume that a fw is blocking the ports. |
| 15:30.18 | wathek | momelod, thank you for testing |
| 15:30.22 | wathek | sercik, thank you so much |
| 15:30.34 | garymc | 5060 says open/filtered |
| 15:31.20 | garymc | service: sip |
| 15:31.37 | garymc | its only showing port 10056 closed |
| 15:31.42 | garymc | nothing else |
| 15:32.50 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 15:32.50 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:33.29 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 15:33.34 | [TK]D-Fender | garymc: [11:30]<garymc>5060 says open/filtered <- IMPOSSIBLE |
| 15:33.44 | [TK]D-Fender | garymc: there is no such thing as "open" |
| 15:33.52 | [TK]D-Fender | garymc: UDP is **StATELESS** |
| 15:34.04 | garymc | [TK]D-Fender : you want a screenshot? |
| 15:34.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:34.29 | [TK]D-Fender | garymc: Oh, I'm sure its telling you that, and I'm also pretty sure you did the wrong test |
| 15:35.01 | [TK]D-Fender | garymc: Not that I haven't seent his many times before. |
| 15:35.03 | [TK]D-Fender | wait.... |
| 15:35.05 | [TK]D-Fender | I have. |
| 15:35.35 | garymc | i just ran this command in the nmap command line "nmap -sU -p U:5060 81.***.***.*** |
| 15:35.38 | kaldemar | [TK]D-Fender: nothing wrong with that |
| 15:36.15 | garymc | im just after a little help in getting my sip phone to wrok |
| 15:36.18 | garymc | *work |
| 15:36.36 | Polysics | nothing, still get wrong password |
| 15:36.48 | garymc | I can call the office extensions but when they pick up it is silent |
| 15:37.45 | kaldemar | garymc: i'd blame your nat settings in asterisk, show a debug of a call and someone will surely push you in the right direction |
| 15:38.02 | [TK]D-Fender | garymc: remove that extr POS router |
| 15:38.13 | *** join/#asterisk gardo (n=gardo@121.97.192.10) |
| 15:38.20 | garymc | I cant remove that |
| 15:39.10 | [TK]D-Fender | garymc: Why not? |
| 15:39.38 | [TK]D-Fender | garymc: and i got the same nmap result as you BTW for whatever it counts as |
| 15:39.45 | garymc | if you mean the Apple Router, it is pluggedinto the main bt router from an external office |
| 15:40.31 | [TK]D-Fender | garymc: I mean strip BOTH of thoses pieces of crap and put another router at the first level |
| 15:40.36 | garymc | right, but what happens when I spend more money on other router and i get the same result? |
| 15:41.13 | garymc | Ive had every other port open ever needed, also used this setup for controlling 50 security cameras |
| 15:41.38 | garymc | without a hitch |
| 15:41.51 | [TK]D-Fender | garymc: And every other story like that doesn't offer any help in debugging your actual issue |
| 15:42.02 | neurosys | Can anyone think of why i might have odd hums and line noise on the EVEN numbered channels from a 2 sangoma a400's? |
| 15:42.12 | garymc | no, but its not practical to go get another router right now |
| 15:42.13 | neurosys | and yes. ive emailed sangoma :) |
| 15:42.13 | [TK]D-Fender | garymc: then slap another NIC in your * box and run it direct |
| 15:42.36 | *** join/#asterisk hfb (n=hfb@pool-98-112-210-252.lsanca.dsl-w.verizon.net) |
| 15:42.39 | garymc | you mean missout the Apple router? |
| 15:42.44 | *** join/#asterisk moa_ (n=moa_@65-19-228-168.vnet-inc.com) |
| 15:43.06 | garymc | I would need to feed a cable about 100 yards |
| 15:43.20 | garymc | under ground, ill phone up the office and see if this can be done |
| 15:43.48 | [TK]D-Fender | garymc: you seem to have serious comprehension issues. Let try this a tad more direct : remove both of the fucking routers and run * DIRECT of the connection right from a raw modem interface. |
| 15:44.15 | [TK]D-Fender | garymc: with a PUBLIC IP direct. unfiltered, no extra routing, NOTHING |
| 15:44.33 | Polysics | uhm |
| 15:44.34 | voipmonk | whoa .... the f-bomb |
| 15:44.38 | Polysics | might it be a problem with NAT |
| 15:44.42 | voipmonk | what did you do, garymc ? |
| 15:44.57 | Polysics | the user i am using has nat=yes, but i am not actually connecting via NAT now |
| 15:45.03 | [TK]D-Fender | voipmonk: Oh, don't freak out about a single use. be worrying when I use multiple in all-caps :) |
| 15:45.10 | voipmonk | heheh |
| 15:45.29 | Polysics | i hope asterisk doesn't report "wrong password" when it is not that :-( |
| 15:45.31 | [TK]D-Fender | Polysics: If you want to debug, then show some... |
| 15:45.46 | voipmonk | quote of the day |
| 15:46.42 | garymc | [TK]D-Fender : THey are connecting a cable direct from the BT modem straight to the asterisk box. So the asterisk Box will bypass the Apple Airport router. Is that more like it? |
| 15:46.47 | iCEBrkr | Who the hell is Onebox.com and why do they think I owe them money? |
| 15:46.55 | iCEBrkr | I've never used them for anything |
| 15:47.04 | denon | iCEBrkr: j2 office service |
| 15:47.22 | denon | j2/jfax |
| 15:47.22 | iCEBrkr | Eh? |
| 15:47.29 | *** join/#asterisk clintc (n=clintc@n128-227-126-214.xlate.ufl.edu) |
| 15:47.29 | iCEBrkr | never used'm |
| 15:47.48 | iCEBrkr | I just keep getting billing failure emails. |
| 15:47.53 | coppice | j2 is evil |
| 15:47.53 | iCEBrkr | Onebox Billing Failure for NONE |
| 15:47.57 | iCEBrkr | Looks like it's working!! |
| 15:47.59 | iCEBrkr | NONE |
| 15:48.03 | iCEBrkr | sighs |
| 15:48.21 | iCEBrkr | I'm scared to email them back. |
| 15:48.29 | [TK]D-Fender | garymc: what IP is * getting? |
| 15:48.36 | Polysics | [TK]D-Fender, all i have is [Sep 14 17:44:16] NOTICE[22335]: chan_sip.c:19958 handle_request_register: Registration from '<sip:1000@192.168.1.7>' failed for '192.168.1.60' - Wrong password |
| 15:48.53 | Polysics | what would you need more to see? the config files? |
| 15:48.58 | [TK]D-Fender | Polysics: PASteBIn your sip config and the failed reg attempt with SIP DEBUG enabled |
| 15:49.00 | coppice | j2 regular try to extract money from asterisk users, using for doing fax to email |
| 15:49.07 | Polysics | sure |
| 15:49.22 | coppice | s/using/usually |
| 15:49.27 | garymc | [TK]D-Fender : Do you mean the internal ip or the staic ip? |
| 15:49.36 | garymc | *static ip |
| 15:49.37 | p3nguin | I ran the suggested "nmap -sU -p U:5060,10001-20000 <server IP>" from a remote computer, and nmap has been sitting there for like an hour without ever displaying anything. What does that indicate? |
| 15:49.42 | [TK]D-Fender | garymc: what IP is * getting? <------------ |
| 15:49.54 | Polysics | this is log: http://pastebin.com/m3de06180 |
| 15:50.38 | garymc | [TK]D-Fender : Fukme do you mean the PUBLIC ip addy or the one the router is handing it???????? |
| 15:51.06 | [TK]D-Fender | garymc: it gets an IP. What is it. |
| 15:51.12 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-13-64.w90-55.abo.wanadoo.fr) |
| 15:51.24 | Polysics | and this is the sip.conf: http://pastebin.com/m62965041 |
| 15:51.25 | [TK]D-Fender | Polysics: SIP/2.0 403 Forbidden (Bad auth) <- check your authuser, realm, etc |
| 15:51.43 | *** join/#asterisk wolfehr (n=wolfehr@81.179.240.149) |
| 15:52.09 | garymc | mine gets 2 ips eth0 = 192.168.0.29 |
| 15:52.13 | [TK]D-Fender | Polysics: and try showing everything i asked for |
| 15:52.19 | sercik | Fender: please me and wathek are trying to call each other via sip but we are unable to do that.... |
| 15:52.23 | garymc | eth1= 10.0.1.11 |
| 15:52.31 | sercik | we have added allowguest=yes to sip.conf |
| 15:52.37 | sercik | we need to do something else |
| 15:52.38 | sercik | ? |
| 15:52.51 | Polysics | [TK]D-Fender, sip.conf is here: http://pastebin.com/m62965041 and log is here: http://pastebin.com/m3de06180 |
| 15:52.53 | Polysics | anything else? |
| 15:52.58 | [TK]D-Fender | garymc: Useless. Get another modem so that * gets the WAN IP |
| 15:53.34 | [TK]D-Fender | Polysics: I see no peer info. I also see no imagebin of your ekiga settings. |
| 15:53.45 | sercik | garymc mqybe you can configure that router as bridge |
| 15:53.54 | sercik | and then use pppoe to conecct to internet |
| 15:53.58 | wolfehr | as long as everything is routed correctly it doesn't matter whether you have the external WAN ip or not garymc |
| 15:54.03 | Polysics | ekiga interface is in italian, i will paste it anyway as the field probably are in the same position |
| 15:54.12 | Polysics | how do i get peers info you need? |
| 15:54.17 | Polysics | sip show peers? |
| 15:54.24 | [TK]D-Fender | Polysics: sip.conf <--------- |
| 15:54.41 | Polysics | my sip.conf is all there, i am using database for sip accounts |
| 15:55.29 | *** join/#asterisk stope (n=nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 15:56.09 | Polysics | ekiga settings are here |
| 15:56.10 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:56.10 | Polysics | http://imagebin.org/63792 |
| 15:56.18 | Polysics | anything else that can help? |
| 15:56.20 | sercik | i don't think fender is gay :) |
| 15:56.41 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 15:57.09 | voipmonk | why would it matter? |
| 15:57.14 | Polysics | i am 100% sure the data in the table and in ekiga is the same, especially the password |
| 15:57.16 | voipmonk | damn I got sucked in.. |
| 15:57.41 | voipmonk | auth user? do u have that, Polysics ? |
| 15:57.55 | Polysics | voipmonk, what do you mean, please? |
| 15:58.34 | voipmonk | Nome di accesso |
| 15:58.49 | voipmonk | where in your sip settings do u have that info? |
| 15:59.22 | [TK]D-Fender | where do I see the DB dump? |
| 15:59.35 | Polysics | voipmonk, database table, name column |
| 16:00.04 | voipmonk | authuser |
| 16:00.19 | Polysics | i don't have that column in the db, is it needed? |
| 16:00.40 | voipmonk | Polysics: do you have an authuser field ? |
| 16:00.53 | voipmonk | you are using it in your softphone |
| 16:01.06 | voipmonk | if u dont have it in sip.conf you should remove it from your softphone and try again |
| 16:01.19 | Polysics | then what is name for? |
| 16:01.48 | Polysics | added authuser and put 1000 in it, no difference |
| 16:03.35 | Polysics | btw, sip.conf only has some general settings |
| 16:04.03 | voipmonk | in the olden days you build a sip user like so: http://pastebin.ca/1565636 |
| 16:04.14 | Polysics | db dump here: http://pastebin.com/m73f30f39 |
| 16:04.23 | voipmonk | I apologie if the SpoonFeed 3.0 has been acting up. |
| 16:04.31 | voipmonk | +z there is a z that goes in there somewhere |
| 16:04.41 | neurosys | LOL! |
| 16:04.55 | neurosys | SpoonFeed 3.0 ... |
| 16:05.02 | Polysics | i can make sip.conf accounts work, problem is with db :-) |
| 16:05.43 | Polysics | or i could answer with [isaidthatabout10times] :-) |
| 16:05.53 | Polysics | [TK]D-Fender, got the DB dump? |
