| 00:00.15 | hesco | [TK]D-Fender: did as you suggested, /usr/sbin/dahdi_cfg returned a prompt, I restarted *, then tested my MeetMe extension again and got the same error |
| 00:01.13 | Qwell | hesco: You need dahdi_dummy, yes |
| 00:02.17 | Aiatek | hesco did you did make config when you compiled linux-tool |
| 00:02.18 | Aiatek | ? |
| 00:03.01 | hesco | I would imagine I did, will go redo that just in case |
| 00:04.09 | [jmc] | hmm |
| 00:04.25 | [jmc] | what if I chose to disable loading of extensions.ael? |
| 00:04.34 | [jmc] | and doing it all through extensions.conf? |
| 00:04.42 | [jmc] | do I have to disable a specific module? |
| 00:05.30 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 00:09.55 | hesco | ok, did make; make install in dadhi-linux, then make; make config; make install in dadhi-tools, then restarted *, tested again to see the same error: "app_meetme.c:861 build_conf: Unable to open pseudo device" |
| 00:11.59 | Aiatek | try this |
| 00:13.05 | Aiatek | dahdi_cfg -vvvvvvvvvvvv |
| 00:13.32 | Aiatek | paste the output |
| 00:15.05 | Qwell | load dahdi_dummy... |
| 00:18.18 | *** join/#asterisk propellerhead (n=yogurt2u@host192.190-230-170.telecom.net.ar) |
| 00:31.38 | [jmc] | hey guys |
| 00:31.50 | [jmc] | is this syntax correct for extensions.conf in asterisk 1.4? |
| 00:32.04 | [jmc] | exten => s,3,Set(TIMEOUT(digit)=5) |
| 00:32.05 | [jmc] | exten => s,4,Set(TIMEOUT(response)=10) |
| 00:32.19 | [jmc] | or is it an * 1.6 thing? |
| 00:32.20 | [TK]D-Fender | [jmc]: yes |
| 00:32.32 | [jmc] | hmm |
| 00:32.34 | [TK]D-Fender | [jmc]: That's 1.2+ |
| 00:32.40 | [jmc] | oh |
| 00:32.43 | [jmc] | ok great :) |
| 00:32.51 | [jmc] | well so that's not the problem |
| 00:32.58 | [jmc] | I've set up this little menu |
| 00:33.10 | [jmc] | I can hear "welcome" when I dial |
| 00:33.15 | [jmc] | and then * hangs up |
| 00:33.22 | [jmc] | without waiting for me to dial anything |
| 00:33.28 | [jmc] | let me pastebin that |
| 00:34.18 | *** part/#asterisk jsolis (n=Jimmy@190.43.77.200) |
| 00:34.59 | [jmc] | http://pastebin.com/m6e48afad |
| 00:35.06 | [jmc] | here it is, my extensions.conf |
| 00:35.31 | [jmc] | should I add a WaitExten() to make it wait for something to be dialed? |
| 00:35.45 | [jmc] | the relevant part is [menu] of course |
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| 00:36.05 | [jmc] | I thought digit timeout and response timeout were enough |
| 00:39.34 | [TK]D-Fender | [jmc]: autofallthough=no <- what you should have for the typical "when 's' runs out we're an IVR" mode of working. |
| 00:39.39 | [TK]D-Fender | [jmc]: Or use WaitExten() |
| 00:40.17 | [TK]D-Fender | [jmc]: this can be set unger [globals] |
| 00:40.20 | [TK]D-Fender | under* |
| 00:40.47 | [jmc] | in sip.conf, right? |
| 00:41.14 | [TK]D-Fender | [jmc]: extensions.conf |
| 00:41.22 | [TK]D-Fender | [jmc]: This is ALL dialplan. |
| 00:41.33 | [TK]D-Fender | [jmc]: Your call is in.. everything else if call processing |
| 00:41.39 | [TK]D-Fender | s/if/is |
| 00:41.39 | [jmc] | autofallthrough is a dialplan option? |
| 00:41.45 | [jmc] | ok |
| 00:41.54 | [TK]D-Fender | [jmc]: Yes, which is why you see it listed during CLI |
| 00:43.26 | [jmc] | I'm trying this: |
| 00:43.29 | [jmc] | [globals] |
| 00:43.35 | [jmc] | autofallthrough = no |
| 00:43.38 | [jmc] | the rest is the same |
| 00:46.09 | [TK]D-Fender | [jmc]: Sorry, that should be under [general] |
| 00:46.24 | [jmc] | oh, ok ;) |
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| 00:57.00 | [jmc] | what's so bad about ALSA in Asterisk? |
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| 00:57.37 | [jmc] | [TK]D-Fender: btw, it works ;) |
| 00:57.39 | [TK]D-Fender | [jmc]: Whats the point of using it at all? |
| 00:59.00 | [jmc] | OSS seems to take exclusive control over my sound device |
| 00:59.09 | [jmc] | and, well, why not? ^^ |
| 00:59.44 | [jmc] | now for the last question tonight |
| 00:59.56 | [jmc] | this one may be enough simple |
| 01:00.20 | [jmc] | I have this IVR starting with the s extension |
| 01:00.39 | [jmc] | and a SIP user which connects to my * |
| 01:00.56 | [jmc] | (in my case, it's Twinkle, but it could be any softphone) |
| 01:01.38 | [jmc] | what should I dial to enter the menu? |
| 01:01.54 | [jmc] | instead of directly calling other users or extensions |
| 01:02.14 | [TK]D-Fender | [jmc]: from whAT YOU'VE SET UP.... "S" <- |
| 01:02.52 | [jmc] | oh, yeah, thought about it |
| 01:02.55 | [jmc] | [Jun 13 03:03:33] NOTICE[7519]: chan_sip.c:14035 handle_request_invite: Call from '100' to extension 's' rejected because extension not found. |
| 01:03.02 | [jmc] | not working though :( |
| 01:03.16 | [jmc] | hmm |
| 01:03.20 | [jmc] | never mind |
| 01:03.30 | [jmc] | I've found what the problem is |
| 01:03.34 | *** join/#asterisk Aiatek (n=Alfio@190.94.58.219) |
| 01:03.45 | [jmc] | the phones are in a different context... |
| 01:04.28 | [TK]D-Fender | [jmc]: You are coming along just fine... |
| 01:04.37 | [jmc] | apologizes |
| 01:04.50 | [TK]D-Fender | [jmc]: I can think of quite a few users who could learn a LOT from you |
| 01:05.12 | [jmc] | from me? |
| 01:05.14 | [jmc] | :D |
| 01:05.27 | [jmc] | thank you |
| 01:05.38 | Aiatek | i think im gonna cry |
| 01:05.43 | Aiatek | :P |
| 01:05.45 | [TK]D-Fender | [jmc]: Yes... you ar doing this from scratch and your questions are the good kind and you are answering several of your own questions which means your eyes are actually open. |
| 01:05.58 | [TK]D-Fender | [jmc]: And there's one now :p |
| 01:06.27 | [jmc] | ^^ |
| 01:06.30 | [jmc] | one of? |
| 01:06.43 | [TK]D-Fender | [jmc]: [jmc] The people who need to learn what you have :) |
| 01:06.53 | [jmc] | oh :D |
| 01:06.54 | [jmc] | I see |
| 01:06.55 | [TK]D-Fender | [jmc]: On that note, I'm off for the night. Later all |
| 01:06.58 | [TK]D-Fender | [jmc]: Keep it up |
| 01:07.03 | [jmc] | :) |
| 01:07.09 | [jmc] | thanks for the support [TK]D-Fender |
| 01:07.12 | Aiatek | please dont go |
| 01:07.17 | [jmc] | I would have never got this far |
| 01:07.24 | [jmc] | night ;) |
| 01:07.49 | Aiatek | we gonna miss you |
| 01:08.03 | [jmc] | Aiatek: well, he's going to sleep |
| 01:08.09 | [jmc] | he's not going to die :D |
| 01:08.13 | Aiatek | hehehehehehe |
| 01:08.33 | Aiatek | bothering a little |
| 01:10.14 | [jmc] | Aiatek: what's the problem? |
| 01:10.31 | Aiatek | with what? |
| 01:10.39 | [jmc] | I'll try to answer if I can :) |
| 01:10.49 | [jmc] | asterisk I think |
| 01:10.57 | [jmc] | since you wanted him to stay |
| 01:10.57 | Aiatek | :) |
| 01:11.08 | Aiatek | hehehehehhehe |
| 01:11.16 | Aiatek | no prob |
| 01:11.47 | Aiatek | i think i can help you a little if you want |
| 01:12.03 | Aiatek | not as master TK but i can help |
| 01:12.39 | [jmc] | :D |
| 01:12.50 | [jmc] | not today, I'm going to bed |
| 01:13.01 | Aiatek | ok |
| 01:13.13 | [jmc] | but I'll keep it in mind :P |
| 01:13.32 | [jmc] | so I can leave that poor TK alone for 5 min :D |
| 01:13.40 | [jmc] | and ask you :D |
| 01:13.53 | Aiatek | ok no prob |
| 01:13.59 | [jmc] | but, again, had enough asterisk for today ^^ |
| 01:14.06 | [jmc] | good night guys :) |
| 01:14.18 | Aiatek | ok |
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| 01:44.58 | hesco | http://pastebin.ca/1458454 |
| 01:45.35 | hesco | this shows output of /usr/sbin/dahdi_cfg -vvvvvvvvvvvvv as well as *CLI> load dadhi-dummy |
| 01:45.56 | hesco | for some reason my dadhi* modules are not being built |
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| 01:47.47 | hesco | Aiatek: sorry about the distractions |
| 01:47.58 | hesco | in response to your earlier question: |
| 01:48.01 | hesco | http://pastebin.ca/1458454 |
| 01:48.04 | hesco | this shows output of /usr/sbin/dahdi_cfg -vvvvvvvvvvvvv as well as *CLI> load dadhi-dummy |
| 01:48.09 | hesco | for some reason my dadhi* modules are not being built |
| 01:48.35 | Aiatek | how many files do you have in ./etc/dahdi |
| 01:48.35 | Aiatek | ? |
| 01:49.12 | Aiatek | i saw that |
| 01:49.14 | hesco | 3 |
| 01:49.14 | Aiatek | how many files do you have in ./etc/dahdi |
| 01:49.18 | hesco | 3 |
| 01:49.20 | Aiatek | ok |
| 01:50.36 | hesco | all comments except for |
| 01:50.38 | hesco | loadzone = us |
| 01:50.38 | hesco | defaultzone=us |
| 01:50.50 | hesco | in /etc/dahdi/system.conf |
| 01:54.41 | hesco | http://pastebin.ca/1458464 |
| 01:55.11 | hesco | that shows a rebuild of the /usr/local/dahdi-tools-2.0.0 directory |
| 01:55.46 | hesco | but still no new dahdi*.so files in /usr/lib/asterisk/modules/ |
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| 04:30.08 | buttons840 | the book says that macros have stack overflow problems, and that nesting macros several layers deep may cause problems, was this fixed? or is it still a problem? |
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| 04:45.23 | buttons840 | lively bunch tonight :) |
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| 05:22.01 | shido6 | indeed |
| 05:33.26 | drmessano | is gonna take you by surprise and make you realize, amanda |
| 05:34.16 | drmessano | is gonna tell you right away and cant wait another day, amanda |
| 05:34.51 | drmessano | is gonna say it like a man and make you understand, amanda |
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| 06:23.28 | carrar | * drmessano is gonna have another drink, then another right away, amanda |
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| 06:33.29 | Pan3D | just got 1.6.2.0-beta2 up and running. |
| 06:33.33 | Pan3D | so far, so good |
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| 08:56.53 | BlackSlik | hello people i am on centOS |
| 08:57.21 | BlackSlik | i get a bash error when trying too run ./configure on dahdi-linux |
| 08:57.47 | BlackSlik | [root@nextdreamnet src]# cd /usr/src/dahdi-linux |
| 08:57.47 | BlackSlik | [root@nextdreamnet dahdi-linux]# clear |
| 08:57.48 | BlackSlik | [root@nextdreamnet dahdi-linux]# ./configure |
| 08:57.48 | BlackSlik | -bash: ./configure: No such file or directory |
| 08:57.48 | BlackSlik | [root@nextdreamnet dahdi-linux]# |
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| 09:19.30 | BlackSlik | hello people i am on centOS |
| 09:20.29 | wierdo | BlackSlik, try make |
| 09:21.39 | tzafrir_laptop | BlackSlik, no need for configure on dahdi-linux |
| 09:21.54 | tzafrir_laptop | (and this is not centos-specific...) |
| 09:22.52 | BlackSlik | k |
| 09:24.36 | BlackSlik | [root@nextdreamnet dahdi-linux]# make |
| 09:24.37 | BlackSlik | make -C drivers/dahdi/firmware firmware-loaders |
| 09:24.37 | BlackSlik | make[1]: Entering directory `/usr/src/dahdi-linux-2.2.0-rc5/drivers/dahdi/firmware' |
| 09:24.37 | BlackSlik | make[1]: Leaving directory `/usr/src/dahdi-linux-2.2.0-rc5/drivers/dahdi/firmware' |
| 09:24.37 | BlackSlik | You do not appear to have the sources for the 2.6.18-128.1.1.el5.028stab062.3 kernel installed. |
| 09:24.37 | BlackSlik | make: *** [modules] Error 1 |
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| 09:37.19 | tzafrir_laptop | BlackSlik, that's a strange kernel name. self-built? |
| 09:38.01 | tzafrir_laptop | What is the output of: uname -a |
| 09:39.37 | BlackSlik | Linux nextdreamnet 2.6.18-128.1.1.el5.028stab062.3 #1 SMP Sun May 10 18:54:51 MSD 2009 i686 i686 i386 GNU/Linux |
| 09:40.21 | *** join/#asterisk gunter (n=gunter@87.127.97.39) |
| 09:41.11 | BlackSlik | because i am new on asterisk |
| 09:41.17 | BlackSlik | i am using this to install |
| 09:41.18 | BlackSlik | http://www.howtoforge.com/asterisk_pbx_linux |
| 09:47.37 | tzafrir_laptop | howtoforge . often source of incomplete and wrong information that nobody bothers fixing |
| 09:48.53 | tzafrir_laptop | BlackSlik, one itreresting error they make there: |