| 16:07.28 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 16:07.51 | Polysics | btw, i did not post extconfig.conf or res_mysql.conf, as i think those are solved (* would say something else if it could not reach mysql or similar) |
| 16:08.13 | p3nguin | Are you sure? |
| 16:08.44 | p3nguin | What if the query is wrong, but asterisk can still reach the database? |
| 16:09.11 | Polysics | i solved some problems related to modules earlier, so i supposed that was out of the way |
| 16:09.22 | Polysics | any way i can see queries as they go to the db? |
| 16:09.43 | voipmonk | i see no authuser in your mysql dump |
| 16:11.46 | Polysics | re-added it |
| 16:11.48 | Polysics | http://pastebin.com/m1b2e6e73 |
| 16:12.46 | voipmonk | ok update the field, reload and have at it - |
| 16:13.05 | Polysics | relaod = sip reload? or restart *? |
| 16:13.07 | voipmonk | you dont NEED authuser tho :) but if u want to use it now u have it in there |
| 16:13.41 | voipmonk | sip reload should do the trick |
| 16:13.43 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.97) |
| 16:15.19 | *** join/#asterisk errotan (n=errotan@62.201.122.169) |
| 16:16.02 | Polysics | apparently the "wrong password" message went away |
| 16:16.18 | *** join/#asterisk cesar_CR (n=cesar@201.192.86.30) |
| 16:16.22 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net) |
| 16:16.38 | Polysics | i get this instead: http://pastebin.com/m6fd3534e |
| 16:17.36 | *** join/#asterisk SebastianS (n=schu@adsl-dyn176.78-98-14.t-com.sk) |
| 16:19.29 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 16:20.35 | Polysics | i'm baffled |
| 16:20.51 | Polysics | i tried taking out the authuser column, just to see what happened |
| 16:21.02 | p3nguin | and nothing changed? |
| 16:21.10 | Polysics | and the password message has disappeared, but i still can't authenticate |
| 16:21.21 | Polysics | so it is a different behavior than what i had before |
| 16:21.37 | p3nguin | I am still suspent of the queries to the db. |
| 16:21.41 | Polysics | same setup, different behavior |
| 16:21.42 | p3nguin | suspect, that is. |
| 16:21.51 | Polysics | can i/how do i see them? |
| 16:21.59 | p3nguin | Look in the mysql config. |
| 16:22.29 | Polysics | the * res_mysql config? |
| 16:22.36 | Polysics | or the mysql itself config? |
| 16:22.48 | p3nguin | I think you said it was res_mysql.conf. |
| 16:23.14 | Polysics | it is |
| 16:23.20 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 16:23.27 | Polysics | not much in there, username, password and stuff |
| 16:24.09 | Polysics | all i have in there: http://pastebin.com/m29e5e44f |
| 16:24.14 | p3nguin | I don't know anything about using MySQL with Asterisk, so I assumed the query string was in there. |
| 16:24.42 | voipmonk | where's your sock file Polysics ? |
| 16:24.58 | p3nguin | Maybe it's using TCP. |
| 16:25.20 | Polysics | i have always used tcp with mysql, figured out it would work anyway |
| 16:26.01 | p3nguin | I like to use domain sockets when the app runs on the same computer as the db, but tcp should be just fine. |
| 16:26.04 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 16:26.42 | Polysics | i am using TCP because i am not sure * and mysql will be on the same machine forever |
| 16:26.56 | Polysics | so i am saving myself i trip here in the future :-) |
| 16:27.05 | Polysics | do you see anything suspicious in the above? |
| 16:28.34 | Polysics | keep seeing : [Sep 14 18:28:20] NOTICE[24677]: chan_sip.c:11358 check_auth: Bad authentication received from '<sip:1000@192.168.1.7>' |
| 16:29.05 | Polysics | it would be ironic for the problem to be ekiga configuration, but i can't tell what is wrong |
| 16:29.46 | Qwell | Why would it be ironic? |
| 16:30.04 | Polysics | ok, it would be plain stupid |
| 16:30.23 | Polysics | but there are 4 fields there, i don't think they can be wrong |
| 16:30.32 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 16:30.49 | Qwell | Of course they can be wrong |
| 16:31.48 | Polysics | they can, but i do not reckon they are :-) |
| 16:32.06 | Qwell | I would put money on it. :) |
| 16:33.16 | Polysics | http://imagebin.org/63792 are the settings (interface is in italian, but the fields should be in the same place) |
| 16:34.36 | *** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net) |
| 16:34.48 | Qwell | and your sip.conf? |
| 16:34.59 | p3nguin | Is your registrar at 192.168.1.7? |
| 16:35.16 | Polysics | http://pastebin.com/m62965041 |
| 16:35.26 | Polysics | p3nguin, that is the ip where * is |
| 16:35.33 | Qwell | there's no peer there... |
| 16:35.34 | Polysics | i suppose that is correct |
| 16:35.42 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 16:35.46 | Polysics | Qwell, using db for table |
| 16:35.50 | Polysics | *for peers |
| 16:35.51 | Qwell | show me |
| 16:35.59 | Polysics | what? the db table? |
| 16:36.33 | Polysics | http://pastebin.com/m1b2e6e73 |
| 16:36.52 | Polysics | got it off the wiki, plus the authuser field i added |
| 16:37.09 | SuPrSluG | remove the authuser |
| 16:37.38 | Polysics | ok, did it, reloading sip |
| 16:38.06 | Polysics | no changes in logging in |
| 16:38.26 | SuPrSluG | are you going throug a db or using sip.conf? |
| 16:38.43 | SuPrSluG | *through* |
| 16:38.53 | Polysics | using db |
| 16:39.00 | p3nguin | There's no peer info in sip.conf, so let's hope he's using the db correctly. |
| 16:39.12 | Polysics | dump of my table is above, minus authuser field i just removes |
| 16:39.15 | Polysics | *removed |
| 16:39.40 | SuPrSluG | pastebin extconfig.conf |
| 16:40.37 | ethicx | when I specify insecure=invite for my extension (voip provider) this is so authentication is not required for both incoming and outgoing calls? |
| 16:40.38 | Polysics | http://pastebin.com/m6cef14ca |
| 16:41.58 | Polysics | uncommented boils down to last 2 lines |
| 16:42.26 | Polysics | now i need to go, i'll get back on this tomorrow |
| 16:42.29 | Polysics | thanks for now |
| 16:42.33 | SuPrSluG | shouldn't it be ->sipusers => mysql,asterisk,sip_accounts |
| 16:42.52 | voipmonk | not if his db is called general |
| 16:43.00 | SuPrSluG | or is the database named general |
| 16:43.15 | Qwell | it isn't |
| 16:43.28 | SuPrSluG | that could be a problem |
| 16:43.44 | p3nguin | Oh, but his configuration is FINE! |
| 16:46.10 | *** join/#asterisk dajhorn (n=dajhorn@206.16.96.160) |
| 16:47.04 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 16:49.22 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 16:52.20 | voipmonk | hehe |
| 16:52.30 | voipmonk | what is the name of your database, sir? |
| 16:52.53 | p3nguin | You realize he left 10 minutes ago? |
| 16:53.10 | voipmonk | "asterisk" |
| 16:53.14 | *** join/#asterisk Xetrov` (n=xetrov@unaffiliated/xetrov/x-827361) |
| 16:53.23 | voipmonk | i've recreated his image in my head |
| 16:56.21 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 17:00.18 | SuPrSluG | he'll be back tomorrow asking the same questions. hopefully someone will remember the fix |
| 17:02.30 | *** join/#asterisk gardo (n=gardo@121.97.192.10) |
| 17:03.52 | p3nguin | "ast_rtp_read: RTP Read too short" ? |
| 17:05.05 | *** join/#asterisk thansen (n=thansen@76.27.110.194) |
| 17:05.47 | SuPrSluG | grandstream? |
| 17:06.27 | p3nguin | Using a Cisco phone on a SIP channel. |
| 17:10.49 | p3nguin | Any idea where to start looking for a problem? |
| 17:13.19 | SuPrSluG | check everything is @ full duplex |
| 17:13.39 | p3nguin | I think it pretty much has to be. |
| 17:14.34 | SuPrSluG | i' ve seen when reboot some switch they go to half duplex. not saying this is the issue, but it could cause something like this. |
| 17:15.38 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 17:16.30 | p3nguin | I've never seen that message before. The only thing I changed was allowing udp port range 10000-20000 inbound from the outside world. |
| 17:17.21 | Samodelkin | hi there |
| 17:17.50 | p3nguin | When I didn't allow that port range, calls still worked well, and that message wasn't present. |
| 17:18.15 | Samodelkin | Is there anyone interested in FreeBSD DAHDI port? |
| 17:18.36 | Samodelkin | I submitted a patch to asterisk bug tracker |
| 17:18.57 | *** join/#asterisk CleanerX (n=nix@p5DC0933A.dip0.t-ipconnect.de) |
| 17:20.42 | tzafrir_laptop | Samodelkin, I know that there's an existing DAHDI port |
| 17:34.47 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 17:36.10 | wcselby | presence on 1.4 is different from presence on 1.2 |
| 17:36.30 | wcselby | just felt like sharing |
| 17:38.39 | *** join/#asterisk outtolunc (n=me@66.218.53.172) |
| 17:41.55 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
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| 17:56.37 | komar_666 | hi all |
| 17:58.09 | *** join/#asterisk jolucara (n=jolucara@186.97.0.22) |
| 17:58.18 | *** part/#asterisk jolucara (n=jolucara@186.97.0.22) |
| 18:00.27 | wopsy | hi |
| 18:01.08 | Katty | stretches |
| 18:01.10 | wopsy | someone know if i can show LDAP users ( name , phone etc etc ) in my Asterisk GUI ? |
| 18:01.32 | *** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net) |
| 18:02.53 | [TK]D-Fender | wopsy: #asteriskgui |
| 18:03.00 | [TK]D-Fender | wopsy: GUI's are not supported in this channel |
| 18:04.20 | wopsy | ok sorry |
| 18:04.31 | *** part/#asterisk wopsy (n=80475@AToulouse-754-1-13-64.w90-55.abo.wanadoo.fr) |
| 18:04.31 | *** join/#asterisk wopsy (n=80475@AToulouse-754-1-13-64.w90-55.abo.wanadoo.fr) |
| 18:04.54 | komar_666 | i've some problemes whith NAT, someone to help? |
| 18:05.04 | Naikrovek | just ask |
| 18:05.16 | Naikrovek | no need to ask us if it's okay to ask |
| 18:05.18 | voipmonk | ~sipnat |
| 18:05.19 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:05.42 | ethicx | ~sipnat |
| 18:05.43 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:05.58 | *** join/#asterisk korcan (n=korcan@ip65-44-169-66.z169-44-65.customer.algx.net) |
| 18:05.59 | voipmonk | ~botsnack |
| 18:05.59 | infobot | voipmonk: thanks |
| 18:06.03 | komar_666 | I've already read all of that |
| 18:06.07 | Naikrovek | komar_666: ask |
| 18:06.10 | voipmonk | so whats your problem? |
| 18:06.11 | [TK]D-Fender | ~areyouadog ? |
| 18:06.11 | infobot | Bark! Bark! |
| 18:06.15 | [TK]D-Fender | infobot: good Boy! |
| 18:06.15 | infobot | thanks, [TK]D-Fender |
| 18:06.17 | ethicx | lol |
| 18:06.35 | ethicx | ~whoareyou ? |
| 18:06.41 | ethicx | =( |
| 18:06.51 | Naikrovek | ~whoami |
| 18:06.52 | infobot | you are naikrovek, or n=jjohnson@63-252-251-77.ip.mcleodusa.net on #asterisk and it's 2009.09.14. Don't believe me? Ask ebil! |