| 09:49.08 | tzafrir_laptop | they give the following line as part of a sample config file: |
| 09:49.11 | tzafrir_laptop | [mark] (this is the username to use in the astman) |
| 09:49.32 | tzafrir_laptop | but forget to add ';' after the ']' |
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| 09:50.35 | tzafrir_laptop | BlackSlik, is this a centos kernel? |
| 09:50.44 | tzafrir_laptop | or your self-built one? |
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| 09:52.29 | tzafrir_laptop | I can't find such a kernel on a centos mirror |
| 09:52.57 | tzafrir_laptop | you need the kernel-devel from the same place you got the kernel package |
| 09:53.09 | tzafrir_laptop | Or, alternatively, the kernel source tree |
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| 10:50.52 | InfiniteInt | Hello, i am new to asterisk and want to know how i can get a connection to the normal phone system? I am living in Germany. |
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| 11:00.02 | BlackSlik | via ssh |
| 11:07.55 | gr0mit | InfiniteInt, pri or Bri? |
| 11:08.24 | InfiniteInt | Sorry, what is pri/bri? |
| 11:08.45 | gr0mit | what is yor original phone system? |
| 11:09.06 | InfiniteInt | a normal analog line |
| 11:09.17 | gr0mit | hm ok |
| 11:09.30 | gr0mit | can u not move it to isdn? |
| 11:09.57 | gr0mit | ah, so you don't have a phone system as in a pabx? |
| 11:10.16 | InfiniteInt | well sure with 5⬠more and i want to cut down my costs |
| 11:10.30 | gr0mit | in uk english a phone system often means pabx |
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| 11:10.54 | gr0mit | ok, well then u need an analogoe FXO card for your asterisk box |
| 11:11.06 | gr0mit | but if i were you i would convert to isdn |
| 11:11.11 | gr0mit | esp in germany |
| 11:11.26 | gr0mit | or port your number to voip and get rid of your analogue line |
| 11:11.34 | gr0mit | but then you will lose adsl |
| 11:11.53 | InfiniteInt | well is it possible to route everything thought TCP/IP |
| 11:12.01 | gr0mit | so in your case, i would spend 5 eur per month extra and get a better system |
| 11:13.39 | InfiniteInt | well i have a europe phone flatrate and why sould isdn be better? |
| 11:16.15 | InfiniteInt | my idea is to change to pay by minute and do every outgoing call via voip and get also incomming calls to my asterisk box |
| 11:18.07 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-694a77cffbba3599) |
| 11:18.13 | ariel_ | Morning |
| 11:27.59 | gr0mit | ok in that case if you have a flat rate, why not call divert to a voip number and have incoming and outgoing via voip? |
| 11:29.30 | gr0mit | anything to avoid an anlogue interface is good, analogue and voip is normally very problematic |
| 11:31.44 | InfiniteInt | as far as i get your point that sound exactly as what i want to do. What do you mean by devert to a voip number? |
| 11:33.36 | *** join/#asterisk puzzled (n=foobar@535335A1.cable.casema.nl) |
| 11:33.38 | ariel_ | anyone the difference between sudo -s or using sudo -i ?? |
| 11:33.40 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 11:34.10 | puzzled | hi |
| 11:34.22 | xnixan | hi, is there a 2 ports bri digium card? |
| 11:35.25 | gr0mit | ok say you are in berlin 030 xxxxx |
| 11:35.37 | gr0mit | you get a voip number from a provider |
| 11:35.43 | gr0mit | say sipgate.de |
| 11:35.52 | InfiniteInt | well, ok |
| 11:35.52 | puzzled | ariel_: iirc -i will use the shell specified for that user in /etc/passwd while -s while first look at the environment variable SHELL and then look in /etc/passwd |
| 11:36.11 | gr0mit | you call-forward your analoue line to your sipgate number also in 030 area |
| 11:36.19 | gr0mit | job done, sir |
| 11:36.50 | puzzled | 3. profit :) |
| 11:37.22 | ariel_ | puzzled: but it's basic setup is about the same just where to look. T/y |
| 11:37.24 | gr0mit | if you have to pay per min for diverted calls it might be a bit pricey |
| 11:37.52 | gr0mit | if your per min rate would cost mor than 5 euros/month, go isdn ! |
| 11:37.53 | InfiniteInt | i'll give this all a try. |
| 11:37.55 | ariel_ | xnixan: I only see a 4 port bri card on digium's web |
| 11:38.10 | gr0mit | ariel_, use a cologne hfc card |
| 11:38.13 | xnixan | ariel_, me too! |
| 11:38.21 | gr0mit | costs about £15 on ebay normally |
| 11:38.44 | gr0mit | sorry, xnixan |
| 11:39.03 | gr0mit | then use bristuffed asterisk |
| 11:39.07 | puzzled | gr0mit: does misdn for hfc cards actually work these days? |
| 11:39.16 | xnixan | gr0mit, for what?? |
| 11:39.17 | gr0mit | or chan_misdn |
| 11:39.22 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 11:39.36 | tzafrir_laptop | gr0mit, asterisk is the same. only thing needs patching is zaptel/dahdi |
| 11:39.37 | gr0mit | i only ever used bristuff when i had bri |
| 11:40.03 | gr0mit | but i cancelled bri a while back and ported numbers to voip |
| 11:40.07 | puzzled | ah ok. so did I but I didn't like the invasiveness of the patch |
| 11:40.14 | gr0mit | saved me !120 per quarter |
| 11:40.14 | xnixan | what about openvox compatibility with asterisk? |
| 11:40.29 | gr0mit | £120 per qtr i mean |
| 11:40.58 | puzzled | gr0mit: same here. last week I ditched the incumbent for Internet. Now I need to port my numbers to voip |
| 11:41.48 | tzafrir_laptop | xnixan, well, if you look at https://issues.asterisk.org/view.php?id=13897 , you'll see that the only cards activly and fully reported working are the openvox cards |
| 11:42.09 | puzzled | hi tzafrir_laptop |
| 11:42.29 | tzafrir_laptop | puzzled, hi |
| 11:42.37 | xnixan | tzafrir_laptop, thanks i will check it! |
| 11:42.37 | xnixan | ok guys, what would be the best solution for 2 isdn ports card to work with asterisk? |
| 11:43.41 | puzzled | xnixan: I don't know. imho bristuff is not the way to go but I don't know about the state of misdn these days. maybe try tzafrir's patches and test? |
| 11:44.14 | puzzled | xnixan: best bri cards I use when the client has $$$ is Eicon Diva Server with chan_capi. Works fabulously |
| 11:44.45 | *** join/#asterisk [jmc] (n=[jmc]@93-45-192-22.ip103.fastwebnet.it) |
| 11:50.28 | [jmc] | is here for your pleasure :P |
| 11:50.55 | gr0mit | here you have to port number before disconnecting |
| 11:51.06 | gr0mit | porting automatically disconnects |
| 11:56.11 | puzzled | gr0mit: same in .nl |
| 11:56.25 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:56.30 | gr0mit | <PROTECTED> |
| 11:56.45 | puzzled | The Hague |
| 11:56.57 | gr0mit | aah ok. i used to live in Drenthe |
| 11:57.31 | gr0mit | helemaal in 't Norden ;-) |
| 11:57.36 | puzzled | heh |
| 12:02.54 | puzzled | tzafrir_laptop: those patches are great. hope that Digium will merge them so we finally get decent hfc-s support in asterisk |
| 12:07.30 | *** join/#asterisk MauL^ (n=maul@88.249.176.176) |
| 12:07.48 | MauL^ | can you recommend me a startup reading with installation for 1.6.1? |
| 12:09.51 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 12:15.44 | [jmc] | ok, here's a new challenge today |
| 12:15.51 | [jmc] | I've got my simple IVR working for now |
| 12:16.02 | [jmc] | what I want to do next |
| 12:16.16 | [jmc] | is to use Festival to make it automatically speak the text I want |
| 12:16.29 | [jmc] | instead of recording and transcoding the messages |
| 12:16.46 | [jmc] | it seems to be "working" |
| 12:17.06 | [jmc] | in the sense that, when I call my IVR I can hear "Welcome to the Asterisk [...]" |
| 12:17.13 | [jmc] | but then it suddenly stops |
| 12:17.18 | [jmc] | it hangs up |
| 12:17.45 | [jmc] | the strange fact is that even MusicOnHold stops after the same little delay |
| 12:20.26 | [jmc] | or, maybe worse |
| 12:20.33 | [jmc] | on the console it goes fine |
| 12:20.42 | [jmc] | when i 'console dial' to my IVR |
| 12:20.54 | [jmc] | via Twinkle it receives the hangup |
| 12:23.24 | *** join/#asterisk madduck (n=madduck@debian/developer/madduck) |
| 12:44.39 | *** join/#asterisk DarkRift (n=dark@65.92.170.93) |
| 12:48.07 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 12:56.30 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:03.30 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 13:07.08 | *** join/#asterisk plq (n=plq@88.249.173.198) |
| 13:09.27 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 13:19.29 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 13:27.41 | *** join/#asterisk lazyb0y0 (n=nriley@68.202.147.152) |
| 13:27.56 | *** part/#asterisk devsys (i=devsys@free.dancing.bot.at.shellium.org) |
| 13:28.22 | *** join/#asterisk devsys (i=devsys@free.dancing.bot.at.shellium.org) |
| 13:29.21 | *** part/#asterisk lazyb0y0 (n=nriley@68.202.147.152) |
| 13:39.41 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 13:42.48 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 13:57.30 | *** join/#asterisk dacs (n=dacs@unaffiliated/dacs) |
| 13:58.19 | dacs | happy, back to Cali, sitting next to my * box and enjoying my SAT so far.:) |
| 14:01.21 | dacs | Good Morning all |
| 14:11.54 | dacs | i finaly got my x-lite working with my * , it register but i get this [Jun 12 13:45:56] NOTICE[9005]: chan_sip.c:17171 handle_request_invite: Call from '2000' to extension '2000' rejected because extension not found. |
| 14:14.36 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 14:15.48 | mltlnx | Good day. Has anyone using 1.6.1.1 been able to get additional parking lots configured (and working)? |
| 14:16.53 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:18.42 | carrar | Morning |
| 14:19.30 | carrar | dacs, what is the context that you have for your xlite peer? |
| 14:19.43 | carrar | (in sip.conf) |
| 14:23.51 | *** part/#asterisk InfiniteInt (n=Infinite@p5B0624DB.dip.t-dialin.net) |
| 14:24.03 | *** join/#asterisk Aiatek (n=Alfio@200.26.171.5) |
| 14:25.09 | Aiatek | hi everybody |
| 14:29.06 | juanIMP | hey Aiatek |
| 14:30.28 | Aiatek | hi <juanIMP> |
| 14:34.39 | mltlnx | Has anyone using 1.6.1.1 been able to get additional parking lots configured (and working)? |
| 14:42.50 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net) |
| 14:43.58 | dacs | carrar: http://pastebin.ca/1458986 |
| 14:46.29 | carrar | Where is your sip.conf? |
| 14:46.51 | dacs | in /etc/asterisk/ ? |
| 14:48.08 | [TK]D-Fender | carrar: :) |
| 14:48.14 | [TK]D-Fender | carrar: Who's on firt? |
| 14:48.17 | [TK]D-Fender | first* |
| 14:48.26 | carrar | whats on firt |
| 14:48.29 | carrar | first! |
| 14:48.38 | [TK]D-Fender | carrar: No, what's on third! |
| 14:48.56 | *** join/#asterisk korihor (n=korihor@200-71-161-128.genericrev.telcel.net.ve) |
| 14:49.54 | dacs | [2000] |
| 14:50.15 | ariel_ | I don't know is on 2nd |
| 14:50.40 | ariel_ | dacs: your context=sip for your sip.conf for the device 2000 |
| 14:56.50 | ariel_ | It's Saturday and a very nice day... |
| 14:57.10 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 14:57.13 | carrar | sure is! |
| 14:57.22 | Aiatek | here its very sunny |
| 14:57.32 | carrar | As is here |
| 14:57.59 | mltlnx | Would someone be willing (interested) in help me test multiple parking lots in 1.6.1? |
| 14:58.08 | mltlnx | I can get it to work. |
| 14:58.58 | ariel_ | it's very sunny here for now |
| 14:59.08 | ariel_ | I don't use 1.6.1 yet |
| 14:59.40 | ariel_ | is still mostly on 1.4.25 and has a few still on 1.2 |
| 15:00.02 | mltlnx | ariel_: thanks |
| 15:00.11 | ariel_ | but what are your errors |
| 15:00.14 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 15:00.34 | ariel_ | oh also I have a few on ABE C 2.03 and C 2.3.3 |
| 15:00.57 | mltlnx | Ah no errors. I simply can not get it to reliably park call on anything other then the default parkinglot |
| 15:01.32 | *** join/#asterisk Jarak01 (n=Jarak01@62.109.75.151) |
| 15:01.44 | [TK]D-Fender | mltlnx: you've been asking for days and haven't shown us anything |
| 15:01.52 | Jarak01 | hello community |
| 15:01.55 | ariel_ | can you post your settings on pastebin and what your using to get to the parking lot? Use pastebin please |
| 15:02.35 | *** join/#asterisk ingenius (n=alektro@190.230.72.176) |
| 15:02.35 | mltlnx | sure, give me a few moments |
| 15:04.11 | Jarak01 | i wont want to be unmannered but can i ask a question? |
| 15:04.15 | Jarak01 | i dont* |
| 15:04.27 | ariel_ | you can always ask |
| 15:04.29 | *** join/#asterisk puzzled (n=foobar@535335A1.cable.casema.nl) |
| 15:04.35 | Jarak01 | okay ariel, thank you |
| 15:04.36 | Jarak01 | i installed asterisknow yesterday but i just cant find it on my network list. i mean i cant open the asterisk web gui. when iam typing in httpd iam getting this msg: apr_sockaddr_info_get() failed for localhostxxxxx |
| 15:04.38 | *** join/#asterisk jmacz (n=jmacz@190.25.174.209) |
| 15:04.38 | ariel_ | asking is the only way to get a reply |
| 15:04.47 | Jarak01 | word |
| 15:05.04 | dacs | ariel_: i don't understand |
| 15:05.34 | Jarak01 | httpd: could not reliably determine the servers fully qualified domain name using 127.0.0.1 |
| 15:05.37 | ariel_ | dacs asterisk dials devices that are in it's context list |
| 15:05.47 | ariel_ | http://IPaddress |
| 15:05.57 | ariel_ | Jarak01: did you install from the iso? |
| 15:06.06 | ariel_ | 1.5 version |
| 15:06.09 | Jarak01 | y i did ariel |
| 15:06.11 | Jarak01 | y |
| 15:06.32 | Jarak01 | OH MY GOD |
| 15:06.33 | Jarak01 | ariel |
| 15:06.36 | Jarak01 | IT Works |
| 15:06.51 | Jarak01 | iam trying to install this asterisk for 1 week |
| 15:06.53 | Jarak01 | and now it works |
| 15:07.04 | ariel_ | typo |
| 15:07.07 | Jarak01 | :D |
| 15:07.10 | ariel_ | rofl |
| 15:07.19 | Jarak01 | iam so freaking happy |
| 15:07.33 | Jarak01 | okay how can i install remote? |
| 15:07.39 | Jarak01 | eg putty |
| 15:07.46 | ariel_ | remote ? |
| 15:07.48 | Jarak01 | should be installed rigt? |
| 15:07.51 | dacs | ariel_: can you explain to me this context list, its getting me so confused |
| 15:08.06 | Jarak01 | y for home purpose |
| 15:08.27 | ariel_ | putty is a windows program comes in a few versions |
| 15:08.32 | Jarak01 | y |
| 15:08.37 | madduck | any idea how to fix |
| 15:08.38 | madduck | app_meetme.c:774 build_conf: Unable to open pseudo device |
| 15:08.43 | madduck | on a pure SIP-asterisk? |
| 15:09.01 | Jarak01 | nevermind there are other questions;) thank you so much ariel |
| 15:09.04 | Jarak01 | for your time |
| 15:09.06 | Jarak01 | at least |
| 15:09.30 | ariel_ | dacs if your sip device should have a context it setup for. In asterisk extensions.conf you should have [thatcontext] area if you need to dial out via another context you need to include them in your sip devices defaults. Read the book |
| 15:09.48 | ariel_ | madduck: ztdummy |
| 15:10.11 | [TK]D-Fender | madduck: instal Zaptel/DAHDI |
| 15:10.53 | minimoi | hi everybody |
| 15:11.42 | minimoi | I'm going to configure an asterisk with realtime configuration |
| 15:12.01 | ariel_ | will stay away from realtime |
| 15:12.32 | minimoi | and i just want to know if it's possible to use sip trunks (register) with my sql database |
| 15:12.33 | minimoi | thx |
| 15:12.47 | ariel_ | yes |
| 15:13.00 | madduck | ariel_, [TK]D-Fender: so I need hardware drivers for conference rooms? |
| 15:13.04 | minimoi | and how please ? |
| 15:13.28 | [TK]D-Fender | madduck: It has always been a requirement |
| 15:14.11 | madduck | i didn't know that. thanks though. what a shame, since i'd really prefer not having to deal with kernel modules |
| 15:14.43 | dacs | ariel_: what other word would yo use! other than context please explain |
| 15:14.50 | ariel_ | madduck: it's needed for meetme due to timing and it's also needed for iax2 trunking. all other items it can do without it. |
| 15:16.01 | ariel_ | wow seems people don't do any start up reading.. |
| 15:16.20 | madduck | ariel_: if you pointed me to some docs, I would |
| 15:16.22 | carrar | You have to read the docs to use Asterisk? |
| 15:16.23 | carrar | WTF |
| 15:16.24 | ariel_ | dacs: you include context that do things if it's not included it will not do it. like groupings |
| 15:16.47 | ariel_ | ~book |
| 15:16.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:17.33 | ariel_ | back when I first started the normal response was read the code. So I guess there was some reading to do back then as well. |
| 15:18.43 | *** join/#asterisk Aiatek (n=Alfio@200.26.171.5) |
| 15:19.05 | madduck | ariel_: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-CHP-6-SECT-7.3 says nothing |
| 15:19.19 | madduck | (about zaptel) |
| 15:20.31 | ariel_ | madduck: Humm: I will need to read it then. But here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe |
| 15:21.06 | ariel_ | madduck: note this line: Please note: A Zaptel timer must be present for conferencing to work! See Asterisk timer |
| 15:22.09 | dacs | ariel_: the book assume i am using zaptel |
| 15:22.16 | madduck | ariel_: found it now. |
| 15:22.19 | madduck | thanks |
| 15:22.21 | [TK]D-Fender | madduck: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html/asterisk-book.html#asterisk-APP-B-109 |
| 15:22.33 | [TK]D-Fender | madduck: Nice disclaimer right there |
| 15:22.57 | [TK]D-Fender | madduck: So it is in the book itself |
| 15:24.35 | dacs | carrar: WTF is wrong with you, i am having a hard time understanding the word context and how to apply it to *. and i thought some here could help me understand it |
| 15:24.52 | [TK]D-Fender | dacs: context is a section in extensions.conf <------------ |
| 15:24.54 | carrar | dacs, I asked you a hour ago to see your sip.conf |
| 15:25.13 | [TK]D-Fender | carrar: Yes, but he showed you his dialplan! |
| 15:25.16 | carrar | heh |
| 15:25.24 | ariel_ | dacs, like I stated think of a context a section that you allot your sip phones to use. |
| 15:25.29 | [TK]D-Fender | carrar: Isn't that enough?!?! What does it take to make you happy?! |
| 15:25.39 | carrar | r00t! |
| 15:25.47 | *** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net) |
| 15:26.12 | ariel_ | which is when I said if he had setup is context correctly and he does not understand context |
| 15:26.54 | carrar | I have leave to go buy a new weed eater |
| 15:26.59 | carrar | better paste that ASAP |
| 15:27.39 | carrar | because if TK has to help you, you are really in for it |
| 15:27.55 | carrar | smirks |
| 15:28.04 | ariel_ | I am waiting for the wife to get back. So I can go do some work. She had to run to the store and spend my money so I have to take care of my little girl for now.... Disney channel is getting to me....argh |
| 15:28.40 | dacs | carrar: ariel_ http://pastebin.ca/1459014 |
| 15:28.45 | ariel_ | wow sponge bob now.... |
| 15:29.35 | dacs | carrar: no, thanks you, if he is the last persone on planet earth that knows * i don't need his help or opinion |
| 15:29.51 | carrar | woah |
| 15:30.32 | carrar | yeah, there is just no one in here that knows anything about asterisk |
| 15:30.36 | carrar | realyl sucks! |
| 15:30.38 | [TK]D-Fender | dacs: Here, instant learning for you : http://pastebin.ca/1459016 |
| 15:30.56 | carrar | bwhahaha |
| 15:31.05 | ariel_ | dacs: like I stated before, your sip is a context in your extensions. for it to have use of inbound you need to include=inbound just under the [sip] section |
| 15:31.32 | carrar | remember to enable: "fucking_contexts = YES" |
| 15:31.33 | jaytee | <dacs> carrar: no, thanks you, if he is the last persone on planet earth that knows * i don't need his help or opinion HAHAHAHAHAHAAAAAAAAAA!!!! |
| 15:31.33 | *** join/#asterisk HorizonXP (n=xitij@69-165-155-154.dsl.teksavvy.com) |
| 15:31.39 | [TK]D-Fender | dacs: a context is a section of extensions.conf which separates what a call is allowed to dial. |
| 15:31.58 | madduck | with ztdummy, the audio channel is choppy |
| 15:32.05 | *** part/#asterisk HorizonXP (n=xitij@69-165-155-154.dsl.teksavvy.com) |
| 15:32.11 | *** join/#asterisk HorizonXP (n=xitij@69-165-155-154.dsl.teksavvy.com) |
| 15:32.24 | [TK]D-Fender | madduck: Running a VM or some kind of more intense service like a file server on it? |
| 15:32.24 | HorizonXP | hey guys, i'm trying to set up my asterisk server with my voip provider |
| 15:32.33 | Aiatek | hello-world |
| 15:32.35 | madduck | it's not a very powerful machine. :( |
| 15:32.40 | [TK]D-Fender | madduck: How bad? |
| 15:32.46 | Aiatek | not hello world |
| 15:32.48 | madduck | pretty bad |
| 15:32.49 | [TK]D-Fender | madduck: And what kind of load on it? |
| 15:32.52 | carrar | heh yeah TK, wasn't even gonna comment on the space in his filename, figured he would see that error on the console |
| 15:32.52 | HorizonXP | it looks like it's registered with their servers, and i got Ekiga on my desktop to connect to Asterisk's echo test, which works |
| 15:32.55 | [TK]D-Fender | madduck: Details... |
| 15:33.06 | HorizonXP | but what do I need to do to allow Ekiga to dial out local numbers? |
| 15:33.08 | madduck | [TK]D-Fender: it's a soekris net5501, load it low though |
| 15:33.15 | jaytee | 286 12mhz with a 287 FPU coprocesser! |
| 15:33.18 | [TK]D-Fender | carrar: For the record dacs here is technically legally blind. |
| 15:33.40 | madduck | [TK]D-Fender: AMD Geode 500 MHz |
| 15:33.41 | dacs | ariel_: carrar jaytee , ok i am a 5ESS wireless guy ok, there is documents that can fill the room for translation and all this stuff , but i don't give people crap about how i am so good with the 5E |
| 15:33.48 | [TK]D-Fender | madduck: Ah... nifty little box... I do get the impression you should be able to get a few calls in a meetme on it though |
| 15:33.56 | [TK]D-Fender | madduck: check your kernel timer <----- |
| 15:34.04 | [TK]D-Fender | madduck: make sure it was compiled for 1000hz |
| 15:34.12 | Aiatek | nifty?? |
| 15:34.19 | dacs | or how good i am working on tellabs 5500 |
| 15:34.22 | madduck | # CONFIG_HZ_1000 is not set |
| 15:34.22 | madduck | CONFIG_HZ=250 |
| 15:34.23 | tzafrir_laptop | [TK]D-Fender, why? |
| 15:34.24 | madduck | :( |
| 15:34.50 | tzafrir_laptop | dahdi_dummy should use hi_res times on kernel 2.6.26 |
| 15:35.05 | tzafrir_laptop | ztdummy as well |
| 15:35.05 | dacs | or the fact i work with packs daily that cost over $50K a pack |
| 15:35.08 | [TK]D-Fender | dacs: madduck And what kernel are you running on it? |
| 15:35.16 | madduck | 2.6.26-2-486 |
| 15:35.25 | dacs | we all learn, and we all have hard time sometime filter shit |
| 15:35.32 | tzafrir_laptop | btw: can that cpu work with -686? |
| 15:35.38 | [TK]D-Fender | tzafrir_laptop: Yup |
| 15:35.40 | madduck | no |
| 15:35.43 | [TK]D-Fender | tzafrir_laptop: AMD Geode |
| 15:35.47 | madduck | never worked for me. |
| 15:35.58 | dacs | but not to sit and insult people because i know how to setup a fucking pbx |
| 15:36.04 | [TK]D-Fender | madduck: Sorry, 586 IIRC |
| 15:36.07 | Aiatek | ~ nifty |
| 15:36.07 | infobot | I feel reniftified! |
| 15:36.10 | dacs | anyways i am past that old fart |
| 15:36.25 | [TK]D-Fender | dacs: I've handed you a rather complete description in there. Anything left you are unclear about following it? |
| 15:36.56 | tzafrir_laptop | madduck, isn't 'cmov' on flags in /proc/cpuinfo an indication it should work with -686? |
| 15:36.59 | Jarak01 | hey there, how do i quit from mysql |
| 15:37.08 | [TK]D-Fender | dacs: Including likely future errors you'd run into on it. |
| 15:37.19 | madduck | tzafrir_laptop: all i know it didn't boot |
| 15:37.29 | carrar | Jarak01, rm mysql, install postgresql |
| 15:37.46 | Jarak01 | iam into mysql -r |
| 15:37.51 | Jarak01 | iam in mysql |