| 18:06.52 | casnik | ~troll ? |
| 18:06.53 | infobot | troll is, like, a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or http://www.catb.org/~esr/jargon/html/entry/troll.html |
| 18:07.07 | komar_666 | m'y client sent SIP/SDP whith in header his public ip but ine caracteristic SDP his local adesss, asterisk use it to initilisa RTP, NAT=yes in sip.conf is ok..... |
| 18:07.57 | Naikrovek | komar_666: describe the problem, please. we'll get to the setup after |
| 18:08.34 | Dovid | anyone ever use NGREP to trace a call over IAX2 ? |
| 18:08.35 | voipmonk | be sure to mention the words router , fifty-sixty, and ten thousand and twenty thousand |
| 18:08.45 | Naikrovek | yes |
| 18:08.54 | komar_666 | port are well configured |
| 18:09.19 | voipmonk | where are my doritos |
| 18:09.21 | voipmonk | brb |
| 18:09.33 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:09.40 | komar_666 | client pub----<publicnetwork>-----ASterisk-----client local |
| 18:10.05 | voipmonk | no phones? |
| 18:10.19 | Naikrovek | i think thats what he means when he says client |
| 18:10.23 | Druken | is there a fairly stable release of 1.6? |
| 18:10.25 | komar_666 | client=softphone Sjphone |
| 18:10.34 | Druken | i'm using 1.6.1.6 at the moment, and well, it sucks |
| 18:10.38 | voipmonk | ding! |
| 18:11.15 | komar_666 | Client pub is well registered whith his public ip but he is sending his local ip in SIP/SDP |
| 18:11.30 | voipmonk | yep |
| 18:11.38 | Naikrovek | ah so your asterisk server has a public IP and a private IP |
| 18:11.44 | Naikrovek | and it's sending the private IP out the SIP header |
| 18:11.52 | [TK]D-Fender | Druken: 1.6.0.15 |
| 18:11.55 | Naikrovek | i have no idea why this happens; it works fine on my server |
| 18:12.05 | Naikrovek | Druken: why does 1.6.1.6 suck |
| 18:12.16 | komar_666 | no externip is configured on asterisk thats ok this side |
| 18:12.38 | komar_666 | that's the client that cause me som issues |
| 18:12.47 | [TK]D-Fender | komar_666: PASTEBIN your failed call with SIP DEBUG enabled and include your sip.conf masking ONLY passwords |
| 18:12.49 | [TK]D-Fender | ~pb |
| 18:12.49 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:12.51 | [TK]D-Fender | ^^^^^^^^^^^ |
| 18:13.05 | voipmonk | thats why the client pays you - they are always full of trouble |
| 18:13.05 | Naikrovek | was wondering when you'd chime in :) |
| 18:13.48 | ethicx | ~list |
| 18:13.49 | infobot | one warez list being sent |
| 18:13.58 | komar_666 | language=fr |
| 18:13.58 | komar_666 | defaultexpirey=1800 |
| 18:13.58 | komar_666 | dtmfmode=auto |
| 18:13.58 | komar_666 | relaxdtmf=yes |
| 18:13.58 | komar_666 | externip=77.198.7.125 |
| 18:13.58 | komar_666 | localnet=192.168.1.0/255.255.255.0 |
| 18:14.00 | komar_666 | nat=yes |
| 18:14.01 | Naikrovek | eek |
| 18:14.02 | Naikrovek | no |
| 18:14.02 | komar_666 | localhost=192.168.1.125 |
| 18:14.02 | Naikrovek | non |
| 18:14.03 | Naikrovek | no |
| 18:14.06 | casnik | lol |
| 18:14.07 | casnik | lol |
| 18:14.10 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 18:14.13 | Naikrovek | here he goes |
| 18:14.14 | komar_666 | doesent matter |
| 18:14.15 | *** kick/#asterisk [komar_666!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 18:14.18 | Naikrovek | kaboom |
| 18:14.25 | *** join/#asterisk komar_666 (n=komar_66@125.7.198-77.rev.gaoland.net) |
| 18:14.29 | ethicx | kabbbbbbbboooooom!!! |
| 18:14.30 | komar_666 | soory |
| 18:14.36 | casnik | pastebin man |
| 18:14.39 | Naikrovek | komar_666: no pasting dude, use a pastebin |
| 18:14.39 | Naikrovek | yah |
| 18:14.42 | Naikrovek | ~list |
| 18:14.43 | infobot | one warez list being sent |
| 18:14.47 | Naikrovek | wtf wares? |
| 18:14.50 | Naikrovek | warez |
| 18:14.51 | Naikrovek | ? |
| 18:14.55 | ethicx | lol..same thing I said |
| 18:14.57 | komar_666 | no |
| 18:15.03 | Naikrovek | not you, komar_666 |
| 18:15.26 | komar_666 | where do i paste? |
| 18:15.31 | ethicx | ~pb |
| 18:15.32 | infobot | [~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://bin.cakephp.org/ , or apt-get install pastebinit |
| 18:15.32 | Naikrovek | komar_666: paste your sip.conf (with passwords ONLY masked out) and a log of the call with SIP debugging turned on |
| 18:15.36 | Naikrovek | ^^^^^ |
| 18:15.54 | Naikrovek | komar_666: www.pasetbin.ca seems to be preferred around here |
| 18:16.48 | komar_666 | SIP.CONF |
| 18:16.51 | komar_666 | [freephonie_in] |
| 18:16.51 | komar_666 | type=peer |
| 18:16.51 | komar_666 | context=fromfree |
| 18:16.51 | komar_666 | host=freephonie.net |
| 18:16.51 | komar_666 | [freephonie_out] |
| 18:16.51 | komar_666 | disallow=all |
| 18:16.53 | komar_666 | username=0954****** |
| 18:16.55 | komar_666 | type=peer |
| 18:16.56 | [TK]D-Fender | Kobaz: Where I referred you to with a giant set of links and an underline |
| 18:16.57 | komar_666 | secret=gwe***** |
| 18:16.59 | komar_666 | qualify=yes |
| 18:16.59 | ethicx | ~WTF! |
| 18:17.00 | infobot | wtf is probably what that's fine? |
| 18:17.02 | *** kick/#asterisk [komar_666!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender) |
| 18:17.10 | Naikrovek | dude |
| 18:17.10 | casnik | guy really has no clue |
| 18:17.11 | ethicx | seriously |
| 18:17.13 | *** join/#asterisk komar_666 (n=komar_66@125.7.198-77.rev.gaoland.net) |
| 18:17.24 | Naikrovek | komar_666: |
| 18:17.25 | [TK]D-Fender | komar_666: LAST warning. Do no spam that shit in here |
| 18:17.26 | komar_666 | WTF? |
| 18:17.34 | Naikrovek | go to http://pastebin.ca/ |
| 18:17.39 | Naikrovek | komar_666: go to http://pastebin.ca/ |
| 18:17.41 | komar_666 | ok thsw |
| 18:17.45 | Naikrovek | paste your stuff in there |
| 18:17.56 | Naikrovek | gotta be very direct with some people i guess |
| 18:18.12 | komar_666 | http://pastebin.ca/1565799 |
| 18:18.14 | Naikrovek | some people are "tell me exactly what to do" people i ugess |
| 18:18.33 | *** join/#asterisk lowtek (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 18:18.42 | Naikrovek | komar_666: okay now turn on SIP debugging and place a call that will show the failure |
| 18:19.24 | komar_666 | sip_general_custom |
| 18:19.26 | komar_666 | http://pastebin.ca/1565802 |
| 18:20.39 | *** join/#asterisk giovani (n=giovani@unaffiliated/giovani) |
| 18:20.57 | [TK]D-Fender | komar_666: [freephonie_out] nat =no <------ put it |
| 18:21.00 | komar_666 | sip debug |
| 18:21.04 | komar_666 | http://pastebin.ca/1565803 |
| 18:21.13 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 18:21.14 | [TK]D-Fender | komar_666: [freephonie_in] nat=no <----- ditto |
| 18:21.15 | giovani | I'm looking to create a conference that dials a ringgroup and has people who answer dropped into the conference, or something to the same effect |
| 18:21.36 | komar_666 | ok i cahnge nat for freephoni_out |
| 18:22.05 | *** join/#asterisk timeshell (n=chatzill@gw.lusi.on.ca) |
| 18:22.05 | [TK]D-Fender | komar_666: and that is not SIP DEBUG |
| 18:22.25 | [TK]D-Fender | giovani: Go lookup "call files" and "AMI Originate" on the WIKI |
| 18:22.27 | [TK]D-Fender | ~wikis |
| 18:22.28 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:23.33 | giovani | [TK]D-Fender: cool -- I'll check them out, thanks |
| 18:23.50 | timeshell | [TK]D-Fender So, any thoughts on that buddies disabling thing we talked about last week? |
| 18:24.22 | [TK]D-Fender | timeshell: Nope, and this is an uber-shit week for me to be of a mind to drill this at home for you unfortunately... you really don't want that button there? |
| 18:24.26 | Naikrovek | i love this channel |
| 18:24.42 | Naikrovek | [TK]D-Fender: it's monday |
| 18:24.46 | Naikrovek | already a shitty week? |
| 18:25.08 | timeshell | Really don't want. It's apparently maxing out the IP601's CPU and memory for the number of items in the list and the rest of the IP501's don't have enough memory to list all our users anyway. |
| 18:25.15 | timeshell | It's more of a bother than benefit. |
| 18:26.57 | Naikrovek | are there any irc clients that let you filter the log based on who is speaking (not talking grep) |
| 18:27.15 | Naikrovek | i mean, it's grep like, but i don't wanna log all channels then grep through the logs |
| 18:27.20 | Naikrovek | somethign in-client would be ince |
| 18:27.57 | Naikrovek | timeshell, can you restate |
| 18:28.08 | timeshell | Eh? |
| 18:28.09 | Naikrovek | what button is maxing out your phone cpu & mem |
| 18:28.13 | timeshell | Buddies |
| 18:28.33 | Naikrovek | integration with OCS? |
| 18:29.27 | Naikrovek | afk a sec |
| 18:29.44 | timeshell | IP601 from my googling seems to have problems with CPU processing of buddies list when lots of notifies come into it. That and the IP501's can't list all my users. The IP601 has been rebooting by itself apparently when it receives too many notifies. |
| 18:29.56 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 18:30.16 | komar_666 | SIP debug |
| 18:30.17 | komar_666 | pastebin - Something - post number 1565812 |
| 18:30.17 | komar_666 | Part of Slepp's Projects — Pastebin — TURL — Imagebin — Filebin |
| 18:30.17 | komar_666 | Feedback -- English French German Japanese |
| 18:30.17 | komar_666 | Hide Menu -- Switch Stylesheets |
| 18:30.17 | komar_666 | Create Upload Newest Tools Donate |
| 18:30.19 | komar_666 | Sign In | Create Account |
| 18:30.20 | *** join/#asterisk viq_ (n=viq@unaffiliated/viq) |
| 18:30.21 | komar_666 | Stuff to Do |
| 18:30.23 | komar_666 | New Post |
| 18:30.25 | komar_666 | Upload a Post |
| 18:30.33 | timeshell | At least, that's a suspicion. It also sits at 100% CPU on boot up for about 5 mins afterwards. |
| 18:30.38 | ariel_ | hello |
| 18:30.50 | komar_666 | http://pastebin.ca/1565812 |
| 18:30.53 | timeshell | At any rate, we have no need for the buddies list and I'd rather just disable it;. |
| 18:30.53 | komar_666 | sip debug |
| 18:30.57 | *** join/#asterisk scunizi (n=scunizi@69.199.151.114) |
| 18:31.03 | komar_666 | look at the end AUDIO in |
| 18:32.27 | komar_666 | what do u think about |
| 18:33.07 | komar_666 | whith nat=yes in peer is asterisl use peer ip header or ip in session description? |
| 18:33.47 | komar_666 | or it's just for his external sip/sdp |
| 18:34.08 | Katty | ponders yard decorations for fall |
| 18:34.42 | Deeewayne | Katty, leaves ? |
| 18:35.06 | Katty | was thinking pumpkins. |
| 18:35.06 | [TK]D-Fender | Deeewayne: No, looks like she's staying... |
| 18:35.09 | Katty | maybe some cornstalks. |
| 18:35.16 | komar_666 | fendeR? |
| 18:35.28 | [TK]D-Fender | Kobaz: No idea on your issue. dump your firewall |
| 18:35.34 | [TK]D-Fender | komar_666: : No idea on your issue. dump your firewall |
| 18:35.40 | [TK]D-Fender | komar_666: "iptables --list" |
| 18:36.08 | komar_666 | public client is behind nat from a BOX |
| 18:36.29 | [TK]D-Fender | komar_666: "iptables --list" <--------- |
| 18:36.39 | *** join/#asterisk KingDavidNYC (n=Chris123@static-141-155-99-50.nycmny.east.verizon.net) |
| 18:36.45 | komar_666 | he has registered whith public ip but send local ip in SIP/SDP |
| 18:37.02 | KingDavidNYC | hello everybody!!! |
| 18:37.07 | komar_666 | hi |
| 18:38.24 | komar_666 | how change Peer audio RTP ipadress in public instead of local? |
| 18:38.36 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 18:41.05 | Katty | ponders fall garland up the porch columns. |
| 18:41.48 | beek | offers a pile of leaves to Katty for decoration... |
| 18:41.53 | Katty | what i want is a HUGE pumpkin. |
| 18:41.59 | Katty | like, so big, it could be a chair. |
| 18:42.12 | Druken | hides his gourd |
| 18:42.25 | Katty | :< |
| 18:42.43 | Katty | beek: leaves will be helpful. we have two huge trees in the front yard to help with that. |
| 18:43.15 | Katty | i'd just like a few pumpkins to put in front of the house. |
| 18:43.33 | Katty | sadly, i doubt i could actually lift them :/ |
| 18:43.53 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 18:44.19 | *** join/#asterisk wathek (n=wathek@41.224.156.190) |
| 18:45.24 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 18:46.08 | ayeso | I am getting the following: ERROR[28732]: pbx.c:2857 ast_func_read: Function SIP_HEADDER not registered, but a `core show application sip_headder` shows that its there... what am I missing? |
| 18:46.20 | Druken | just ask one of the guys here, i'm sure they'll put one right next to your scratching post... |
| 18:46.34 | Katty | lol |
| 18:46.48 | Katty | there is no scratching post at my house |
| 18:46.49 | lesouvage | Is SetCallerPres(prohib) supposed to work with 1.4.26.1 dialing out using a SIP account of a SIP provider? |
| 18:47.00 | Katty | but there are a lot of squirrels, bunnies, and birds (= |
| 18:47.05 | *** join/#asterisk okaratas (n=toor@fsf/member/okaratas) |
| 18:47.39 | Druken | where in the planet does kitty live? |
| 18:47.44 | Druken | er, katty, my bad |
| 18:48.21 | imcdona | <PROTECTED> |
| 18:48.36 | imcdona | Any idea how to fix this? |
| 18:49.00 | Katty | Druken: missouri (= |
| 18:49.34 | Druken | oh, cool cool |
| 18:49.45 | Katty | maybe i could find a giant weatherproof basket. |
| 18:49.48 | Katty | and fill it with gourds |
| 18:49.56 | Katty | straw's too messy for the yard. :/ |
| 18:50.50 | lesouvage | imcdona: you should the number of lines to 1, that is the idea of using agents and queues. An gent with a multiline phone is a bad combination. |
| 18:50.52 | Druken | agrees |
| 18:51.24 | Druken | i'm not worried about decorating for that till oct 1 |
| 18:51.35 | Druken | then i'll worry about it |
| 18:51.39 | Katty | that will be here in 2 weeks |
| 18:51.40 | KingDavidNYC | hello guys, is it true that I can configure a polycom phone so that it does not need provisioning? |
| 18:51.52 | scunizi | imcdona: to expand on what lesouvage said, put the agents phone in PBX mode no key system mode. |
| 18:51.59 | scunizi | *not |
| 18:52.08 | Druken | Katty: great, so i got two weeks :) |
| 18:52.34 | Katty | oh!! oh!!! |
| 18:52.37 | Katty | a ghost made out of a pillowcase! |
| 18:52.48 | ayeso | I am getting the following: ERROR[28732]: pbx.c:2857 ast_func_read: Function SIP_HEADDER not registered, but a `core show application sip_headder` shows that its there... what am I missing? |
| 18:52.48 | Katty | that'd make a nice touch............but...that needs to wait until the end of october. *sigh* |
| 18:53.12 | Druken | my ghost is a blow up one, standing behind a tombstone |
| 18:54.07 | casnik | KingDavidNYC, I have never had one that didn't require it |
| 18:54.15 | [TK]D-Fender | KingProvisioning your phone IS configuring it. |
| 18:54.23 | Katty | Druken: do you decorate with any flowers? |
| 18:54.29 | Katty | Druken: i've been thinking about some seasonal mums |
| 18:54.35 | [TK]D-Fender | KingDavidNYC: And You can configure it direct on the phone itself or via its web interface. |
| 18:54.48 | [TK]D-Fender | KingDavidNYC: However people that do that should be dragged out and shot. |
| 18:54.55 | casnik | ^ |
| 18:55.02 | [TK]D-Fender | grumbles "...and survivors should be shot AGAIN" |
| 18:55.28 | casnik | oh Garfield ... |
| 18:55.44 | Druken | Katty: normally yeah, i have gardens, just recently moved, so no gardens here |
| 18:57.02 | Katty | Druken: http://farm4.static.flickr.com/3423/3231243475_dfe4306430_b.jpg |
| 18:57.06 | ayeso | How do I register a function? |
| 18:57.06 | imcdona | how do I put a phone in PBX mode? |
| 18:57.10 | komar_666 | DOes somebody have peer outside public network? |
| 18:57.19 | Katty | Druken: i'd love to pull that off. |
| 18:57.23 | [TK]D-Fender | imcdona: PARDON? |
| 18:57.32 | komar_666 | francais fender? |
| 18:57.34 | KingDavidNYC | What I mean is: method # 1) loading configuration files via tftp server 2) blocking tftp server conf, no firmware files, just configure as if a lynksys phone, just enter ip and username.... I just want to know if I am right or wrong |
| 18:57.48 | [TK]D-Fender | komar_666: ? |
| 18:57.56 | imcdona | scunizi: how do I put a phone in PBX mode? or agent mode? |
| 18:58.10 | komar_666 | pardon mean sorry in french |
| 18:58.21 | [TK]D-Fender | imcdona: neither of those terms are valid |
| 18:58.25 | Druken | Katty: that's not healthy for a house.. it erodes the bricks |
| 18:58.36 | [TK]D-Fender | Kobaz: Je le savais pas... |
| 18:58.43 | [TK]D-Fender | komar_666: rather.. |
| 18:58.43 | imcdona | @[TK]D-Fender he must mean a single line appearance |
| 18:58.45 | SuPrSluG | KingDavidNYC: you can and should auto provision polycom phones through ftp. |
| 18:58.56 | [TK]D-Fender | komar_666: And it is close in meaning in english |
| 18:59.00 | imcdona | thank for all you help |
| 18:59.08 | lowtek | KingDavidNYC: ... because polycoms will upload very useful information via ftp |
| 18:59.13 | komar_666 | ^^ lol |
| 18:59.19 | [TK]D-Fender | imcdona: there is no "mode" for phones like this. |
| 18:59.20 | Katty | Druken: http://farm4.static.flickr.com/3238/3022464964_497f343052_o.jpg |
| 18:59.31 | Katty | Druken: that's the house i'm working with |
| 18:59.32 | SuPrSluG | KingDavidNYC: you set up dhcp options to point to the ftp server and away they go. |
| 18:59.35 | imcdona | I didn't think so. |
| 18:59.45 | scunizi | imcdona: I don't know the specifics for programming the asterisk sys. but in pbx mode you have one button where all calls come in on and you "park" calls if needed. Key system you have programmed buttons to match lines.. one button per line.. if that makes any sense. |
| 18:59.50 | [TK]D-Fender | KingDavidNYC: I prefer fTP personally.. |
| 19:00.37 | imcdona | scunizi: looks like I will have to find a way to remove all the line appearances from Bria. It works with call waiting disabled on polycom phones ok |
| 19:00.58 | [TK]D-Fender | KingDavidNYC: and you have no need to set DHCP options for this. On Polycom's jsut tell it the server IP, user & pass and the rest is a cakewalk |
| 19:02.21 | scunizi | imcdona: the idea behind a que and "one button" setup is so the que will know that a particular endpoint is busy and will move the next incoming call to an open phone or answer it and park them in an orbit until the line is free |
| 19:02.53 | Naikrovek | timeshell: you can disable buddies via the sip.conf if it's an issue |
| 19:03.27 | Naikrovek | timeshell: sip.conf has a config section where you set it up, I believe; disable that section or some part of the config that would render it non-working and your problem is solved. |
| 19:03.32 | timeshell | Naikrovek LOL!!! That's the reason I'm looking for help. IT ISN'T WORKING |
| 19:03.35 | Naikrovek | timeshell: i'll be able to look at it a bit more when i get home |
| 19:03.44 | timeshell | I've already tried to diable it from sip.conf |
| 19:04.18 | timeshell | s/diable/disable |
| 19:04.29 | Naikrovek | timeshell: okay then check individual phone config files, if it's not set up in there, then somehow it was done on the webui. reset the phone so th at local configuration is cleared, and see if that fixes it for that phone. |
| 19:04.44 | timeshell | Naikrovek Already reset the phone. |
| 19:04.47 | timeshell | Same deal. |
| 19:04.50 | Naikrovek | really |
| 19:04.50 | *** join/#asterisk bluOxigen (n=asad@119.73.65.19) |
| 19:05.01 | Naikrovek | did you reset the phone to clear device setting |
| 19:05.06 | timeshell | And I've set the option both in the sip.cfg AND the phone config file. |
| 19:05.12 | timeshell | Ys |
| 19:05.15 | timeshell | 468* |
| 19:05.16 | Naikrovek | wow unreal |
| 19:05.45 | Naikrovek | what version of the SIP firmware are you using |
| 19:05.50 | timeshell | 3.1.3 |
| 19:05.53 | timeshell | C |
| 19:06.12 | Naikrovek | okay cool |
| 19:06.18 | raden_work | Naikrovek, afternoon |
| 19:06.21 | Naikrovek | now why in the dickety would disabling it not disable it |
| 19:06.25 | Naikrovek | howdy do, raden_work |
| 19:06.32 | Naikrovek | i gotta drive home, i'll be back online when i get there |
| 19:06.41 | Naikrovek | in the meantime i'll do some thinkin' |
| 19:06.46 | timeshell | cool |
| 19:06.51 | timeshell | I'm always around. |
| 19:06.57 | timeshell | Sort of... I never log out. |
| 19:06.59 | timeshell | :p |
| 19:07.13 | wcselby | ahhh |
| 19:07.20 | wcselby | all lunched now |
| 19:08.52 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 19:11.04 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 19:16.49 | komar_666 | does somebody have peer in public network? |
| 19:19.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:21.29 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 19:21.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 19:24.47 | *** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net) |
| 19:28.12 | Skeeter- | ~directory access |
| 19:34.47 | *** join/#asterisk andres833 (n=andres83@190.144.75.22) |
| 19:39.18 | *** join/#asterisk digilink-laptop (n=digilink@68-242-110-188.pools.spcsdns.net) |
| 19:40.53 | *** join/#asterisk jolucara (n=jolucara@186.97.14.138) |
| 19:46.06 | digilink-laptop | hi all having an issue with Asterisk. I am running Asterisk 1.6.1.6 and DAHDI compiled from source. I have an OpenVOX TDM400 card with a POTS line connected to an FXO port on port 1. If I hang up before the call gets to voicemail, or hang up while the voicemail greeting is playing, it does not appear to detect the hangup condition. However, if I do leave a voice mail, it will record it and release the line accordingly |