| 15:37.53 | ariel_ | funny thing is that I am a old fart.. |
| 15:37.54 | Jarak01 | but i cant quit |
| 15:37.58 | Jarak01 | i restarted now |
| 15:37.59 | carrar | <PROTECTED> |
| 15:38.04 | Jarak01 | ahh ke |
| 15:38.20 | Jarak01 | thank you carrar |
| 15:38.37 | madduck | yeah, ztdummy makes asterisk prompts useless |
| 15:38.40 | HorizonXP | any ideas on why Ekiga is saying "User not found" when i try to dial a landline number? |
| 15:38.52 | dacs | ariel_: but you respect your self, i never saw you insult anyone |
| 15:39.00 | HorizonXP | i.e. sip: 1235551234@192.168.10.100 |
| 15:39.01 | madduck | calls wirk fine |
| 15:39.01 | [TK]D-Fender | HorizonXP: What does * say? |
| 15:39.02 | madduck | work |
| 15:39.10 | HorizonXP | where that ip is my asterisk |
| 15:39.18 | HorizonXP | [TK]D-Fender: I dunno, lemme check its logs |
| 15:39.19 | [TK]D-Fender | HorizonXP: And don't put @IP" in your dial line. Only the number to dial |
| 15:39.21 | tzafrir_laptop | hmm.... on my alix system indeed zttest gives 400% |
| 15:39.32 | tzafrir_laptop | madduck, ==^ :-( |
| 15:39.32 | [TK]D-Fender | HorizonXP: No logs. Check live CLI with SIP DEBUG enabled only |
| 15:39.41 | dacs | Jarak01: what linux flav you use? |
| 15:39.53 | HorizonXP | [TK]D-Fender: it's running on an ubuntu router with an init.d script |
| 15:39.55 | [TK]D-Fender | tzafrir_laptop: 400%? Man you go all out! |
| 15:39.59 | HorizonXP | should i kill it and start manually? |
| 15:40.06 | [TK]D-Fender | HorizonXP: OS doesn't matter for this |
| 15:40.24 | dacs | try ctrl+alt+F1 or F2 or F# and do ps -ef | grep mysql and kill -9 PID |
| 15:40.52 | HorizonXP | [TK]D-Fender: no i know, i just wanted to convey that it's running as a daemon right now. should i kill that and start it manually from the console? |
| 15:41.02 | [TK]D-Fender | HorizonXP: * should be running, you should be connected to the CLI and watching your call |
| 15:41.16 | ariel_ | asterisk -r |
| 15:41.19 | [TK]D-Fender | HorizonXP: Go connect to CLI. It *should* be running as a daemon |
| 15:41.37 | [TK]D-Fender | HorizonXP: ariel_ has just handed you a reminder for something your should know by heart by now |
| 15:41.44 | HorizonXP | ariel_: thanks! |
| 15:41.53 | HorizonXP | [TK]D-Fender: yeah I saw that |
| 15:41.59 | HorizonXP | [TK]D-Fender: I just installed asterisk today |
| 15:42.16 | [TK]D-Fender | HorizonXP: Ok, then we'll expect that to have sunk in by tomorrow ;) |
| 15:42.45 | HorizonXP | it rejected it because the extension was not found |
| 15:42.47 | madduck | zttest reports -200 for me. :( |
| 15:44.22 | *** join/#asterisk rjune_ (n=rjune@38.103.117.250) |
| 15:44.39 | [TK]D-Fender | HorizonXP: Then you need to go correct your dialplan/ |
| 15:45.10 | madduck | no meeting rooms for me. :( |
| 15:45.20 | HorizonXP | [TK]D-Fender: seems like it, but these are the settings provided by my provider |
| 15:45.29 | [TK]D-Fender | madduck: Go grab app_conference. It doesn't require zaptel. |
| 15:45.41 | [TK]D-Fender | HorizonXP: this has nothing to do with your provider anymore |
| 15:45.55 | [TK]D-Fender | HorizonXP: the only thing they tell you is the sip.conf settings. the dialplan is 100% YOUR job |
| 15:46.07 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-206.phlapa.fios.verizon.net) |
| 15:46.22 | HorizonXP | [TK]D-Fender: yeah, but they gave me that too. I put it on pastebin: http://pastebin.com/d21f794fc |
| 15:46.36 | madduck | [TK]D-Fender: no debian package provides that file. :( |
| 15:46.37 | [TK]D-Fender | HorizonXP: A car radio manufacturer will give you instructions on how to connect it to your amp... they will not give you an instruction book on how to drive your cal |
| 15:46.39 | [TK]D-Fender | car* |
| 15:46.50 | dacs | ariel_: take a look http://pastebin.ca/1459029 |
| 15:46.55 | [TK]D-Fender | maddthere are no packages of any kind. You always have to compile this 3rd party app |
| 15:47.35 | [TK]D-Fender | dacs: And I did tell you 3 times in there that I was pretty sure that file you're referring to does not exist |
| 15:49.17 | Jarak01 | carrar , ariel? |
| 15:49.31 | HorizonXP | [TK]D-Fender: that's all well and good... do you have a link i could read to set up a dialplan, so i can compare it to my existing one? |
| 15:49.54 | Jarak01 | i changed my sql password of asterisk but no i get FATAL ERROR DB ERror :( |
| 15:49.56 | ariel_ | dacs: seems your sound files are either not installed or not in correct location or rights are off them. check your /var/lib/asterisk/sound for files |
| 15:50.07 | ariel_ | Jarak01: yes |
| 15:50.13 | [TK]D-Fender | dacs: And Aiatek also said as much |
| 15:50.16 | Jarak01 | why is that? |
| 15:50.29 | Jarak01 | do i have to restart mysql if yes how? |
| 15:50.32 | ariel_ | Jarak01: I don't use real time... |
| 15:50.41 | [TK]D-Fender | HorizonXP: The BOOK... |
| 15:50.43 | [TK]D-Fender | ~book |
| 15:50.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:50.59 | HorizonXP | well then |
| 15:51.00 | [TK]D-Fender | HorizonXP: the dialplan is 95% of Asterisk. You need to master this. |
| 15:51.00 | Jarak01 | how you mean real time? |
| 15:51.02 | HorizonXP | LOL thanks |
| 15:51.24 | ariel_ | ok what is pending in that dial plan? |
| 15:51.35 | dacs | ariel_: all sound files are there |
| 15:51.48 | [TK]D-Fender | dacs: the one you are referencing does not exist |
| 15:51.49 | ariel_ | touch them |
| 15:52.54 | dacs | ariel_: you mean to do touch /var/lib/asterisk/sounds/*.* |
| 15:53.12 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 15:53.40 | *** join/#asterisk davevg-btwtech (n=davevg__@nj-67-76-177-147.sta.embarqhsd.net) |
| 15:53.45 | [TK]D-Fender | ariel_: What would changing a datestamp do? |
| 15:54.00 | [TK]D-Fender | ariel_: Especially for a file that does not exist? |
| 15:54.00 | *** join/#asterisk AsteriskDom (n=Alfio@200.26.171.5) |
| 15:54.08 | Jarak01 | SET PASSWORD FOR freepbx@localhost=PASSWORD('new password'); do i have to use the signs: ()` |
| 15:54.13 | Jarak01 | in sql |
| 15:54.38 | ariel_ | I was actually not finished, I actually was talking about check to make sure file was there ls name |
| 15:57.08 | *** join/#asterisk pulpster (n=pulpster@p16.eregie.pub.ro) |
| 15:57.36 | pulpster | hello |
| 15:58.53 | pulpster | I have the following problem: I need to run some PHP script when I call a number of my Win32 Asterisk PBX. All set ok, except that I cannot find out a way of installing the PHP AGI module on Win32 Asterisk. How Do i do that ? |
| 15:58.56 | dacs | ariel_: files are there |
| 15:59.32 | [TK]D-Fender | dacs: Not the one * looking for. |
| 16:00.12 | kaldemar | dacs: find /var/lib/asterisk/sounds/ -name 'Hello*' |
| 16:00.14 | [TK]D-Fender | pulpster: AsteriskWIN32 is not supported here |
| 16:00.30 | dacs | kaldemar: it is there |
| 16:00.38 | [TK]D-Fender | dacs: Show us |
| 16:00.45 | kaldemar | dacs: not according to asterisk. |
| 16:01.11 | Aiatek | use beep |
| 16:01.20 | pulpster | do you have an idea where I could find some answers for my problem ? |
| 16:01.30 | Aiatek | dont complicate |
| 16:01.35 | [TK]D-Fender | pulpster: Can you call an AGI in any other language? |
| 16:01.58 | [TK]D-Fender | Aiatek: Complicate? How complicated is it to show us that a file you think exists actualyl does? |
| 16:02.06 | kaldemar | what is the php agi module? |
| 16:02.11 | pulpster | yes, Perl but I want PHP specificaly - becuase it's related to the rest of my project |
| 16:02.21 | [TK]D-Fender | kaldemar: 3rd party PHP lib. |
| 16:02.54 | [TK]D-Fender | kaldemar: Well go to thier site and go download it. Its just a piece of PHP code... |
| 16:03.02 | [TK]D-Fender | pulpster: Well go to thier site and go download it. Its just a piece of PHP code... |
| 16:03.21 | kaldemar | [TK]D-Fender: roger. and hell no to the latter. |
| 16:03.30 | [TK]D-Fender | pulpster: The term "install" doesn't really apply. |
| 16:03.32 | dacs | kaldemar: how do i check |
| 16:03.50 | kaldemar | dacs: check what? |
| 16:03.53 | [TK]D-Fender | dacs: [12:00]<kaldemar>dacs: find /var/lib/asterisk/sounds/ -name 'Hello*' |
| 16:04.00 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 16:04.20 | [TK]D-Fender | dacs: He already gave you that answer |
| 16:04.46 | dacs | kaldemar: nm |
| 16:10.52 | dacs | kaldemar: how do i check why my sound are not working |
| 16:11.11 | ariel_ | dacs your post said it's looking for Hello World,, the actual file name should be hello-world |
| 16:11.12 | [TK]D-Fender | dacs: the file does not exist. That is the problem. |
| 16:11.46 | ariel_ | dacs spelling and case lettering has to be correct |
| 16:12.16 | *** join/#asterisk IgorG (n=igorg@host-92-124-179-77.pppoe.omsknet.ru) |
| 16:12.24 | dacs | ariel_: ok will check that right now |
| 16:14.10 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 16:14.19 | ariel_ | nice wife is back from publix's. We now have snacks back in our house......junk food here I come.... |
| 16:14.49 | rjune_ | woot |
| 16:15.42 | Jarak01 | hehe |
| 16:16.02 | Jarak01 | my wife is just buying greenstuff |
| 16:16.16 | Jarak01 | so no junk food for me until i go buy maself |
| 16:16.30 | Jarak01 | healthy stuff |
| 16:16.36 | Jarak01 | = greenstuff ;) |
| 16:16.38 | dacs | ariel_: are you in Fl? |
| 16:16.47 | ariel_ | dacs: yes |
| 16:17.00 | ariel_ | oh she even got some bacon....yummy |
| 16:17.05 | dacs | ariel_: i just came back from there |
| 16:17.42 | dacs | ariel_: i was in alamonte spring..@ Lucent facility for AnyPath Training |
| 16:19.02 | Jarak01 | ariel if you wont stop this, i have to leave the chat and go buy me some junkfood |
| 16:19.07 | ariel_ | dacs I am way south.....almost in the keys...I am close enough to see margarita ville |
| 16:19.22 | ariel_ | rofl |
| 16:19.29 | dacs | ariel_: lol |
| 16:19.36 | [TK]D-Fender | Jarak01: If thats enough to push you over, you're doomed to fail anyway.... Real test is physical exposure |
| 16:19.56 | dacs | ariel_: that was not the case for hello-world file |
| 16:23.35 | Jarak01 | one more question for yo: when iam klicking on panel, in freepbx, i recieve the errormsg: "client/server version mismatch" |
| 16:23.45 | Jarak01 | does someone has an idea why? |
| 16:23.58 | Jarak01 | actually i think i know why |
| 16:24.02 | Jarak01 | but how to fix |
| 16:26.37 | [TK]D-Fender | Jarak01: ... |
| 16:26.39 | [TK]D-Fender | ~freepbx |
| 16:26.39 | 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 |
| 16:26.42 | [TK]D-Fender | ^^^^^^ |
| 16:27.13 | rjune_ | [TK]D-Fender, I've found you to be immensely helpful to me. |
| 16:27.30 | rjune_ | though I tend to ask generic questions |
| 16:29.23 | [TK]D-Fender | rjune_: Keep an open mind, don't give up and as you ask clearer an more specific questions, the answers will come. |
| 16:29.50 | [TK]D-Fender | rjune_: Give up or ignore advice and you'll spend forever on stupid stuff. |
| 16:30.06 | ariel_ | dacs: next thing is are the rights correct for access to the voice files? |
| 16:30.24 | [TK]D-Fender | rjune_: Great examples of that here all the time. You can't even hand the answer to some people on a silver platter. |
| 16:30.29 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 16:30.52 | [TK]D-Fender | ariel_: Still a few dozen more hoops for you to jump through :) |
| 16:36.57 | dacs | ariel_: i did chmod +x all of them |
| 16:40.48 | *** join/#asterisk ivanvujisic (n=ivanvuji@91.148.80.48) |
| 16:41.20 | *** join/#asterisk xpot-mobile (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:42.39 | ivanvujisic | I'm trying to hide my set up made in extensions.conf, so no customer can't read/copy it, any idea? |
| 16:43.39 | [TK]D-Fender | ivanvujisic: Don't give them a login account to your server. |
| 16:43.44 | [TK]D-Fender | ivanvujisic: Or physical access. |