| 19:48.00 | voipmonk | your problem seems to be with a voicemail setting |
| 19:48.30 | voipmonk | check the sample voicemail config for anything relating to silence detection or activity |
| 19:48.42 | voipmonk | then tweak from there |
| 19:53.43 | *** join/#asterisk stix (n=stix@x1-6-00-1e-2a-87-8d-aa.k1.webspeed.dk) |
| 19:56.02 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 19:57.04 | stix | Hi guys. If I constantly need to count the members of a queue and put the results in a db, which approach should I take? Have some php-script running all the time, which uses the manager interface to check the queue? |
| 19:58.09 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 19:58.33 | ariel_ | hello everyone |
| 19:59.36 | *** join/#asterisk fskrotzki (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 19:59.49 | ariel_ | question about using system calls I need to be able to capture what is sent back from the perl call it seems no matter what we try we get "0" back all the time. Anyone have any idea on how to get other info returned? I am using asterisk 1.6 |
| 20:00.05 | voipmonk | stix - you could just grab orderlyq :) |
| 20:00.25 | russellb | ariel_: check out the SHELL function |
| 20:00.32 | stix | voipmonk: how do you mean? |
| 20:00.44 | voipmonk | well what else do u want to do, stix? |
| 20:00.55 | voipmonk | give me the eagle eye view |
| 20:01.31 | stix | voipmonk: I just want to list the number of callers waiting in the queue and put that number in a DB |
| 20:02.12 | ariel_ | russellb: I was under the impression the using system was a SHELL function can you explain more please? |
| 20:02.22 | russellb | *CLI> core show function SHELL |
| 20:03.33 | ariel_ | ok thank let me see if that works |
| 20:04.10 | *** join/#asterisk korihor (n=korihor@190.77.83.180) |
| 20:04.19 | *** join/#asterisk came0 (n=came0@rrcs-71-42-53-211.se.biz.rr.com) |
| 20:07.38 | *** join/#asterisk Naikrovek (n=jjohnson@98.214.112.102) |
| 20:08.24 | Naikrovek | timeshell: are you sure your phones are downloading the new configs? do your logs show the config file downloads |
| 20:08.35 | Naikrovek | i'm assuming you've not figured it out yet... |
| 20:08.41 | Naikrovek | probably incorrectly |
| 20:08.58 | bmoraca | effing polycoms :P |
| 20:09.12 | Naikrovek | thems fightin' words |
| 20:09.14 | Naikrovek | :P |
| 20:09.32 | Naikrovek | loves his 320s and 330s |
| 20:09.52 | bmoraca | i like them and think they work well and the quality is great...but configuration is a nightmare |
| 20:10.04 | casnik | I have a crap ton of 650 sip on hand right now |
| 20:10.23 | Naikrovek | bmoraca: it can be if you try to do a lot of stuff at once. getting them working then adding in config options works best for me |
| 20:10.25 | casnik | since my company installed Cisco and all |
| 20:10.36 | Naikrovek | casnik: looking to sell? |
| 20:10.43 | [TK]D-Fender | Configuration = no biggie |
| 20:10.58 | Naikrovek | why'd they install cisco? asterisk couldn't handle it (hah) or what |
| 20:11.08 | casnik | nope I am going to build an overnight Polycom powered cluster using all the phones processing power! |
| 20:11.20 | casnik | find the cure for cancer |
| 20:11.31 | Naikrovek | heh |
| 20:11.37 | casnik | well , at the time I wasn't a VOIP guy |
| 20:11.44 | casnik | (not that it would have helped) |
| 20:11.44 | Naikrovek | they do have reasonable processors in them |
| 20:11.47 | Naikrovek | k |
| 20:12.17 | casnik | and the contractor we got was incompetent .... so they just bit the bullet and installed all cisco voip |
| 20:12.46 | casnik | so I have about 300 phones and free reign to build whatever I want with em |
| 20:13.08 | *** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202) |
| 20:13.18 | Naikrovek | what in the world could you do with that many if people already have phones |
| 20:13.39 | Naikrovek | you /could/ give them to naikrovek free of charge |
| 20:13.39 | casnik | I dunno , L2Asterisk+Opensips I guess hehe |
| 20:13.47 | dustybin | i now have a polycom 321 sitting next to my bed, i have turned it on :D |
| 20:14.02 | casnik | I am in integration now , so I can sit here and puzzle myself to headblowup |
| 20:14.09 | dustybin | Running 'sip.id' |
| 20:14.30 | Naikrovek | dustybin: do you have it pointed at a tftp or ftp or ftps or http or https server to download a config? |
| 20:14.32 | casnik | (it's actually about 100 phones) |
| 20:14.51 | dustybin | Naikrovek: nothing at all, first im checking my dhcp server, make sure it can give out addresses |
| 20:15.08 | dustybin | Naikrovek: this is first time i have switched it on |
| 20:15.20 | *** join/#asterisk Micc (n=Micc@c-98-225-59-171.hsd1.wa.comcast.net) |
| 20:15.22 | dustybin | i have the main menu up now, the time 12:00 is flashing |
| 20:15.25 | *** join/#asterisk gabri-shatana (n=shatana@95.235.120.253) |
| 20:15.27 | gabri-shatana | hi |
| 20:15.50 | dustybin | ok dhcp server is running |
| 20:15.56 | dustybin | i need to google a guide |
| 20:15.56 | casnik | oh crap ... gabri-shatana is back ... hide the lettuce |
| 20:16.12 | Naikrovek | dustybin: okay. do you have an asterisk system already setup |
| 20:16.17 | gabri-shatana | i hate lettucce |
| 20:16.28 | casnik | we know ... |
| 20:16.37 | gabri-shatana | lol |
| 20:16.50 | dustybin | Naikrovek: yes, i need to setup a sip.conf for it first |
| 20:16.57 | gabri-shatana | Enter your USERNAME to connect to the 'asterisk' database: |
| 20:17.06 | gabri-shatana | what is it? |
| 20:17.31 | Gokee2 | Morning all, anyone feel like helping me figure out why I can't receive incoming calls from sipgate but outgoing works fine? sip.conf is at http://pastebin.com/d6d18dc20 and my dialplan is http://pastebin.com/d1f341a7c thanks! |
| 20:17.35 | Naikrovek | dustybin: make sure you set up the time server or it'll continue to blink the time |
| 20:17.42 | Naikrovek | Gokee2: nat? |
| 20:17.55 | Gokee2 | Naikrovek, Ya, I have a ipcop gateway |
| 20:18.04 | Naikrovek | i'm betting that's your problem |
| 20:18.11 | gabri-shatana | casnik, ? |
| 20:18.14 | dustybin | has a feeling this is going to take a _long_ time to setup |
| 20:18.15 | Naikrovek | it'll be the first place to look anyway |
| 20:18.20 | Naikrovek | dustybin: not that long |
| 20:18.44 | casnik | gabri-shatana, whatever you set it to? |
| 20:18.45 | Gokee2 | I have it setup to forward UDP for 5060 and 10000-20000 |
| 20:18.54 | casnik | gabri-shatana, root? |
| 20:18.57 | gabri-shatana | casnik, freepbx |
| 20:19.00 | dustybin | Naikrovek: instead of using tftp / ftp, cant i just put in the details on the phone, its only 1 phone |
| 20:19.03 | gabri-shatana | no isn't root |
| 20:19.12 | Naikrovek | dustybin: yes |
| 20:19.27 | dustybin | Naikrovek: i understand if i had a big network, but for just one phone... |
| 20:19.33 | Naikrovek | Gokee2: cool, i believe those are the correct ports |
| 20:19.45 | Naikrovek | dustybin: you can probably forget about sip.conf altogether then |
| 20:19.54 | Gokee2 | I am not real sure of where/what I should have in asterisk for nat stuff. I have tried stuff in both extensions.conf and in sip.conf |
| 20:19.57 | Naikrovek | http://your.phone's.ip.address/ |
| 20:20.15 | gabri-shatana | casnik, ? |
| 20:20.24 | spck | anyone get blocked callerid unmasking to work when not using a sip trunk? |
| 20:20.50 | casnik | gabri-shatana, I have no idea really .... I'm not the club pro |
| 20:21.02 | gabri-shatana | ok ty |
| 20:21.07 | casnik | =p |
| 20:21.07 | gabri-shatana | [TK]D-Fender, |
| 20:21.20 | Naikrovek | Gokee2: turn on sip debugging and try to make an incoming call to your pbx. then pastebin the result |
| 20:21.22 | *** join/#asterisk toddejohnson (n=toddejoh@ppp-70-226-213-11.dsl.spfdil.ameritech.net) |
| 20:21.27 | dustybin | Naikrovek: can i display what ip address the phone is using from the menu? |
| 20:21.41 | Gokee2 | Naikrovek, sip debugging? That sounds handy.... |
| 20:21.42 | Naikrovek | dustybin: from memory: uh.. menu, status, network, tcpip |
| 20:21.51 | dustybin | ok thank |
| 20:22.09 | dustybin | excellent |
| 20:22.16 | dustybin | 192.168.1.243 |
| 20:22.23 | [TK]D-Fender | gabri-shatana: GUI's are NOT supported here <--- |
| 20:22.34 | Naikrovek | dustybin: default User is Polycom (capital P) default passwd is 456 |
| 20:22.40 | dustybin | :-) |
| 20:22.56 | casnik | ah so that is a php error? |
| 20:22.58 | dustybin | im in :-) |
| 20:23.19 | Naikrovek | dustybin: i don't remember the exact setting you need to touch, but get ready for a lot of waiting while the phone reboots |
| 20:23.41 | Naikrovek | dustybin: set the time server, set the sip server, set the extension and authoirization (password) and make it match what you have in asterisk |
| 20:23.44 | dustybin | a lot of people say the web menu is cack, it looks good to me |
| 20:23.57 | dustybin | ok :-) |
| 20:24.15 | Naikrovek | dustybin: it reboots every time you make a change; makes it a real serious pain in the fanny for any more than 1 phone |
| 20:24.17 | dustybin | im not sure if my time server is listening on my local address on my server |
| 20:24.26 | dustybin | i see |
| 20:24.41 | dustybin | i use ntpd on my server |
| 20:24.50 | dustybin | will the phone look at that? |
| 20:24.57 | Naikrovek | if you tell it to |
| 20:25.01 | [TK]D-Fender | dustybin: If you tell it to. |
| 20:25.04 | Naikrovek | you have to tell it which server to pull time from it |
| 20:25.10 | dustybin | ok |
| 20:25.11 | [TK]D-Fender | Naikrovek: Ok, You get this one ;) |
| 20:25.11 | Naikrovek | s/from it/from/ |
| 20:25.27 | Gokee2 | Naikrovek, I added "sip debug" into the file sip.conf. restarted asterisk and tried a test call but I don't see any useful messages in /var/log/asterisk/? |
| 20:25.28 | Naikrovek | [TK]D-Fender: you take the harder problems like Gokee2's |
| 20:25.41 | [TK]D-Fender | Naikrovek: I wasted valuable keystrokes on capitalization & puntuatio :p |
| 20:25.46 | Naikrovek | [TK]D-Fender: lol |
| 20:25.52 | p3nguin | dustybin: lsof -i udp:123 |
| 20:26.07 | Naikrovek | Gokee2: sip debug is something you set up on the asterisk CLI |
| 20:26.16 | Gokee2 | Oooo |
| 20:26.40 | [TK]D-Fender | ok, checkout time, later all |
| 20:26.46 | Naikrovek | [TK]D-Fender: laters |
| 20:26.56 | *** join/#asterisk darkdrgn2k3 (n=darkdrgn@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 20:27.49 | darkdrgn2k3 | hey guys can any one help me figure out how to configure my nortel MCS voip account into asterisk? |