| 16:43.56 | [TK]D-Fender | ivanvujisic: Because physical access = can do whatever they want |
| 16:44.13 | [TK]D-Fender | ivanvujisic: This is a Linux question anyway, not an * one |
| 16:44.20 | ivanvujisic | yeap, I know it, but customer wont pay if I dont give them root privileges |
| 16:44.38 | ivanvujisic | sorry, I know its common Linux question |
| 16:44.42 | [TK]D-Fender | ivanvujisic: If they have root you can't stop them. Seriously.... |
| 16:44.52 | [TK]D-Fender | ivanvujisic: You can't stop root <- |
| 16:45.25 | rjune_ | [TK]D-Fender, You misunderstand, I'm using freepbx, but I ask generally how something works, not how does freepbx do it |
| 16:45.51 | [TK]D-Fender | rjune_: Oh for THOSE questions... yeah... keep the GUI part out of it and yeah we don't mind. |
| 16:46.30 | [TK]D-Fender | rjune_: but if you're a "how do I set my trunk?!?!" twit.... well.. thats another story |
| 16:47.11 | rjune_ | I've asked bout DID information on POTS Lines |
| 16:48.29 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:48.29 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:49.11 | [TK]D-Fender | rjune_: Yeah, thats largely in "pink unicorn" territory. |
| 16:49.19 | rjune_ | [TK]D-Fender, is it possible to forward caller id information from an incoming line back to an outgoing line? |
| 16:50.00 | rjune_ | aka, I worked out how to make an external number an extension, but when somebody dials in, then dials that extension, it always says private caller |
| 16:50.18 | [TK]D-Fender | rjune_: You don't set callerid on POTS lines. Analog = dumb |
| 16:50.36 | rjune_ | Outgoing dialing is via sip trung |
| 16:50.38 | rjune_ | trunk |
| 16:50.52 | [TK]D-Fender | rjune_: Most of what you asked IS GUI config... so not covered here |
| 16:51.21 | [TK]D-Fender | rjune_: As for setting callerID on your SIP provider... they'd have to allow you as well |
| 16:51.32 | rjune_ | I already know they allow it |
| 16:52.13 | rjune_ | though I fail to see how "is it possible to write a dialplan that pulls the caller id from an incoming call and pushes it to an external extension" is GUI config |
| 16:53.30 | [TK]D-Fender | rjune_: You don't pull callerid. You simply have it or you don't |
| 16:55.18 | ivanvujisic | rjune_: Just turn on - sip set debug on |
| 16:57.12 | mltlnx | Has anyone tried making multiple parking lots in 1.6? |
| 16:57.16 | [TK]D-Fender | rjune_: And I had misread onew word out from your previous question... |
| 16:57.40 | [TK]D-Fender | mltlnx: Are you any closer to showing us your failed attempts and configs like we asked you for repeatedly over the last few hours? |
| 16:58.50 | mltlnx | [TK]D-Fender: ah, I got side tracked reading the change log....sorry coming soon. |
| 16:59.34 | *** join/#asterisk Aiatek (n=minombre@190.94.57.227) |
| 17:02.04 | rjune_ | [TK]D-Fender, perhaps I'm not asking the proper question then. CID information exists on incoming call. I want to relay the CID to an external(cellphone) extension, so that when someone calls in and dials that extension, the proper caller id information is displayed instead of private |
| 17:02.33 | [TK]D-Fender | rjune_: go look at the outbound call and see what is actually happening. |
| 17:03.21 | mltlnx | [TK]D-Fender: http://pastebin.com/m35d49c16 |
| 17:04.50 | rjune_ | [TK]D-Fender, I'll have a log shortly |
| 17:05.26 | [TK]D-Fender | rjune_: No logs, only live CLI with SIP debug & max verbose. |
| 17:05.37 | [TK]D-Fender | mltlnx: Where is your failed attempt |
| 17:10.00 | mltlnx | [TK]D-Fender: Shouldn't extension 800 be available to me from with the [testing] context? I have include => testpark |
| 17:10.23 | [TK]D-Fender | mltlnx: You've been asked several times to show us the PROBLEM. |
| 17:10.33 | [TK]D-Fender | mltlnx: Where is the FAILED call? |
| 17:11.11 | mltlnx | Call from '100' to extension '800' rejected because extension not found. |
| 17:12.12 | mltlnx | 100 is in the [testing] context, testpark is the declared context in parkinglot_test. |
| 17:12.13 | [TK]D-Fender | mltlnx: Look in the sample config and you will not see "parkext" in a custom lot |
| 17:12.32 | [TK]D-Fender | mltlnx: And you are not showing enough of your config to see if you followed the OTHER directions the sample provides you either |
| 17:13.08 | mltlnx | Ok let me look again....Thank you |
| 17:13.15 | [TK]D-Fender | mltlnx: I might take it at face value that what you showed me was your ENTIRE features.conf for which I would tell you "Where is [general] section and all the other normal stuff?" |
| 17:27.44 | mltlnx | [TK]D-Fender: Sorry for being so slow with this....http://pastebin.ca/1459107 -- That is my complete feature.conf |
| 17:29.56 | mltlnx | [TK]D-Fender: I see what you mean about the parkext in a custom lot. This should either be set in the dialplan or in the general section(as a deafult for all?) |
| 17:32.53 | [TK]D-Fender | mltlnx: that value is GENERAL only, on per lot. |
| 17:33.16 | [TK]D-Fender | not* |
| 17:33.48 | mltlnx | [TK]D-Fender: I see |
| 17:34.16 | madduck | hm, i am getting really funky noises in the speaker when i make a call through my server. can you try to dial echo@madduck.net and see if they exist outside too? |
| 17:36.22 | *** join/#asterisk AsteriskDom (n=minombre@190.94.57.227) |
| 17:40.45 | *** join/#asterisk mechbangirc (n=mech@mbl-65-186-216.dsl.net.pk) |
| 17:43.30 | mltlnx | [TK]D-Fender: I cant seem to get this working....What else can I provide you? Do you know of any sample examples multi parking lots? |
| 17:44.00 | *** join/#asterisk sikanrong (n=sikanron@211.105.219.87.dynamic.jazztel.es) |
| 17:44.08 | [TK]D-Fender | mltlnx: You set a single exten to aprk calls under general. This applies to everyone. |
| 17:44.35 | [TK]D-Fender | mltlnx: the exten you use to park the call has no impact on the LOTS that it will use. That will depend on your following the OTHER instructions. |
| 17:45.05 | sikanrong | anybody have any experience with voovox? |
| 17:45.19 | *** join/#asterisk Failrar (n=Failrar@tunnel1088.ipv6.xs4all.nl) |
| 17:45.38 | sikanrong | trying to get incoming calls, have my softphone registered (and it can even make outgoing calls), but the incoming calls don't seem to work, softphone never rings, all i get is congestion |
| 17:46.04 | sikanrong | I have everything NATted right too |
| 17:46.09 | sikanrong | (I think) |
| 17:46.26 | mltlnx | [TK]D-Fender: Are the other instructions: to set the channel var parkinglot? |
| 17:47.10 | [TK]D-Fender | mltlnx: its tells you there is a var and what its for. Go follow the instructions |
| 17:47.28 | *** join/#asterisk rob0 (n=rob0@cardinal.lizella.net) |
| 17:48.53 | mechbangirc | hi no matter what i do, i am not able to query mysql database through my dialplan |
| 17:49.03 | mltlnx | [TK]D-Fender: OK, thanks. Trying now. |
| 17:50.20 | mechbangirc | i get a warning app_addon_sql_mysql.c:311 aMYSQL_query: aMYSQL_query: mysql_query failed. saying sql syntax error |
| 17:50.23 | mechbangirc | any idea |
| 17:50.27 | *** join/#asterisk louben (n=louben@212-70-216-131.ath.static.tee.gr) |
| 17:54.28 | madduck | it was a client problem. doh. |
| 17:55.01 | tompaw | What's the best way to use mysql backend in my dialplan? DBQuery? |
| 17:55.06 | tompaw | (1.6) |
| 17:55.29 | mechbangirc | tompaw: i want to know also for 1.4 |
| 17:56.05 | tompaw | mechbangirc: isn't the addon dedicated to storing cdrs only? |
| 17:56.48 | [TK]D-Fender | tompaw: No |
| 17:57.13 | mechbangirc | tompaw: you can add column to that table and set the values in that table like: Set(CDR(new_field)="value") |
| 17:57.33 | tompaw | ok |
| 17:57.58 | tompaw | [TK]D-Fender: so if I want my dialplan to use mysql, but NOT for cdrs, is DBQuery the way to go? |
| 17:58.04 | mechbangirc | i want to know how can i get data from other tables from mysql? |
| 17:58.36 | [TK]D-Fender | tompaw: func_odbc , MySQL() from addon's, AGI, take your pick |
| 17:58.43 | [TK]D-Fender | tompaw: depends what else you need to do around it |
| 17:59.01 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 17:59.34 | tompaw | [TK]D-Fender: just a simple set of crud operations. I need my dialplan to handle 100 "virtual" channels. Each of them is the same physical connection, but with different "pin" dtmf'ed during the call. |
| 17:59.58 | tompaw | [TK]D-Fender: so I need mysql to store a) those pins (dtmf sequences) and more importantly b) info about which "channel" is busy |
| 18:00.55 | mechbangirc | [TK]D-Fender: do you put forward slashes after each word in query command in MYSQL()? |
| 18:03.34 | HorizonXP | i'm having trouble configuring asterisk to receive incoming calls |
| 18:03.50 | HorizonXP | i dial my DID from another line, it rings twice, and then I hear nothing |
| 18:05.10 | mltlnx | [TK]D-Fender: I have tried what you suggested. Here are my results: http://pastebin.ca/1459129 |
| 18:06.18 | HorizonXP | how can I tell whether * is actually receiving a call? |
| 18:06.51 | mechbangirc | HorizonXP: it is shown on CLI |
| 18:06.54 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 18:06.54 | *** join/#asterisk lftsy (n=lftsy@ratatouille.leurent.eu) [NETSPLIT VICTIM] |
| 18:06.54 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) [NETSPLIT VICTIM] |
| 18:06.54 | *** join/#asterisk eppigy (n=Dave@216-139-245-58.aus.us.siteprotect.com) [NETSPLIT VICTIM] |
| 18:06.54 | *** join/#asterisk e0n` (n=xorl@sys-adm.org) |
| 18:06.54 | *** join/#asterisk Takapa (i=vegard@junior.svanberg.no) |
| 18:06.54 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) [NETSPLIT VICTIM] |
| 18:06.54 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 18:06.54 | *** mode/#asterisk [+o angler] by irc.freenode.net |
| 18:07.13 | HorizonXP | mechbangirc: ok, so if I see nothing on the CLI, then * isn't receiving any kind of call at all to my DID? |
| 18:07.41 | mechbangirc | HorizonXP: if it is not on CLI then it is not there |
| 18:08.02 | HorizonXP | mechbangirc: ok, so then it's a different configuration issue then |
| 18:08.07 | mltlnx | .......loosing my mind |
| 18:09.38 | mechbangirc | HorizonXP: sure it could be networking problem or port issue |
| 18:11.08 | [TK]D-Fender | HorizonXP: Could mean you aren't watching SIP DEBUG for it |
| 18:12.34 | mechbangirc | [TK]D-Fender: he does not see anything on CLI. |
| 18:12.54 | [TK]D-Fender | mechbangirc: And * CLI does not show SIP DEBUG until you TELL IT TO |
| 18:13.13 | [TK]D-Fender | mechbangirc: And without it who cares. |
| 18:13.28 | mechbangirc | [TK]D-Fender: yes however even if you do not have a context to receive calls you see incoming calls |
| 18:14.31 | [TK]D-Fender | mechbangirc: No, you don't |
| 18:15.20 | [TK]D-Fender | mechbangirc: If the call is REFUSED you will not see anything without SIP debug |
| 18:15.25 | mechbangirc | [TK]D-Fender: you see messages like call rejected/dropped because of bla bla |
| 18:15.38 | [TK]D-Fender | mechbangirc: That also depends on verbose |
| 18:15.55 | mechbangirc | [TK]D-Fender: if you have sip debugged set then you get packets not verbose messages |
| 18:16.08 | [TK]D-Fender | mechbangirc: you can get BOTH |
| 18:16.21 | [TK]D-Fender | mechbangirc: they are completely separate modes of debugging |
| 18:16.21 | mechbangirc | [TK]D-Fender: true i always run asterisk with 10+ verbosity |
| 18:16.31 | mechbangirc | [TK]D-Fender: i agree |
| 18:18.28 | *** part/#asterisk mechbangirc (n=mech@mbl-65-186-216.dsl.net.pk) |
| 18:20.45 | mltlnx | [TK]D-Fender: Hey there, you must be bored of this issue, But if you have any suggestions or can refer me to sample that would be great. |
| 18:21.35 | [TK]D-Fender | mltlnx: You still aren't showing me a FAILED CALL. And I don't see your corrected configs. You asked about the channel variable YOU HAVE TO SET. I did not see you do this. |