| 20:27.51 | Gokee2 | Ok I did "CLI> sip set debug SIP Debugging re-enabled"then tried a test call and got nothing..... And now it just spouted something |
| 20:28.08 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) |
| 20:28.19 | Naikrovek | Gokee2: if you got nothing when you attempted to make a call, then something on your network is blocking the call from coming in |
| 20:29.01 | dustybin | phone is restarting |
| 20:29.34 | Gokee2 | Ok, it has "From: "asterisk" <sip:asterisk@192.168.0.47>;tag=as5a0b7316" that *should read my external ip right? Which spot should I be setting nat on? In sip.conf at the sipgate spot? |
| 20:29.41 | dustybin | do polycoms use linux? |
| 20:29.57 | Naikrovek | i doubt it |
| 20:29.59 | dustybin | embedded linux |
| 20:30.14 | casnik | no they boot too slow |
| 20:30.21 | casnik | lol |
| 20:30.22 | Naikrovek | pfft |
| 20:31.16 | dustybin | WOW |
| 20:31.19 | dustybin | TIME WORKS!!!! |
| 20:31.33 | Naikrovek | i don't understand the linux fascination with boot time. a decent operating system boots maybe twice a year |
| 20:31.42 | Naikrovek | a decent unix operating system, rather |
| 20:31.44 | casnik | Naikrovek, I was just playin |
| 20:31.54 | Naikrovek | i know but some people really do care how fast linux boots |
| 20:31.57 | casnik | that is true |
| 20:32.16 | Gokee2 | Naikrovek, I think its mostly those strange people who like to turn there computers off |
| 20:32.16 | dustybin | there are 2 menus SIP LINES |
| 20:32.18 | casnik | a bad kernel config can take forever to boot |
| 20:32.42 | p3nguin | More accurately, a bad kernel won't boot at all. |
| 20:32.42 | dustybin | right i see |
| 20:32.47 | dustybin | i can have 2 lines on this phone |
| 20:32.47 | casnik | or that |
| 20:33.03 | Naikrovek | dustybin: yes by default the polycoms have two lines, and can handle two calls per line |
| 20:33.31 | Gokee2 | Naikrovek, So it has things like "Contact: <sip:asterisk@192.168.0.47>" thats never gonna work right? I mean there is no way to get back to my system.... |
| 20:34.35 | dustybin | what transport method should i specify: DNSaptr ? |
| 20:34.41 | dustybin | never heard of that.. |
| 20:35.01 | Naikrovek | i'm not sure what to say until you get your network config up and running. if your asterisk system isn't seeing the incoming calls then i can't diagnose asterisk problems |
| 20:35.04 | dustybin | TCPpreferred sounds better |
| 20:35.04 | Naikrovek | dustybin: that's fine |
| 20:35.11 | Naikrovek | dustybin: DNSaptr |
| 20:35.15 | dustybin | aye ok |
| 20:35.39 | Naikrovek | set the server to your asterisk server, uh.. i can't remember all the settings. let me poke around on my phone. |
| 20:35.52 | darkdrgn2k3 | anyone, help me configure outbound routes for my MCS SIP? |
| 20:36.30 | dustybin | I have filled in all the Identification bit |
| 20:36.46 | Naikrovek | dustybin: fill in your server and port for the server 1 and the outbound proxy |
| 20:36.47 | dustybin | on the Server part, i have specified the asterisk server and port |
| 20:37.29 | Naikrovek | okay and in the lines section fill in the username and password you have set up on your asterisk server, if you have them |
| 20:37.36 | dustybin | ok |
| 20:38.46 | dustybin | restarting phone |
| 20:39.15 | Naikrovek | it should come up and be visible in 'show sip peers' in asterisk if you've set everything up properly |
| 20:39.27 | dustybin | Naikrovek: i need to edit the lines section, then restart again! |
| 20:39.30 | casnik | MMO TIME! |
| 20:39.31 | casnik | peace! |
| 20:39.33 | dustybin | i just did the server section |
| 20:39.39 | Naikrovek | dustybin: hehe see? not so convenient |
| 20:39.43 | *** join/#asterisk ZX81 (n=ZX81@121.74.247.131) |
| 20:39.55 | Naikrovek | only gotta reboot once if you set it up in the config files |
| 20:39.58 | dustybin | Naikrovek: seems odd why its designed like that.. |
| 20:40.11 | Naikrovek | any more than one phone and config file is a necessity |
| 20:40.13 | gabri-shatana | hi |
| 20:40.21 | gabri-shatana | i have a voip in number |
| 20:40.25 | *** join/#asterisk jlnt (n=jlnt@cisco2.jlmail.com) |
| 20:40.27 | gabri-shatana | i want to use it for control my p |
| 20:40.31 | gabri-shatana | *pc |
| 20:40.31 | gabri-shatana | like |
| 20:40.43 | gabri-shatana | if i call and i press 2 something happens |
| 20:40.54 | Naikrovek | uh |
| 20:40.59 | p3nguin | I think this was already covered. |
| 20:41.04 | lowtek | gabri-shatana: Your question was answered this morning gabri, use the System() command and RTFM |
| 20:41.11 | p3nguin | Seems like deja vu. |
| 20:41.28 | gabri-shatana | ok.. |
| 20:41.38 | jlnt | sup |
| 20:41.41 | jlnt | quick question |
| 20:41.45 | Naikrovek | ask it |
| 20:41.49 | jlnt | I know this is an asterisk now question |
| 20:42.02 | jlnt | but how do I set DIDs for Dahdi channels now |
| 20:42.52 | jlnt | I tried using the Zap Channel DIDs |
| 20:42.53 | Naikrovek | jlnt: maybe ask in freepbx |
| 20:43.08 | Naikrovek | jlnt: i've done this but i don't know the details atm |
| 20:43.18 | voipmonk | zzzz |
| 20:43.27 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 20:43.33 | jlnt | yeah |
| 20:43.40 | jlnt | it's seems kinda tricky now :P |
| 20:44.11 | wcselby | there's an asterisk-gui channel\ |
| 20:44.16 | dustybin | i have communication :D |
| 20:44.20 | dustybin | [Sep 14 21:43:55] NOTICE[30612]: chan_sip.c:20032 handle_request_register: Registration from '<sip:192.168.1.65@192.168.1.65>' failed for '192.168.1.243' - No matching peer found |
| 20:44.22 | darkdrgn2k3 | every one is dead in -gui |
| 20:44.22 | wcselby | as in, #asterisk-gui |
| 20:44.46 | wcselby | ah well, I can't help you either way |
| 20:45.15 | Naikrovek | wcselby: you're thinking #asterisknow |
| 20:45.35 | dmz | hey y'all, if i have a sip client that doesn't have a "transfer" button, how can i transfer a call back into another extension |
| 20:45.53 | Naikrovek | dustybin: do you have a 'host' line in your extension config |
| 20:46.02 | wcselby | asteriskNow just uses the asterisk-gui as a front end for asterisk, I thought. that's why I suggested #asterisk-gui (which I've been to before and gotten an answer) |
| 20:46.02 | SuPrSluG | features.conf |
| 20:46.12 | Naikrovek | wcselby: they use freepbx |
| 20:46.19 | Naikrovek | wcselby: since asterisknow v 1.5 |
| 20:46.27 | wcselby | ahh |
| 20:46.32 | wcselby | oh well, I don't use either |
| 20:46.37 | SuPrSluG | dmz:features.conf |
| 20:46.39 | wcselby | i tried using the gui, it broke my asterisk :P |
| 20:46.51 | wcselby | i didn't spend a lot fo time trying to make it work after that |
| 20:47.59 | Naikrovek | wcselby: really? i find it much easier to use freepbx on systems that have it. |
| 20:48.13 | wcselby | i haven't tried freepbx yet |
| 20:48.56 | wcselby | i meant I tried installing asterisk-gui and I couldn't get it to work the way i wanted, and my system stopped responding the way i wanted, so I uninstalled it and just use the config files now instead of gui |
| 20:49.06 | wcselby | i may give freepbx a try with my next install |
| 20:49.06 | Naikrovek | wcselby: yeah fair enough |
| 20:49.14 | darkdrgn2k3 | whats the defautl l/p for freepbx? |
| 20:49.23 | wcselby | ~freepbx |
| 20:49.23 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 20:49.32 | wcselby | sorry I don't know |
| 20:49.34 | Naikrovek | wcselby: freepbx has a lot of pretty nice features built in |
| 20:49.45 | wcselby | that's what I've heard |
| 20:49.47 | Naikrovek | darkdrgn2k3: freepbx/fpbx i think |
| 20:50.02 | Naikrovek | i should stop answering those questions in here |
| 20:50.18 | darkdrgn2k3 | Naikrovek: just curious.. im not really using freepbx |
| 20:52.50 | darkdrgn2k3 | ok is there any way to test an outbound sip trunk without logging in? |
| 20:52.53 | *** join/#asterisk jasonwoot (n=some@69.73.89.233) |
| 20:53.21 | gabri-shatana | [Sep 14 22:53:08] NOTICE[2427]: chan_sip.c:16929 reload_config: Unable to load config sip.conf |
| 20:53.24 | gabri-shatana | what mean? |
| 20:53.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:53.46 | darkdrgn2k3 | gabri-shatana: that your sip.conf is missing or invalid? |
| 20:53.53 | gabri-shatana | i check |
| 20:54.01 | Naikrovek | gabri-shatana: missing, incorrect permissions, or locked for reading somehow |
| 20:55.41 | gabri-shatana | http://pastebin.com/d12b92e93 |
| 20:56.45 | Naikrovek | gabri-shatana: is that your sip.conf |
| 20:56.50 | gabri-shatana | yeah |
| 20:56.54 | dustybin | [Sep 14 21:56:29] NOTICE[30612]: chan_sip.c:20032 handle_request_register: Registration from '<sip:192.168.1.65@192.168.1.65>' failed for '192.168.1.243' - No matching peer found |
| 20:56.57 | dustybin | strange |
| 20:57.09 | dustybin | sip:192.168.1.65@192.168.1.65 <-- that looks wrong |
| 20:57.19 | Naikrovek | dustybin: yeah it does. |
| 20:57.40 | gabri-shatana | Naikrovek, where's the error? |
| 20:57.49 | Naikrovek | gabri-shatana: i don't know |
| 20:57.58 | gabri-shatana | mhh |
| 20:58.12 | Naikrovek | is ; the comment character for asterisk config files? |
| 20:58.20 | p3nguin | yes |
| 20:58.35 | dustybin | there is a server section > outbound proxy > server 1 > server 2 |
| 20:58.41 | gabri-shatana | what's the command to restart asterisk? |
| 20:58.51 | dustybin | there is a line section > identification > server 1 > server 2 |
| 20:59.03 | Naikrovek | gabri-shatana: asterisk -rx "module reload" |
| 20:59.10 | p3nguin | gabri-shatana: oddly, "restart" will do it. |
| 20:59.17 | Naikrovek | dustybin: on your phone? |
| 20:59.23 | dustybin | in the web menu |
| 20:59.25 | dustybin | yes |
| 20:59.30 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:59.48 | Naikrovek | dustybin: address = your extension under line config |
| 20:59.53 | gabri-shatana | [Sep 14 22:59:45] NOTICE[2427]: chan_sip.c:16929 reload_config: Unable to load config sip.conf |
| 20:59.57 | gabri-shatana | nothing.. |
| 20:59.57 | dustybin | OHHHHH |
| 21:00.07 | Naikrovek | dustybin: authid = your extension also |
| 21:00.27 | Naikrovek | label = what the phone shows next to the phone icon |
| 21:00.48 | Naikrovek | gabri-shatana: is the file readable by root or whomever you're running asterisk as? |
| 21:00.52 | gabri-shatana | debian-server:/etc/asterisk# ls -ld /etc/asterisk /etc/asterisk/sip.conf |
| 21:00.52 | gabri-shatana | drwxr-xr-x 3 asterisk asterisk 4096 14 set 22:57 /etc/asterisk |
| 21:00.52 | gabri-shatana | -rw-r----- 1 root root 422 14 set 22:57 /etc/asterisk/sip.conf |
| 21:01.07 | Naikrovek | gabri-shatana: use a pastebin from now on, but i see your issue |
| 21:01.11 | gabri-shatana | ok |
| 21:01.16 | gabri-shatana | permission problem? |
| 21:01.26 | Naikrovek | gabri-shatana: chmod asterisk.asterisk /etc/asterisk/sip.conf |
| 21:01.39 | gabri-shatana | chmod: invalid mode: `asterisk.asterisk' |
| 21:01.46 | raden_work | is there a click to call interface anyone can recomend for windows ? |
| 21:01.55 | Naikrovek | gabri-shatana: sorry, meant chown |
| 21:01.56 | p3nguin | chown asterisk:asterisk /etc/asterisk/sip.conf |
| 21:01.59 | Naikrovek | gabri-shatana: not chmod |
| 21:02.04 | Naikrovek | thank you p3nguin |
| 21:02.26 | gabri-shatana | ok |
| 21:02.29 | gabri-shatana | now sip work |
| 21:02.53 | gabri-shatana | System() is a functio? |
| 21:02.58 | gabri-shatana | *function? |
| 21:03.00 | KingDavidNYC | Hi, anybody please can give me some light as to what files I need to upload to a polycom phone, please? |
| 21:03.30 | KingDavidNYC | ... and where can I find those files |
| 21:03.46 | Naikrovek | KingDavidNYC: download firmware, it has those files |
| 21:04.53 | KingDavidNYC | where?.. as far as I know, you need to be a dealer to download the files from polycom |
| 21:05.02 | Naikrovek | haha no |
| 21:05.10 | Naikrovek | what model of phone do you have |
| 21:05.22 | KingDavidNYC | 601 |
| 21:05.54 | Naikrovek | KingDavidNYC: http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip601.html |
| 21:07.15 | wcselby | anyone ever seen this error - ERROR[9941]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe |
| 21:07.25 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:07.29 | wcselby | it happens after an agi call, sometimes |
| 21:07.43 | KingDavidNYC | Naikrovek: wow man, that's owesome tip |
| 21:07.44 | Naikrovek | wcselby: nope |
| 21:07.58 | Naikrovek | KingDavidNYC: read the docs and you'll be a polycom expert in no time |
| 21:08.19 | KingDavidNYC | Naikrovec: thank you man |
| 21:08.59 | wcselby | i can save that error for another day |
| 21:09.16 | wcselby | doesn't seem to be having a negative effect on the pbx |
| 21:09.58 | Naikrovek | KingDavidNYC: no problem |
| 21:11.04 | dustybin | grrr my lines keep on defaulting back to old settings after i submit a save |
| 21:11.41 | p3nguin | You should probably configure the files and stick them on the tftpd. |
| 21:11.59 | dustybin | yep |
| 21:12.24 | p3nguin | You've already spent far more time in the phone than you would have in a text editor. |
| 21:12.28 | *** join/#asterisk friehmaen (i=freeman@xers.de) |
| 21:12.37 | *** join/#asterisk keulin (n=cray@bne75-6-82-229-246-155.fbx.proxad.net) |
| 21:14.10 | Naikrovek | yes, p3nguin is right on that. download the firmware and look at the sample files |
| 21:14.40 | p3nguin | It's really not that much work even if you do have only one phone. |
| 21:14.42 | Naikrovek | then change 000000000000.cfg to match your phone mac address, and just set everything up. when that's all done, reset the phone to forget its config (it's in the menu) |
| 21:15.13 | wcselby | adios folks |
| 21:23.16 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 21:23.20 | raden_work | our caller ID from vitelity never has a 1 preceding it which makes callback a PITA is there a way to fix this ? |
| 21:23.53 | *** join/#asterisk jolucara (n=jolucara@186.97.14.138) |
| 21:23.56 | p3nguin | raden_work: You know why there is no 1 on it? |
| 21:24.32 | raden_work | i do not |
| 21:24.37 | p3nguin | raden_work: 1 isn't part of the phone number. But yes, there are two ways you can fix it. |
| 21:24.45 | lowtek | raden_work: It's not a matter of fixing, nothing is broke. Just use Dial(...${EXTEN:1}...) to trim off the preceeding 1. |
| 21:24.47 | raden_work | that would make sense |
| 21:25.30 | p3nguin | raden_work: You can either modify your outgoing dial plan to add put a 1 on the front of 10-digit numbers when you dial, or you could modify the incoming caller ID to add a 1 on the front of it. The former is the preferred method. |
| 21:25.49 | raden_work | now this presents other issues |
| 21:26.03 | raden_work | our one provider has a 1 in front of the number on caller is |
| 21:26.04 | raden_work | id |
| 21:26.04 | p3nguin | What issue? |
| 21:26.25 | p3nguin | Tell them to fix it, because that's wrong. |
| 21:26.44 | p3nguin | 1 is an access number, just like 00 is an access number for a lot of international calling. |
| 21:27.00 | lowtek | raden_work: vitelity sends 10-digit caller id's. |
| 21:27.10 | raden_work | correct |
| 21:27.15 | p3nguin | That's what he said already. |
| 21:27.22 | raden_work | callcentric and broadvoice i get 11 |
| 21:27.24 | hesco | When one party hangs up on a Bridg()'d conversation, is there any sort of feedback? I'm getting BRIDGERESULT=SUCCESS when the connection is made. What happens later? 'NoOp(Our Bridge() result is: ${BRIDGERESULT})' is yielding me: 'Our Bridge() result is: ' |
| 21:27.48 | p3nguin | raden_work: Your DID from broadvoice has a 1 on the front? |
| 21:27.57 | lowtek | Just add a 1 ... what's the big deal? Am I missing something? |
| 21:27.59 | p3nguin | raden_work: I mean, the caller ID on that DID. Sorry. |
| 21:28.31 | raden_work | p3nguin, yeah |
| 21:28.37 | raden_work | broadvox sorry |
| 21:28.48 | p3nguin | raden_work: Like I said, the preferred method is to allow both 10 and 11 digit dial. |
| 21:29.13 | raden_work | p3nguin, ill play with my dial plan |
| 21:29.22 | p3nguin | exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@yourprovider) |
| 21:29.28 | p3nguin | exten => _NXXNXXXXXX,n,Dial(SIP/1${EXTEN}@yourprovider) |
| 21:29.37 | raden_work | thank you :) |
| 21:29.47 | Naikrovek | p3nguin knows what he's talking about |
| 21:29.55 | p3nguin | Modify according to your priorities if you need to. |
| 21:29.57 | hesco | how do I tell the dialplan when a Bridge()'d connection is broken by one of the parties to the call? Is there some Channel Veriable I'm missing? |
| 21:30.24 | dustybin | <PROTECTED> |
| 21:30.31 | Naikrovek | dustybin: there ya go |
| 21:30.40 | dustybin | THANKS FOR YOUR HELP :) |
| 21:30.42 | Naikrovek | now set up a meetme conference and see if you can connnect to it |
| 21:30.46 | Naikrovek | s/connnect/connect/ |
| 21:32.05 | dustybin | it works perfectly! |
| 21:32.11 | dustybin | excellent sound quality |
| 21:32.14 | Naikrovek | yes |
| 21:32.25 | Naikrovek | one of the advantages of polycoms, they sound great |
| 21:32.40 | Naikrovek | even the speakerphone is awesome on those |
| 21:32.47 | dustybin | ACE |
| 21:32.53 | Naikrovek | and you can set them up for paging & intercom as well |
| 21:33.05 | dustybin | people are sleeping, but im going to test the ring tone |
| 21:33.15 | Naikrovek | sleeping? where are you |
| 21:33.38 | dustybin | at home |
| 21:33.43 | Naikrovek | har har |
| 21:33.43 | voipmonk | on the mothership |
| 21:33.44 | dustybin | UK |
| 21:33.48 | Naikrovek | UK nice |
| 21:34.09 | *** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com) |
| 21:35.44 | garymc | If i wanted to just use my asterisk box to make calls through the internet, do i need some sort of telco plan to make calls to normal PSTN lines? |
| 21:36.03 | Naikrovek | garymc: yes. but normal lines are POTS lines. PSTN = voip |
| 21:36.10 | Naikrovek | ~pstn |
| 21:36.11 | infobot | hmm... pstn is Public Switched Telephone Network, or "please stop the nonsense" |
| 21:36.12 | garymc | oh sorry |
| 21:36.21 | Naikrovek | oh wait maybe i'm wrong |
| 21:36.28 | garymc | lol |
| 21:36.37 | p3nguin | PSTN is mostly the same as POTS. |
| 21:36.40 | Naikrovek | I always thought PSTN = packet switched telephone network. POTS = plain old telephone service |
| 21:36.54 | p3nguin | public switched telephone network |
| 21:36.54 | Naikrovek | garymc: nevermind me then |
| 21:36.57 | garymc | I was just trying to understand how that would work |
| 21:37.07 | Naikrovek | garymc: but yes you gotta have service, or a LOT of dundi peers |
| 21:37.22 | p3nguin | raden_work: I even allow 7-digit dial for local numbers. You'll probably want to configure that, too. If you are in area code 314, for example, you can use something like the following: exten => _NXXXXXX,n,Dial(SIP/1314${EXTEN}@yourprovider) |
| 21:37.31 | garymc | Who provides that service? I thought the asterisk box did all that for you some how? |
| 21:37.52 | raden_work | p3nguin, thanks :) |
| 21:38.01 | Naikrovek | garymc: that would be a voice provider. there are some really super cheap ones, and there are some good ones, some are both i think |
| 21:38.03 | Naikrovek | ~voiceprovider |
| 21:38.06 | p3nguin | garymc: Make calls... as opposed to make and receive them? |
| 21:38.24 | garymc | like gives you a telephone number etc? |
| 21:38.38 | Naikrovek | garymc: yes. infobot knows but i forgot the keyword |
| 21:38.42 | p3nguin | garymc: To get a phone number, you'll need what's called a DID. |
| 21:39.03 | p3nguin | garymc: To make calls, you need what's known as termination. |
| 21:39.11 | garymc | And id purchase that from say BT - Then they would charge me to make and recieve calls |
| 21:39.21 | garymc | throught voip |
| 21:39.21 | p3nguin | garymc: The two are mutually exclusive. |
| 21:39.46 | p3nguin | garymc: You can get a free DID, but it won't be a local phone number. |
| 21:39.51 | Naikrovek | you're just confusing the question i think p3nguin |
| 21:39.53 | garymc | Im going with ISDN for now , but im just curious incase i had to switch in future |
| 21:40.12 | p3nguin | naikrovek: hmm? |
| 21:40.37 | lowtek | lol |
| 21:40.39 | Naikrovek | p3nguin: he just wants to know what it takes to get a fully functioning telephone |
| 21:40.40 | garymc | so i suppose its still pretty expensive to set all that up? |
| 21:40.46 | p3nguin | Let's try this a different way. |
| 21:40.46 | Naikrovek | garymc: no |
| 21:40.53 | p3nguin | garymc: Do you want to receive calls? |
| 21:40.59 | p3nguin | garymc: Do you want to make outgoing calls? |
| 21:41.02 | garymc | cos id need a good Broadband connection: Thats costs alot of cash where my office is |
| 21:41.13 | garymc | p3nguin both |
| 21:41.20 | p3nguin | Then you need a DID and termination. |
| 21:41.46 | Naikrovek | a DID (phone number) costs like $1/month with termination service |
| 21:41.52 | p3nguin | You can get termination (outgoing calls) from VoIP.ms for as low as 1.04 cents per minute. |
| 21:41.59 | Naikrovek | yes it's pretty cheap |
| 21:42.13 | garymc | ok im in the uk though, im sure it costs more |
| 21:42.21 | p3nguin | You can get a free DID from ipcomms.net, but it won't be a local number. |
| 21:42.37 | p3nguin | oh, that makes things a little different. |
| 21:43.16 | p3nguin | I'm not familiar with any UK providers at all. |
| 21:43.34 | garymc | yeah me neither lol |