| 18:22.01 | [TK]D-Fender | mltlnx: You need to wake the hell up, stop whining and show what the hell you are DOING. |
| 18:23.09 | HorizonXP | ok, so i think it was my firewall. i switched that off for now. i started the CLI with vvv, so now I see * is receiving the call |
| 18:23.13 | mltlnx | I did in a previous post: http://pastebin.ca/1459129 Posted there is the context from extensions.conf as well as the failed call. |
| 18:23.37 | mltlnx | Is this not what you are asking for? |
| 18:23.48 | HorizonXP | but it's going crazy trying to dial out somewhere else when it receives the call, even though all it's supposed to do is Answer() |
| 18:24.23 | [TK]D-Fender | mltlnx: do not use the CHANNEL FUNCTION. this is a VARIABLE |
| 18:31.20 | mltlnx | [TK]D-Fender: So I used Set(parkinglot=parkinglot_test) with no luck. Is this what you had in mind? |
| 18:32.34 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:32.44 | [TK]D-Fender | mltlnx: thats a mltlnx You are doing it wrong. Read the samples again |
| 18:33.36 | mltlnx | Samples? What samples? From features.conf? |
| 18:33.38 | [TK]D-Fender | mltlnx: itmlytKeep playing with this. |
| 18:33.51 | [TK]D-Fender | I'm off for a while |
| 18:34.08 | mltlnx | ok thanks |
| 18:34.37 | eppigy | good day |
| 18:34.39 | eppigy | I am dave |
| 18:35.05 | HorizonXP | * sees the call, but it's not answering it. somethings up with my dialplan, but i don't know what |
| 18:35.30 | *** join/#asterisk NateHB (n=NateHB@72.34.90.74) |
| 18:36.21 | HorizonXP | i was missing the _ before the number |
| 18:36.23 | HorizonXP | fudge! |
| 18:36.25 | HorizonXP | lol |
| 18:38.54 | NateHB | hey guys, i probably should have asked in #Freeebpx, but im using a f*cked irc client ATM, hey I recieved a call froma frei nd last night it came in from a did that I have set to forward via a misc destination to my cell phone, but i cant find the phone number to call him back |
| 18:40.03 | NateHB | i need to find this phone number, came in last night\ |
| 18:42.24 | NateHB | and just so you know how important this is, he called me last night with a formula for unlimited free fuel, that will transform the world, and end suffering and bring world peace and probably lead to the Federation of Planets and shit, so its really imporatns tthat I find this number |
| 18:42.41 | eppigy | comedy gold |
| 18:42.58 | [TK]D-Fender | desperate retad |
| 18:43.06 | [TK]D-Fender | not even worth spelling right :p |
| 18:43.11 | eppigy | true |
| 18:43.54 | [TK]D-Fender | NateHB: Go look in your CDRS to see the callerID he came in as |
| 18:44.24 | NateHB | no, really its real, i just need some help to find what CID he dialed in with, F-ing pbx in a flash does not forward caller id info in this scenario, so I need to get the info from the FreeBPX admin website |
| 18:44.44 | [TK]D-Fender | NateHB: Go look in your CDR to see the callerID he came in as <----------- |
| 18:45.30 | mltlnx | <PROTECTED> |
| 18:45.39 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 18:46.32 | eppigy | ITS REALLY REAL |
| 18:56.31 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 18:56.34 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 19:07.09 | mltlnx | still no success |
| 19:10.20 | NateHB | nothing, damn fate of the world hangs in ballance |
| 19:11.37 | *** part/#asterisk NateHB (n=NateHB@72.34.90.74) |
| 19:11.54 | rob0 | Sic transit gloria mundi. |
| 19:13.51 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 19:16.12 | *** join/#asterisk Octothorpe (n=Octothor@pdpc/supporter/professional/octothorpe) |
| 19:20.16 | *** join/#asterisk vvpalin (n=vvpalin@fay.dreamhost.com) |
| 19:25.21 | *** join/#asterisk Jarak01 (n=no-body0@e179197197.adsl.alicedsl.de) |
| 19:25.37 | Jarak01 | hello! |
| 19:26.02 | Jarak01 | ariel_ are you there? |
| 19:26.27 | Jarak01 | hmmm guess no. |
| 19:27.03 | Jarak01 | i need a hint. i installed asterisknow and now i dont know what to do^^ |
| 19:27.20 | Jarak01 | what can i do with asterisk? |
| 19:27.50 | Jarak01 | how can i use it, i mean i have installed an isnd-card but i also have the newest fritz!box |
| 19:27.57 | Jarak01 | anyone who can help? |
| 19:28.07 | Jarak01 | or who can at least show me the direction |
| 19:31.13 | gr0mit | well this group is really only for native asterisk supooirt |
| 19:31.32 | Jarak01 | isnt it the same |
| 19:31.39 | gr0mit | if you are running a gui version then you are not really in the right place |
| 19:32.17 | Jarak01 | y but i dont have to use the gui version do i? |
| 19:32.22 | gr0mit | if you did an apt-get install asterisk or built from source the nwe are here to assist |
| 19:32.35 | gr0mit | well the gui adds a lot of complexity |
| 19:32.50 | gr0mit | which i have never used |
| 19:33.20 | Jarak01 | i tried to install asterisk with opensuse 10.1 |
| 19:33.23 | *** join/#asterisk jtodd (i=e95ss4un@ns.fox-den.com) |
| 19:33.23 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 19:33.33 | Jarak01 | for 1 week |
| 19:33.37 | Jarak01 | was soo hard |
| 19:33.37 | gr0mit | it is possible to make a pbx with only a few lines of config |
| 19:33.55 | gr0mit | it is hard to start with |
| 19:33.57 | Jarak01 | iam a linux nerd |
| 19:33.58 | Jarak01 | y |
| 19:34.07 | gr0mit | but so is riding a bicycle. |
| 19:34.24 | Jarak01 | y but i rad so many tutorials and arcticles |
| 19:34.29 | Jarak01 | 1 weeek at least 3 hours |
| 19:35.07 | Jarak01 | nevermind;) |
| 19:35.19 | Jarak01 | so is it bad to have a gui? |
| 19:35.40 | kaldemar | i never ran into any trouble with opensuse in versions 9.1->11.1 |
| 19:35.54 | gr0mit | well, if you want to understand what is goiong on you wll have prbs with the gui |
| 19:36.19 | gr0mit | so, with your original*installation, is it still running? |
| 19:36.39 | gr0mit | Or did you wipe it and start again? |
| 19:37.27 | gr0mit | All my*installations have been done on a Debian environment |
| 19:38.00 | gr0mit | but no reason why your sse should have problems |
| 19:38.33 | gr0mit | so, do you have a working demo*installation on your open sluice box? |
| 19:40.07 | gr0mit | waits for the update |
| 19:40.56 | gr0mit | thinks that Jarak01 has gone to sleep |
| 19:43.33 | Jarak01 | gr0mit |
| 19:43.35 | Jarak01 | i wiped it |
| 19:43.44 | gr0mit | oh dear. |
| 19:43.47 | Jarak01 | and installed this asterisknow |
| 19:43.48 | Jarak01 | y y |
| 19:44.06 | gr0mit | Well, if you had a working Debian etch installation I could talk you through getting a basic asterisk system running |
| 19:44.24 | gr0mit | but*now is something of a rat's nest which I have never used |
| 19:44.27 | Jarak01 | gr0mit you use msn? |
| 19:44.37 | gr0mit | yes |
| 19:46.17 | gr0mit | so, do you have hard phonesor soft phones |
| 19:47.59 | *** join/#asterisk mltlnx (n=mltlnx@cpe-24-193-11-14.nyc.res.rr.com) |
| 19:50.40 | Jarak01 | where can i check for usernamen and password |
| 19:52.34 | gr0mit | on your voicemail box? |
| 19:53.29 | Jarak01 | y but i cant logg into the voicemails box ^^ |
| 19:53.54 | gr0mit | okay here's the problem. We have no idea how you have configured this*now |
| 19:54.30 | [TK]D-Fender | Jarak01: You've been told before. GUI's are not supported here |
| 19:54.36 | gr0mit | if you want support with*now, then you will need to find a support forum that can help. If you want to do a raw*installation either from source all using it then installation package, and we will be able to assist |
| 19:57.32 | Jarak01 | okay iam sorry. |
| 19:57.49 | Jarak01 | when i install asterisk normal, can i install a gui? |
| 19:57.55 | Jarak01 | for user |
| 19:58.17 | gr0mit | not recommended. |
| 19:58.31 | gr0mit | You always know exactly where you are with text files |
| 19:58.43 | gr0mit | and you can edit them from anywhere in the world via ssh |
| 19:58.51 | Jarak01 | y as i do know |
| 19:58.55 | Jarak01 | putty |
| 19:59.35 | Jarak01 | okay do you have any actual tutorials? becoz the tutorials i found for opensuse are from 2005/6 |
| 19:59.37 | Jarak01 | ^^ |
| 19:59.41 | Jarak01 | ahh nevermind |
| 19:59.44 | Jarak01 | i use google for this |
| 19:59.47 | Jarak01 | thank you anyway |
| 19:59.48 | gr0mit | so, if you want something that you could administer easily and reliably use text. If you want something that looks pretty but is hopelessly involved and muddlesome then your gui is fine. |
| 20:00.03 | gr0mit | There is a very good book on*which you can download for free |
| 20:00.17 | gr0mit | ~book |
| 20:00.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:00.51 | gr0mit | Andy Hughes the raw textbased files, you will get a lot of support in here |
| 20:01.01 | gr0mit | I mean, if you use the raw textbased files |
| 20:01.10 | gr0mit | Dragon NaturallySpeaking is not very good! |
| 20:01.59 | Jarak01 | thank you |
| 20:02.10 | Jarak01 | gr0mit |
| 20:02.19 | Jarak01 | but youre a gosu in linux |
| 20:02.24 | Jarak01 | iam faar far away from that |
| 20:02.39 | gr0mit | <PROTECTED> |
| 20:02.41 | [TK]D-Fender | Jarak01: * != linux |
| 20:02.48 | gr0mit | which is why I am keen to help you get something working |
| 20:02.57 | [TK]D-Fender | Jarak01: There are books, and sites. |
| 20:03.34 | gr0mit | I owe the last five years of my career and my future business venture to the people that got me up and running in this forum |
| 20:03.53 | gr0mit | when I also had no experience of*, and had never compiled anything from source before |
| 20:04.01 | Jarak01 | same to me |
| 20:04.07 | Jarak01 | but i compiled |
| 20:04.18 | Jarak01 | days ago |
| 20:04.22 | Jarak01 | had only problems |
| 20:04.22 | gr0mit | okay, so you compiled* |
| 20:04.31 | gr0mit | okay but what problems did you have? |
| 20:04.33 | Jarak01 | y for asterisk |
| 20:04.42 | gr0mit | Did you get it to the point where it compiled? |
| 20:04.42 | Jarak01 | and nobody was able 2 help me out |
| 20:04.51 | Jarak01 | opensuse |
| 20:04.54 | gr0mit | Well, we are all volunteers here |
| 20:05.11 | gr0mit | I think you'd find that quite a lot of us here would be happy to assist on a consultancy basis |
| 20:05.14 | Jarak01 | zaptel |
| 20:05.16 | Jarak01 | libpri |
| 20:05.25 | gr0mit | but the whole purpose of open source is contributing and benefiting each other |
| 20:05.32 | Jarak01 | word |
| 20:05.43 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 20:06.43 | gr0mit | so if you got*to the point at which it compiled were you able to run it and gets to the command line? |
| 20:10.04 | gr0mit | drums his paws on the desk |
| 20:10.25 | Jarak01 | ^^ |
| 20:10.34 | *** join/#asterisk propellerhead (n=yogurt2u@host241.190-31-158.telecom.net.ar) |
| 20:10.50 | Aiatek | anybody here has make a clsuter installation of asterisk with the new features in *1.6 or DRBD |
| 20:10.54 | Aiatek | ? |
| 20:10.54 | Jarak01 | http://pastebin.com/m7ffe257c |
| 20:10.55 | Jarak01 | for axample |
| 20:11.04 | Jarak01 | example |
| 20:11.11 | Aiatek | cluster |
| 20:11.25 | Jarak01 | then i recieved msg like no permission |
| 20:11.43 | Jarak01 | ah nevermind:( |
| 20:11.48 | Jarak01 | gr0mit |
| 20:11.54 | Jarak01 | thank you for your patience;) |
| 20:12.16 | gr0mit | well, the easiest way is to run*as root |
| 20:12.23 | Jarak01 | well y i know;) |
| 20:12.24 | gr0mit | but not recommended for production environments |
| 20:12.27 | Jarak01 | this wasnt the problem |
| 20:12.40 | Jarak01 | make version was kind of wrong |
| 20:12.43 | Jarak01 | i fixed it |
| 20:13.01 | *** join/#asterisk sikanrong (n=sikanron@78.105.219.87.dynamic.jazztel.es) |
| 20:13.03 | Jarak01 | but then i had many otherproblems |
| 20:13.09 | gr0mit | okay, |
| 20:13.21 | gr0mit | so am I right in thinking that you didn't even get to the point where you could run*? |
| 20:13.22 | Jarak01 | anyways : what do i need for phoning with a hardware phone? |
| 20:13.35 | Jarak01 | right! |
| 20:13.51 | Jarak01 | that was the worst of all:D |
| 20:13.55 | sikanrong | hey all, having problem here - hooked it up so that my voovox account calls my asterisk box via IAX2 - which seems to work great - anyway the problem is it's meant to call my cellphone, but it doesn't, it SAYS it's calling in the log, but then it says it answers (without even ringing!) |
| 20:13.58 | sikanrong | dig the log: pastie.org/510931 |
| 20:14.04 | gr0mit | Well, to start with I would use a Debian distribution |
| 20:14.05 | sikanrong | anybody experienced this before? |
| 20:14.14 | gr0mit | and then just apt get install* |
| 20:14.15 | Jarak01 | yes i will |
| 20:14.16 | Jarak01 | debian |
| 20:14.35 | sikanrong | when i do this using the softphone it actually works and makes the outbound call |
| 20:14.43 | sikanrong | something is going wrong within asterisk i think |
| 20:14.46 | sikanrong | though it's hard to say |
| 20:14.48 | gr0mit | you can get phone calls working with about five lines of configuration files |
| 20:14.57 | gr0mit | if that |
| 20:15.20 | sikanrong | right - it *is* mostly working as well, |
| 20:15.26 | gr0mit | when you want to start using ISDN or analog cards, it gets slightly more complicated |
| 20:15.37 | gr0mit | I gave up a long time ago with analog |
| 20:16.09 | [TK]D-Fender | sikanrong: Where are you calling to? |
| 20:16.16 | sikanrong | basically I've got a DID from voovox, and it goes to my IAX2 user, and then gets redirected to a context which looks a lot like Dial(_X,1,1480xxxxxxx@voovox-outbound); |
| 20:16.19 | gr0mit | but I've got various flavours of ISDN running in the past with asterisk boxes as far afield as Bogota Caracas Basingstoke and Melbourne |
| 20:16.27 | sikanrong | that's not the number though, i'm calling to spain |
| 20:16.36 | gr0mit | and I even have MFC or two running on a box in Mexico |
| 20:16.49 | sikanrong | but if I dial from my softphone and use the exact same context, the cellphone actually rings! |
| 20:17.01 | sikanrong | but only when i connect the call via the inbound IAX2, then it doesn't work |
| 20:17.10 | sikanrong | err, it kinda works; look at the log |
| 20:17.18 | sikanrong | it says i answer, but phone doesn't even ring! |
| 20:17.25 | sikanrong | sorry btw, i know it's a weird/complicated issue |
| 20:17.30 | sikanrong | but this is what I have to go on... |
| 20:17.51 | gr0mit | that sounds like a telco problem, sikanrong |
| 20:18.13 | gr0mit | it should not answer the channel unless the called party actually answers |
| 20:18.21 | Jarak01 | what do i need to install when i want to use isdn and holdonmusic |
| 20:18.30 | sikanrong | gr0mit: but then why would it complete the call correctly when i dial the same extention from a softphone? |
| 20:18.37 | Jarak01 | phoning from a hardphone |
| 20:19.05 | Aiatek | <[TK]D-Fender> have you try yet the cluster funtion of asterisk 1.6? |
| 20:19.51 | [TK]D-Fender | Aiatek: nope |
| 20:20.09 | Aiatek | ok |
| 20:20.18 | gr0mit | sikanrong, okay, you have a an incoming call from IAX which you want to forward out on to the PSTN? |
| 20:20.31 | sikanrong | yes indeed, via a SIP provider |
| 20:20.45 | Aiatek | i have one running but with DRBD |
| 20:21.01 | gr0mit | If you terminate the incoming call on your soft phone, then you have two-way audio, do you? |
| 20:21.14 | sikanrong | yeah, actually - everything works like magic then |
| 20:21.16 | sikanrong | very weird |
| 20:21.16 | Aiatek | but i like to try with just asterisk without DRBD |
| 20:21.49 | gr0mit | And you are using the same codec on incoming and outgoing? (Grommet clutches at straws |
| 20:22.34 | sikanrong | the issue (in the log) comes about when it says SIP/voovox-outbound-xxxxx answered IAX2/voovox - the codecs seem to not be the issue, i have gsm and ilbc set on both |
| 20:23.04 | sikanrong | it's really the improper termination, I'm saying, since the phone doesn't even ring - it's not really an audio issue |
| 20:23.19 | gr0mit | I was wondering if he was a one-way audio problem |
| 20:23.30 | gr0mit | but it does not seem like that from what you've said |
| 20:23.42 | gr0mit | one-way audio is frequently caused by NAT problems |
| 20:23.44 | sikanrong | I'd be *happy* with a one-way audio problem, but the phone not ringing is really the weird bit :) |
| 20:24.00 | gr0mit | I agree! |
| 20:24.09 | sikanrong | like i said though, if i do this from softphone and dialing out from asterisk via the voovox-outbound, everything works like magic |
| 20:24.19 | sikanrong | two-way sound, phones ringing, whole thing |
| 20:24.25 | gr0mit | And the soft phone is connected to your local asterisk box, right? |
| 20:24.36 | sikanrong | yeah, it's registered via the local asterisk box |
| 20:24.51 | sikanrong | thanks for all the help, btw |
| 20:25.05 | gr0mit | <PROTECTED> |
| 20:25.15 | sikanrong | i think so, most recent in ubuntu repositories |
| 20:25.38 | gr0mit | Can you check? I tried 1.6 but gave up and reverted to 1.4 |
| 20:25.45 | gr0mit | I had various audio related problems |
| 20:25.51 | sikanrong | how do I find out? |
| 20:25.59 | gr0mit | go to the*see a lie |
| 20:26.00 | sikanrong | asterisk --version isnt working |
| 20:26.02 | gr0mit | CLI |
| 20:26.07 | sikanrong | I'm in cli also |
| 20:26.10 | sikanrong | what's the command? |
| 20:26.13 | gr0mit | I think it is core show version |
| 20:26.22 | sikanrong | 1.4.21 |
| 20:26.34 | gr0mit | okay, a fairly stable release |
| 20:26.44 | gr0mit | I am using 1.4.22 on my production boxes |
| 20:27.24 | sikanrong | maybe you're right and it is a telco problem, it really *seems* like it would be an asterisk issue though |
| 20:27.36 | sikanrong | why would the SIP user report an answer without ever ringing the phone, i mean? |
| 20:27.42 | sikanrong | and why only under these conditions?? |
| 20:27.46 | gr0mit | have you tried calling a land line instead of a mobile? |
| 20:27.59 | sikanrong | no, actually - i should do... |
| 20:28.05 | sikanrong | what's a good USA # to call |
| 20:28.09 | gr0mit | Wonder if the mobile voicemail is cutting into some reason? I do think it is probably an asterisk problem though |
| 20:28.11 | sikanrong | 1-800-... |
| 20:28.20 | gr0mit | one 800 Airways |
| 20:28.29 | sikanrong | but in #s that would be... |
| 20:28.54 | rob0 | 1-800-328-7448 |
| 20:29.00 | sikanrong | thx |
| 20:29.16 | rob0 | um, no, that's not airways |
| 20:29.28 | gr0mit | okay, can you paste a chunk of your extensions.conf file into a paste bin |
| 20:29.38 | rob0 | that's EAT-SHIV |
| 20:29.43 | gr0mit | blanking out any passwords or telephone numbers, of course |
| 20:29.45 | sikanrong | hahaha |
| 20:29.54 | sikanrong | sure, let me pastie.caboo.se that beast |
| 20:30.09 | sikanrong | btw - same issue with 1800eatshiv |
| 20:30.21 | gr0mit | ok |
| 20:30.32 | sikanrong | i wonder maybe if using IAX instead of IAX2 could be a solution? |
| 20:30.42 | gr0mit | I think this must be a configuration problem |
| 20:30.53 | sikanrong | i have a feeling too.. I'm pastieing.. |
| 20:32.01 | [TK]D-Fender | IAX has not existed for over half a decade |
| 20:32.26 | gr0mit | he went out before I joined the*Revolution |
| 20:32.49 | sikanrong | alright gr0mit |
| 20:33.01 | sikanrong | config files at pastie.org/510948 |
| 20:33.20 | sikanrong | help the noob!!! sorry about the random seeming ruby syntax |
| 20:33.42 | sikanrong | crap - btw |
| 20:33.51 | sikanrong | in the iax config |
| 20:33.53 | gr0mit | okay, looking now. |
| 20:34.05 | sikanrong | it's supposed to be context=softphone |
| 20:34.19 | sikanrong | i mean, in extensions.conf it's really called [softphone] |
| 20:34.29 | sikanrong | but i renamed it cellphone to avoid confusion (only on the pastie) |
| 20:34.37 | sikanrong | but then didn't rename it in the iax.conf part |
| 20:34.45 | sikanrong | so beware, that part is actually right in the configs |
| 20:36.41 | gr0mit | okay, well I can't see anything glaringly obvious |
| 20:37.40 | *** join/#asterisk mohsen-ece (n=ahmed@41.196.81.139) |
| 20:37.49 | sikanrong | yeah - i can't either... |
| 20:37.51 | sikanrong | so weird! |
| 20:38.33 | *** part/#asterisk mohsen-ece (n=ahmed@41.196.81.139) |
| 20:38.34 | gr0mit | what happens if you try calling a number in Spain for example or wherever you are |
| 20:38.47 | sikanrong | well that's the thing |
| 20:38.51 | sikanrong | If i do it from softphone |
| 20:38.53 | sikanrong | works perfect |
| 20:39.12 | sikanrong | when the IAX user does it, for some reason it answers the call without actually ringing it (face-palm...) |
| 20:39.48 | gr0mit | is confident that there is a DOH! somewhere |
| 20:40.45 | sikanrong | me too gr0mit, i can't wait to figure out what it is... |
| 20:40.59 | sikanrong | I'm getting hungry though, been at this for HOURS. promised myself I would do it this weekend.. |
| 20:41.09 | sikanrong | anyway - I'm still in it for now. |
| 20:41.20 | gr0mit | so your incoming IAX calls come in to context soft phone |
| 20:41.53 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.70) |
| 20:42.02 | sikanrong | yeah |
| 20:42.11 | sikanrong | and all that has is the one Dial command |
| 20:42.30 | gr0mit | and your context soft phone contains exten => _X.,1,Dial(SIP/18003287448@voovox-outbound) |
| 20:42.36 | sikanrong | yeah |
| 20:42.48 | gr0mit | so whatever you dial should call that number |
| 20:42.52 | sikanrong | indeed |
| 20:42.57 | sikanrong | and from softphone, it does |
| 20:43.10 | sikanrong | in the log it even says "Calling xxxxxx" |
| 20:43.17 | sikanrong | but then it just says "answered" on next line |
| 20:43.30 | sikanrong | from softphone it says "voovox-outbound-32984 is ringing..." |
| 20:43.34 | sikanrong | which is correct |
| 20:43.45 | Aiatek | thats for incoming calls |
| 20:43.46 | Aiatek | ? |
| 20:43.50 | sikanrong | yeah |
| 20:44.01 | Aiatek | did you tried with s |
| 20:44.09 | gr0mit | and have you taken a full sip trace? |
| 20:44.17 | sikanrong | let me try the full sip trace |
| 20:44.29 | gr0mit | it might give you a clue |
| 20:45.21 | sikanrong | wow, that's a lot.. gonna have to think about this one.. |
| 20:45.22 | gr0mit | one thing I have come across relates to caller ID. One of my wholesalers doesn't connect the call if I don't send the correct outgoing caller ID |
| 20:45.40 | gr0mit | and it might not like the incoming caller ID from iax |
| 20:45.58 | gr0mit | it might be worth putting a line in to change the caller ID before you make the outgoing sip call |
| 20:46.26 | gr0mit | just a thought. |
| 20:47.02 | sikanrong | check out http://pastie.org/510954 |
| 20:47.07 | sikanrong | it's the sip dump |
| 20:49.05 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:49.15 | sikanrong | line 225 is when (for no reason at all) it answers the incoming call |
| 20:51.53 | [TK]D-Fender | sikanrong: Contact: <sip:463910@192.168.1.3> <--- your server is not correctly set up for NAT. |
| 20:51.58 | [TK]D-Fender | ~sipnat |
| 20:51.58 | 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 |
| 20:52.00 | [TK]D-Fender | ^^ |
| 20:52.33 | gr0mit | yes, I was just getting to that point to! |
| 20:52.36 | sikanrong | god i hope this is the deal |
| 20:53.09 | sikanrong | no - i do have it all configured for NAT though |
| 20:53.17 | sikanrong | i mean, ports are forwarded and everything |
| 20:53.26 | gr0mit | yes, but you need to rewrite the IP address in the zip had |
| 20:53.31 | gr0mit | zip header |
| 20:53.35 | gr0mit | sip header |
| 20:53.43 | sikanrong | how do I do that though? |
| 20:54.05 | gr0mit | if you or a DSL line has a static IP address that is fairly straightforward |