| 21:43.47 | Naikrovek | maybe dustybin knows |
| 21:43.47 | garymc | Im using BT the main ones for our ISDN30 |
| 21:43.51 | Naikrovek | he's from the UK |
| 21:59.24 | raden_work | is threre any click to dial apps anyone can recomend for windows ? that would work with browser, groupwise, and quickbooks. |
| 21:59.26 | raden_work | ? |
| 22:00.04 | p3nguin | http://gizmo5.com/pc/ maybe? |
| 22:01.22 | drmessano | Wait |
| 22:01.30 | drmessano | Browser, groupwise, and quickbooks |
| 22:01.35 | drmessano | WTF |
| 22:02.02 | drmessano | I didnt know we had app_quickbooks working yet |
| 22:02.09 | Qwell | drmessano: 1.6.3 |
| 22:02.20 | drmessano | Sweet! |
| 22:02.32 | drmessano | I was getting tired of app_msmoney |
| 22:03.02 | *** join/#asterisk profXavier (n=chatzill@unaffiliated/neverblue) |
| 22:03.17 | drmessano | raden_work: May I ask you a question? |
| 22:05.42 | profXavier | when someone calls out, their phone works fine, when someone calls them, it goes directly into voicemail. looking at * I get this error: "Remote host can't match request NOTIFY to call '5239fdfb15d20e5153a2f7365ce2ea4d at 69.80.215.12'. Giving up" |
| 22:06.16 | Naikrovek | profXavier: phones configured for "nat=yes" ? |
| 22:06.47 | profXavier | yes |
| 22:06.53 | raden_work | drmessano, yes .. |
| 22:07.14 | profXavier | Naikrovek: yes, nat=yes |
| 22:07.20 | Naikrovek | profXavier: okay |
| 22:07.27 | profXavier | should it be ? |
| 22:08.15 | raden_work | drmessano, we make tons of calls so either a click to dial for just a browser or all of it would even be better |
| 22:08.15 | Naikrovek | profXavier: yes, probably. can you turn on SIP debugging, make an incoming call that would incorrectly go to voicemail, then paste the log into a pastebin please? |
| 22:08.30 | dwery | hello. is there a way to have an hint extension to monitor the status of a DAHDI channel? |
| 22:08.39 | Naikrovek | raden_work: there are those for outlook, and some Jabber IM programs if you have that all set up properly |
| 22:09.00 | profXavier | how can you see the current level of debugging set ? |
| 22:09.11 | *** part/#asterisk kb3ien (n=kb3ien@ool-45766a2d.dyn.optonline.net) |
| 22:09.39 | raden_work | profXavier, like sip debugging ? |
| 22:09.43 | Naikrovek | profXavier: "asterisk -r" if you see a ton of stuff fly by (not just colored stuff) then it's probably on. if you only see activity when calls are made/received it's not on |
| 22:09.45 | hesco | garymc: you want an ITSP |
| 22:09.50 | hesco | ~itsp |
| 22:09.51 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:10.02 | Naikrovek | hesco: he's gone i think |
| 22:10.09 | profXavier | Naikrovek: roger that, ill do what you said |
| 22:10.25 | raden_work | sip set debug on |
| 22:10.27 | Naikrovek | profXavier: i'm going AFK, but lots of folks in here can help you. |
| 22:10.27 | raden_work | sip set debug off |
| 22:10.34 | raden_work | core set verbose # |
| 22:10.43 | raden_work | Naikrovek, later |
| 22:11.11 | hesco | sorry, should have finished catching up on the scroll buffer I guess, before I posted. |
| 22:11.13 | KingDavidNYC | raden_work: I am very good at click-to-call apps.. anything you need, just let me know |
| 22:11.18 | hesco | thought I had reached the end |
| 22:11.51 | raden_work | KingDavidNYC, quickbooks ? |
| 22:13.31 | KingDavidNYC | I've done for outlook |
| 22:13.49 | KingDavidNYC | but dont see why not |
| 22:14.50 | KingDavidNYC | is quickbooks tapi compliant? |
| 22:15.28 | drmessano | How does quickbooks apply? |
| 22:16.06 | jasonwoot | ~quickbooks is hot |
| 22:16.07 | infobot | okay, jasonwoot |
| 22:16.27 | jasonwoot | lol, I can update the bot? |
| 22:16.29 | drmessano | quickbooks is the new sexy telephony frontend, apparently |
| 22:18.03 | drmessano | Screw deskphones, softphones, and PIM integration, Quickbooks is poised to replace BearShare as the #1 telephony app |
| 22:19.10 | jasonwoot | can you restrict infobot so that it only responds in certain channels, and doesn't allow updates? |
| 22:19.53 | drmessano | infobot, can you be contained, yo? |
| 22:20.03 | drmessano | infobot: can you be contained, yo? |
| 22:20.24 | drmessano | He said "Bitch, please", I do believe |
| 22:20.36 | drmessano | ~slap infobot |
| 22:20.37 | infobot | ACTION slaps infobot, keep your grubby fingers to yourself! |
| 22:20.46 | drmessano | ~recursion |
| 22:20.47 | infobot | To understand recursion, you must first understand recursion. |
| 22:20.59 | profXavier | ok, here is my pastebin --> http://pastebin.com/d6207758a |
| 22:26.43 | *** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar) |
| 22:29.20 | *** join/#asterisk jaytee (n=jaytee@unaffiliated/jaytee) |
| 22:31.00 | profXavier | I just logged in under the user's account on my softphone and its working |
| 22:31.12 | profXavier | must be something on his PC |
| 22:31.30 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 22:32.49 | raden_work | KingDavidNYC, i would presume sooo because it works with outlook like click on a address to email out of outlook from quickbooks |
| 22:33.45 | Qwell | raden_work: that would be standard URI handling... |
| 22:33.59 | raden_work | Qwell, sorry |
| 22:34.05 | raden_work | KingDavidNYC, still around ? |
| 22:34.15 | *** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca) |
| 22:41.24 | *** join/#asterisk thx2000 (n=bob@ip68-96-95-215.oc.oc.cox.net) |
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| 22:42.35 | KingDavidNYC | raden_work: big difference. I would assume quickbooks uses the the net framework to pull info from outlook |
| 22:42.57 | KingDavidNYC | raden_work: in any case, anything tapi, just let me know |
| 22:43.19 | *** join/#asterisk jetlagmk2 (i=jetlag@70.104.75.14) |
| 22:43.25 | thx2000 | I've got a couple soft phones connecting in via SIP through a PPTP VPN tunnel on a pfsense firewall. I've tried just about every config option I can throw at it. Sometimes it will work and sometimes it wont. Dialing internal apps works fine (VM, Conferences, other extensions), however dialing and receiving calls from a trunk fails probably 65% of the time |
| 22:43.27 | thx2000 | Any Ideas? |
| 22:48.32 | p3nguin | Is that normal, to have your phones connecting on a VPN? |
| 22:51.15 | thx2000 | It's a decent way to avoid nat, add some security and permit multiple services to a roadwarrior |
| 22:53.46 | *** join/#asterisk ebroad (n=EB@72.11.213.195) |
| 22:54.04 | *** part/#asterisk ebroad (n=EB@72.11.213.195) |
| 22:54.32 | p3nguin | So it was more to permit a full solution for a portable client, as opposed to simply trying to contain the voice traffic within an encrypted channel? |
| 22:54.47 | *** join/#asterisk iksik (i=xk@livedata.pl) |
| 22:55.49 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 22:58.27 | scunizi | p3nguin: using vpn makes the phone think it's connected from within the lan of the server and avoids firewall issues. on both ends. |
| 22:58.27 | thx2000 | A little of both, however security was not as big of a concern. |
| 22:59.38 | thx2000 | If security was a primary concern, I wouldn't have gone w/ PPTP, but regardless it's still better than raw sip |
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| 23:10.48 | *** join/#asterisk jolucara (n=jolucara@190.102.217.169) |
| 23:13.57 | *** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net) |
| 23:27.38 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 23:30.20 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 23:31.01 | KingDavidNYC | anybody please help me program this polycom phone :( |
| 23:31.33 | KingDavidNYC | I cant figure out why it says "url call is disabled" |
| 23:32.01 | retentiveboy | Anyone using the Access-URL SIP header to make Polycom phones load a XML page when a call comes in? Wondering if it's possible before I spend too much time on it :) |
| 23:34.04 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 23:34.26 | retentiveboy | KingDavidNYC: Are you configuring the phone from the keypad or what? |
| 23:37.19 | KingDavidNYC | retentiveboy: no man, from the web site... actually I am doing a "standalone" provisioning |
| 23:37.57 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:37.57 | KingDavidNYC | retentiveboy: I dont get it, I have network access on that phone |
| 23:38.10 | retentiveboy | KingDavidNYC: Sorry, I've only ever set them up to load configs on boot from m y * machine. |
| 23:38.14 | *** join/#asterisk martianixor (n=martiani@unaffiliated/martiancode) |
| 23:38.45 | KingDavidNYC | ok |
| 23:39.18 | Naikrovek | KingDavidNYC: yes you can configure them to dial SIP urls |
| 23:39.18 | Naikrovek | but it's been a while since I've done it |
| 23:39.25 | Naikrovek | I think the default behavior is to dial SIP URLS |
| 23:39.31 | Naikrovek | like, before ANY config |
| 23:39.55 | KingDavidNYC | Naikrovek: I just wanted it to register to an asterisk server |
| 23:40.01 | Naikrovek | that is easier |
| 23:40.05 | dwery | anyone has mwi working on the thomson st2030? |
| 23:40.48 | KingDavidNYC | Naivkrovek: thanks to a doc you gave me, I found how to do the "standalone" programming from the web |
| 23:41.05 | Naikrovek | KingDavidNYC: cool. pointing it toward an asterisk server is easier than doing standalone i think |
| 23:41.08 | *** join/#asterisk jolucara (n=jolucara@186.97.25.153) |
| 23:41.20 | KingDavidNYC | but I dont know why the heck it is not sending anything to the server |
| 23:41.28 | KingDavidNYC | the ip is correct |
| 23:41.46 | retentiveboy | If you were using configs on the server, we could look at the :) |
| 23:42.03 | retentiveboy | s/at the/at them/ |
| 23:42.29 | KingDavidNYC | thank you, but I am using configs |
| 23:44.09 | KingDavidNYC | what does transport=Naptr means? |
| 23:44.10 | *** join/#asterisk DrukenLPY (n=jdumais@70.54.242.169) |
| 23:45.33 | retentiveboy | The default sip.cfg has voIpProt.server.1.transport="DNSnaptr". I've not overridden that and mIne work. |
| 23:45.36 | *** join/#asterisk Schmee (n=zaphod@ppp100-124.static.internode.on.net) |
| 23:46.37 | dwery | It seems that the "sip:sip" bug in the mwi notifications is still there: To: <sip:sip:200@192.168.2.20:5060;user=phone>;tag=c0a80101-2de3 |
| 23:51.00 | *** part/#asterisk scunizi (n=scunizi@69.199.151.114) |
| 23:51.25 | *** part/#asterisk _joe (n=joseph@craptome.org) |
| 23:51.53 | *** part/#asterisk Schmee (n=zaphod@ppp100-124.static.internode.on.net) |
| 23:53.11 | KingDavidNYC | retentiveboy: on the keypad, sip configuration, server, I have as only options for transport: naptr, tcp only, udp only |
| 23:53.59 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