| 20:54.11 | [TK]D-Fender | [16:53]<sikanrong>no - i do have it all configured for NAT though <--- look at the contact header I just pasted. No, it is NOT correct |
| 20:54.42 | sikanrong | right - get that, but i mean how do I change it? I haven't entered the local address in any of my asterisk config files |
| 20:54.44 | [TK]D-Fender | sikanrong: Go follwo the guide I provided you |
| 20:54.50 | gr0mit | can you paste bin the whole of your sip.conf? |
| 20:55.05 | gr0mit | in the general section there are things you can set up |
| 20:55.11 | [TK]D-Fender | gr0mit: No need, we know its wrong and he hasn't done anything. |
| 20:55.19 | gr0mit | so that*spoofs its zip headers and sends out your public IP address |
| 20:55.36 | eppigy | what |
| 20:55.40 | sikanrong | for real? |
| 20:55.48 | sikanrong | like bindaddr or something? |
| 20:55.57 | gr0mit | <PROTECTED> |
| 20:56.03 | eppigy | yall trippin |
| 20:56.06 | gr0mit | but you also need to tell*what your internal IP address range is |
| 20:56.11 | sikanrong | eppigy: elaborate? |
| 20:56.20 | eppigy | yall talking about nat/pat |
| 20:56.24 | eppigy | I mean whats the deal |
| 20:56.28 | [TK]D-Fender | sikanrong: Go follow the guide I provided you <---------------------- |
| 20:56.29 | eppigy | i see the word spoofing |
| 20:56.31 | eppigy | being used |
| 20:56.43 | eppigy | and it firghtens me |
| 20:56.48 | sikanrong | Fender: the guide is confusing as I don't know which of the 9 categories I fall into |
| 20:56.48 | eppigy | im not gonna lie |
| 20:57.05 | sikanrong | I'm trying though... |
| 20:57.29 | eppigy | DI DI MAO |
| 20:57.32 | [TK]D-Fender | sikanrong: FIRST BLOODY LINK |
| 20:58.09 | gr0mit | # Asterisk as a SIP client behind nat, connecting to outside SIP Proxies |
| 20:58.29 | *** join/#asterisk Jarak01 (n=no-body0@f054124108.adsl.alicedsl.de) |
| 20:58.33 | Jarak01 | what is better for isdn phoning? sip, zap(dahdi compatible), enum, dundi or iax |
| 20:58.36 | sikanrong | alright - also, I'd like to point out that when I do this from the softphone, the SIP headers look exactly the damn same, except everything totally works |
| 20:58.52 | sikanrong | so, why is this suddenly the issue? |
| 20:59.23 | Chainsaw | Jarak01: Only zap or DAHDI can even connect to an ISDN phone line. |
| 20:59.38 | Jarak01 | ah i c, thanks alot chainsaw |
| 20:59.48 | Chainsaw | Jarak01: enum & dundi have nothing to do with ISDN and neither does SIP. |
| 20:59.56 | Jarak01 | okay |
| 21:00.00 | gr0mit | I think it is all quite complex |
| 21:00.04 | Jarak01 | but its still voip? |
| 21:00.10 | [TK]D-Fender | Chainsaw: And plenty MORE than Zap/DAHDI can. |
| 21:00.45 | Chainsaw | [TK]D-Fender: It could, and then you'd likely interface through SIP or IAX. |
| 21:00.56 | Chainsaw | [TK]D-Fender: Suppose I should have said "question is ambiguous" and refused to answer. |
| 21:01.01 | gr0mit | sikanrong, when you call from your soft phone, since you don't have reinvite=no, the call may in fact be handed off directly to your soft phone |
| 21:01.39 | sikanrong | hmm... |
| 21:01.41 | gr0mit | can you paste the general section of your sip.conf please |
| 21:01.58 | gr0mit | I think the answer to your trouble will probably be here |
| 21:02.37 | sikanrong | pastie.org/510966 |
| 21:04.58 | gr0mit | okay, well I can't see anything which you need in order to resolve the NAT problems |
| 21:05.24 | sikanrong | hmm... |
| 21:06.39 | gr0mit | http://pastie.org/510968 |
| 21:06.47 | gr0mit | these are my headings |
| 21:07.08 | gr0mit | basically, you tell*watch the external IP address is, and when to use it in place of your internal IP address |
| 21:07.29 | gr0mit | sorry about the typing, I use Dragon NaturallySpeaking and it is not terribly good for abbreviations |
| 21:08.40 | Aiatek | one question <gr0mit> about hte paste |
| 21:08.57 | gr0mit | <PROTECTED> |
| 21:09.02 | Aiatek | localnet=192.168.100.201/255.255.255.0 ; Internal NETWORK address |
| 21:09.10 | Jarak01 | what are dialrules |
| 21:09.28 | Aiatek | you have to put the ip address of the pbx or the network |
| 21:09.29 | Aiatek | ? |
| 21:09.57 | gr0mit | <PROTECTED> |
| 21:10.09 | Aiatek | i know that |
| 21:10.23 | gr0mit | <PROTECTED> |
| 21:10.28 | gr0mit | I haven't told you that! |
| 21:10.35 | Aiatek | because i was here |
| 21:10.47 | Aiatek | checking the paste |
| 21:10.49 | Aiatek | etc |
| 21:11.29 | gr0mit | The external IP is the public IP address that my asterisk box maps to |
| 21:11.43 | sikanrong | k, am looking |
| 21:11.52 | gr0mit | my asterisk box itself has two internal IP addresses |
| 21:12.07 | gr0mit | bindaddr just tells asterisk only to concern itself with the 201 |
| 21:12.15 | Aiatek | but what i want to know is localnet its fr ip or for the network |
| 21:12.42 | gr0mit | local net tells asterisk when it should not mangle the sip headers |
| 21:12.52 | drmessano | localnet is the network, not the specific IP |
| 21:12.54 | Jarak01 | zap extensions are kind of usernames ainit? |
| 21:13.05 | Aiatek | you put an specific ip |
| 21:13.08 | Aiatek | in the paste |
| 21:13.14 | drmessano | 192.168.1.0/255.255.255.0 |
| 21:13.23 | Aiatek | for localnet |
| 21:13.24 | drmessano | etc |
| 21:13.38 | drmessano | well thats not correct |
| 21:13.42 | gr0mit | yes, I might have got it wrong them! |
| 21:13.55 | Aiatek | ok |
| 21:14.06 | gr0mit | The difference is that my configuration works! |
| 21:14.13 | Aiatek | it could be localnet=192.168.100.0/24 |
| 21:14.25 | drmessano | localnet tells asterisk to use its internal IP vs the external IP set with externhost/externip |
| 21:14.30 | gr0mit | I agree that is what it should be but I don't think*support/notation |
| 21:14.34 | drmessano | for that range of IPs |
| 21:14.41 | Aiatek | yes |
| 21:14.48 | Aiatek | it support / |
| 21:14.49 | drmessano | asterisk does support /24 |
| 21:15.18 | gr0mit | stand corrected and will corrected file in due course |
| 21:15.46 | gr0mit | I read this file probably three years ago and haven't changed since |
| 21:15.51 | Aiatek | :P |
| 21:15.58 | gr0mit | I have learnt a lot about subnets since then |
| 21:16.08 | *** join/#asterisk s14ck (n=s14ck@190-76-85-145.dyn.movilnet.com.ve) |
| 21:16.58 | Aiatek | hi s14ck |
| 21:17.06 | gr0mit | anyhow, enough of all this. I am going to bed. sikanrong, good luck and I hope you get it working. |
| 21:17.11 | s14ck | Aiatek, whaz up man? |
| 21:17.17 | Aiatek | fine |
| 21:17.20 | sikanrong | good night, thanks for the help! |
| 21:17.32 | gr0mit | You're welcome |
| 21:18.09 | s14ck | Aiatek, tell me what news about the web site? |
| 21:18.41 | Aiatek | i found a guy who will help us with the desing part |
| 21:19.29 | *** join/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:19.49 | knarfly | anyone know about the bsd port for asterisk? |
| 21:20.52 | s14ck | Aiatek, nice |
| 21:25.04 | *** part/#asterisk knarfly (n=vtserije@c-75-74-113-9.hsd1.fl.comcast.net) |
| 21:29.16 | *** join/#asterisk mltlnx (n=mltlnx@firewall.mserve.com) |
| 21:31.08 | *** join/#asterisk HorizonXP (n=xitij@69-165-155-154.dsl.teksavvy.com) |
| 21:59.27 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 22:01.46 | dacs | is there is a proper way to uninstall asterisk, i am trying to uninstall 1.4 and install 1.4 |
| 22:02.28 | [TK]D-Fender | <PROTECTED> |
| 22:03.51 | astrutt | <PROTECTED> |
| 22:04.13 | dacs | funny :) |
| 22:07.45 | rob0 | The answer to that might depend on how you installed it to begin with. |
| 22:08.20 | rob0 | If from source, you'd be asking in the right place, but you should look in the Makefile first. |
| 22:08.37 | *** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 22:09.17 | dacs | rob0: yes from source |
| 22:09.26 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 22:11.47 | [TK]D-Fender | dacs: Just install over. But why do you feel you have to anyway? |
| 22:17.08 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 22:17.08 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 22:20.08 | *** join/#asterisk devyll (n=email@213.233.93.48) |
| 22:20.45 | dacs | just trying to make a clean install |
| 22:20.58 | devyll | DAHDI_ATTACH_ECHOCAN failed on channel 1: Invalid argument (22) . I installed the digium card after installing dahdi modules. Do you think it may work if I reinstall the dahdi packages ? |
| 22:22.39 | tzafrir_laptop | devsys, what version of dahdi? |
| 22:23.17 | [TK]D-Fender | dacs: just wipe /etc/asterisk and /usr/lib/asterisk/modules and start over |
| 22:24.18 | tzafrir_laptop | [TK]D-Fender, that's not the modules of asterisk |
| 22:25.27 | [TK]D-Fender | tzafrir_laptop: Sure looks like it to me on CentOS anyway |
| 22:32.22 | devyll | tzafrir_laptop 2.1.0.4 |
| 22:32.49 | [TK]D-Fender | tzafrir_laptop: What modules are you thinking of? |
| 22:33.01 | tzafrir_laptop | dahdi (kernel...) |
| 22:35.57 | mltlnx | So in 1.6.1.1 parked calls that timeout do not get back to the parker. Apparently the dial string that is crafted uses pipes(|) instead of commas. .....Seems to be fixed in trunk though. I cannot however get the parkinglot channel variable acknowledged. Any ideas? |
| 22:37.26 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 22:38.02 | *** join/#asterisk nix8n82 (n=nate@mo-65-41-196-62.sta.embarqhsd.net) |
| 22:39.36 | devyll | tzafrir_laptop , can you help me please ? |
| 22:39.59 | tzafrir_laptop | could you please pastebin your /etc/dahdi/system.conf ? |
| 22:40.14 | Jarak01 | zap extensions are kind of usernames ainit? |
| 22:47.17 | devyll | tzafrir_laptop it is generated with dahdi_genconf. no customization done. I will pastebin it now |
| 22:47.37 | tzafrir_laptop | ok. so it should have echocancel lines |
| 22:47.55 | tzafrir_laptop | what version of asterisk? self built or from a package? |
| 22:48.00 | devyll | it has. |
| 22:48.25 | carrar | w00t |
| 22:49.07 | devyll | it is from a package. custom package done by the company I work for. |
| 22:49.10 | devyll | system.conf : http://pastebin.com/d23abaada |
| 22:49.23 | devyll | asterisk version : Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 22:51.43 | tzafrir_laptop | any chance that the package was built vs. dahdi-linux 2.2 ? |
| 22:52.58 | tzafrir_laptop | <PROTECTED> |
| 22:56.24 | *** join/#asterisk wonderworld (n=w@ip-62-143-16-28.unitymediagroup.de) |
| 22:59.54 | *** join/#asterisk devyll (n=email@213.233.93.48) |
| 23:00.22 | devyll | tzafrir_laptop : I believe that I currently have dahdi 2.1 |
| 23:01.07 | devyll | tzafrir_laptop and dahdi came also in custom packages. I also believe that the asterisk custom package should work perfectly with the dahdi packages. |
| 23:02.44 | tzafrir_laptop | I wonder if you can check with strace the actual ioctl number used there. Not sure how to get the thread id, though |
| 23:03.44 | devyll | I will try to talk with my colleagues who build the packages. hope they will find the problem also. |
| 23:03.47 | devyll | thank you tzafrir_laptop |
| 23:03.47 | infobot | de rien, devyll |
| 23:20.37 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 23:20.44 | ruben23 | hi |
| 23:22.14 | ruben23 | got idea on my error during installation of asterisk 1.2 on my centos distro |
| 23:22.25 | ruben23 | http://pastebin.com/m7b613677 |
| 23:34.43 | *** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net) |
| 23:35.09 | *** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com) |
| 23:50.09 | *** join/#asterisk dacs (n=dacs@unaffiliated/dacs) |
| 23:51.11 | dacs | is * Asterisk 1.4.25.1 stable ? |
| 23:53.15 | rob0 | from /topic, "1.4.25.1 (2009/06/05)," means that since 2009/06/05, there has been no need for a panic release. Since that's over a week, I'd guess it's pretty safe. (But I use 1.6.) |